有谁知道4W E&M的写法和表示方法是出自哪个标准？[电力自动化时代技术论坛]: "E&M Leads Signaling E&M前引作信
E&M前引作信是电讯工业中的一种信令类型。它显示采用与电话的耳朵（接听）和嘴巴（传输）相对应的电话听筒。
E&M Signaling E&M信令
E&M信令是DS0时隙值上采用的一种信令方法。例如，信令比特被用来显示呼叫的状态：挂断、接听、来电提示和拨号脉冲。
E&M: recEive and transMit接收和传输
接收和传输(E&M)，又被称为耳朵和嘴巴，是通常被用于双向交换对交换或交换对网络连接的中继安排。传输和接收也被用在E1和T1数字界面中。"
2008年9月29日星期一
FXO、FXS、E&M 接口的区别[电力自动化时代技术论坛]
FXO、FXS、E&M 接口的区别[电力自动化时代技术论坛]: "FXO、FXS、E&M 接口的区别
FXO、FXS、E&M 接口的区别
fxs 提供震铃电压，FXO 不提供!
FXO、FXS、E&M三种都是模拟信令，就好比足球比赛有美式足球和英式足球一样。
不同信令方式对控制信号的分类识别都不同，应用场合也不同。
在应用中可以简单理解为：
FXO为普通电话机接口，需要远程馈电；
FXS接口为PBX的内线分机接口，向远程馈电；
E&M为专用的一般用在PBX中继线接口。
CISCO提供的说法是：
FXO用于连接PSTN，二线（因为PSTN向用户馈电）
FXS用于连接POT普通电话机，二线（因为电话机需要FXS提供馈电信号）
E&M用于连接PBX，CISCO语音路由器可以设置二线或四线（因为PBX上可能配置E&M接口板）
在实际应用中PBX情况为：
PBX的中继线需要远程馈电；
内线分机提供馈电。
中继接口可能选择普通的模拟中继板和E&M接口。
因此，在使用中：
如果PBX上配置了E&M接口板当之无愧采用E&M接口，这也同CISCO的说法是一样的；
FXS接口用于连接电话机，连接PBX的普通中继线也可以，用CISCO设备的FXS接口向PBX中继线提供馈电信号；
FXO接口用于连接PSTN，用于连接PBX内线分机也可以，由PBX向FXO接口提供馈电信号。
以上是对于用户PBX的分析，对于局用机则更加灵活。
在实际使用中还发现，在使用FXO连接分机线是存在不挂机现象（本人实验结果是，在IOS12.1.4时为FXO接口对FXO接口通信不正常；在IOS12.1.5时FXO接口对所有接口的呼入可以自动挂断，但需要至少3~5分钟，用户不可能接受）。
根据各用户使用情况，本人认为小型的VOIP网络（不采用E1接口），采用FXS连接PBX普通中继线方式最佳，使用方便，而FXO接口不成熟，E&M接口对PBX有特殊要求，在PBX上另配E&M接口价格不低。
补充下：
家里的普通电话线，如果接上万用表你就可以测到电压。这说明电信局那端的是FXS接口，那么你的电话机就是FXO口。
公司的PBX有外线口和内线口之分，PBX的外线口是FXO，PBX的内线口是FXS。
FXO不能接FXO，FXS也不能接FXS（2个带电压的接在一起会烧线路），只能是FXS接FXO。
有电压的才能提供模拟的信令，就是说FXS提供信令。
模拟的信令是什么呢？就是一些频率。比如中国的国标是：
拨号音：450Mhz 的连续音
忙音：450Mhz 0.35s<On Time(ms)> 0.35<Off Time(ms)
等。。。。
为什么FXO的挂断有问题？因为它不提供信令，它是识别FXS来的信令（但很多情况下FXO识别不到挂断的频率音）。
怎么最好的解决这个问题？在电信局申请“极性反转”，并在FXO的配置中配置支持“极性反转”。"
FXO、FXS、E&M 接口的区别
fxs 提供震铃电压，FXO 不提供!
FXO、FXS、E&M三种都是模拟信令，就好比足球比赛有美式足球和英式足球一样。
不同信令方式对控制信号的分类识别都不同，应用场合也不同。
在应用中可以简单理解为：
FXO为普通电话机接口，需要远程馈电；
FXS接口为PBX的内线分机接口，向远程馈电；
E&M为专用的一般用在PBX中继线接口。
CISCO提供的说法是：
FXO用于连接PSTN，二线（因为PSTN向用户馈电）
FXS用于连接POT普通电话机，二线（因为电话机需要FXS提供馈电信号）
E&M用于连接PBX，CISCO语音路由器可以设置二线或四线（因为PBX上可能配置E&M接口板）
在实际应用中PBX情况为：
PBX的中继线需要远程馈电；
内线分机提供馈电。
中继接口可能选择普通的模拟中继板和E&M接口。
因此，在使用中：
如果PBX上配置了E&M接口板当之无愧采用E&M接口，这也同CISCO的说法是一样的；
FXS接口用于连接电话机，连接PBX的普通中继线也可以，用CISCO设备的FXS接口向PBX中继线提供馈电信号；
FXO接口用于连接PSTN，用于连接PBX内线分机也可以，由PBX向FXO接口提供馈电信号。
以上是对于用户PBX的分析，对于局用机则更加灵活。
在实际使用中还发现，在使用FXO连接分机线是存在不挂机现象（本人实验结果是，在IOS12.1.4时为FXO接口对FXO接口通信不正常；在IOS12.1.5时FXO接口对所有接口的呼入可以自动挂断，但需要至少3~5分钟，用户不可能接受）。
根据各用户使用情况，本人认为小型的VOIP网络（不采用E1接口），采用FXS连接PBX普通中继线方式最佳，使用方便，而FXO接口不成熟，E&M接口对PBX有特殊要求，在PBX上另配E&M接口价格不低。
补充下：
家里的普通电话线，如果接上万用表你就可以测到电压。这说明电信局那端的是FXS接口，那么你的电话机就是FXO口。
公司的PBX有外线口和内线口之分，PBX的外线口是FXO，PBX的内线口是FXS。
FXO不能接FXO，FXS也不能接FXS（2个带电压的接在一起会烧线路），只能是FXS接FXO。
有电压的才能提供模拟的信令，就是说FXS提供信令。
模拟的信令是什么呢？就是一些频率。比如中国的国标是：
拨号音：450Mhz 的连续音
忙音：450Mhz 0.35s<On Time(ms)> 0.35<Off Time(ms)
等。。。。
为什么FXO的挂断有问题？因为它不提供信令，它是识别FXS来的信令（但很多情况下FXO识别不到挂断的频率音）。
怎么最好的解决这个问题？在电信局申请“极性反转”，并在FXO的配置中配置支持“极性反转”。"
e&m fax relay
NetYourLife - Powered by Discuz! Board: "另外这个语音只有总部有几部IP电话，其他主要是用模拟电话打的，走程控交换机进入E&M接口
[ 本帖最后由 dazing 于 2008-3-4 03:22 PM 编辑 ] 作者: eve 时间: 2008-3-4 03:56 PM
兄弟，总公司内部没有FAX吗，看你dial-peer的配置，没有做FAX relay啊 作者: dazing 时间: 2008-3-4 04:28 PM
有 FAX接到程控交换机上了 FAX relay这个是什么作用？怎么配？ 作者: eve 时间: 2008-3-4 04:49 PM
FAX relay
工作原理
1，数字信号经过G3调制成模拟
2，然后再通过G3解调成数字
3，数字信号直接封装进IP,这样就避免的采样和量化，达到更好的传输效果
relay通过复原原始信号，而不是通过采样和量化来处理，使得效果更好
relay的标准由很多，cisco或T38
voice service voip
fax protocol t38 ls-redundancy 0 hs-redundancy 1 fallback cisco
或者在dial-peer下也可以配置，命令差不多
还有建议你配置DTMF realy，针对按键信息，可以在dial-peer下配置dtmf relay，效果是，比如你打声讯电话的尸首会提示你按1是中文，按2是英文这种IVR的应用时，防止按键信息被G711转没了 作者: dazing 时间: 2008-3-4 05:33 PM
受教了！
就是说FAX relay可以让FAX的流量不经过模拟-数字-模拟的转换？因为我的路由器和PBX都用的E&M接口 如果不配置FAX relay的话现在传真机应该也能使吧？
另外DMTF relay我当时看到了 但是没有看懂 不知道什么时候用 怎么用 作者: eve 时间: 2008-3-4 06:02 PM
不是，FAX relay是确保让FAX的模拟信号不经过G711或者G720采样和编码，因为这2种编码对语音来说是最好的，但对FAX来说会造成失真，导致发出去的文档多几个字或少几个字 作者: dazing 时间: 2008-3-4 06:08 PM
FAX的从PBX出来的时候是模拟的吧？到达路由器不用把它变成数字信号也可以传输吗？ 作者: eve 时间: 2008-3-4 07:26 PM
VOIP啊，不 IP传的是 数字信号啊 作者: dazing 时间: 2008-3-5 01:47 PM
那就是说FAX的模拟信号到达路由器之后，如果用了FAX relay的话，将采用另一种模拟转数字的编码方式来使FAX信息不会发生丢失是吗？ 作者: eve 时间: 2008-3-5 01:55 PM
不用FAX relay的时候，FAX发出的是模拟信号（本来是数字信号，经过T38编码成的模拟信号），到了路由器后，会根据你dial-peer里的配置进行G711或者G729的codec编码，这些编码对语音来说是很好的，基本不会造成声音的失真，但是FAX的模拟信号经过这些编码后，会造成失真，从而丢失一些信息，这不是我们希望的，所以引入FAX relay技术，由于FAX的模拟信号从FAX出来后会经过T38或者其他形式的FAX编码，用了这个技术后，不会对FAX的模拟信号进行G711或G729编码了，而是直接解调T38的编码，还原成数字信号后直接加报头封装进IP 作者: dazing 时间: 2008-3-5 05:00 PM
哦 我明白了 FAX会把信息通过T38转成模拟信号，所以使用FAX relay在用T38给转回数字信号，这样就没有失真了
真是谢谢你了！因为这是我第一次调VOIP，而且是自己翻CISCO文档一点一点做的，所以肯定有些地方做的不够完整，再次谢谢你的帮助！
对了，
voice service voip
fax protocol t38 ls-redundancy 0 hs-redundancy 1 fallback cisco
这句里面
ls-redundancy 0
hs-redundancy 1
fallback cisco
是什么意思呀？"
[ 本帖最后由 dazing 于 2008-3-4 03:22 PM 编辑 ] 作者: eve 时间: 2008-3-4 03:56 PM
兄弟，总公司内部没有FAX吗，看你dial-peer的配置，没有做FAX relay啊 作者: dazing 时间: 2008-3-4 04:28 PM
有 FAX接到程控交换机上了 FAX relay这个是什么作用？怎么配？ 作者: eve 时间: 2008-3-4 04:49 PM
FAX relay
工作原理
1，数字信号经过G3调制成模拟
2，然后再通过G3解调成数字
3，数字信号直接封装进IP,这样就避免的采样和量化，达到更好的传输效果
relay通过复原原始信号，而不是通过采样和量化来处理，使得效果更好
relay的标准由很多，cisco或T38
voice service voip
fax protocol t38 ls-redundancy 0 hs-redundancy 1 fallback cisco
或者在dial-peer下也可以配置，命令差不多
还有建议你配置DTMF realy，针对按键信息，可以在dial-peer下配置dtmf relay，效果是，比如你打声讯电话的尸首会提示你按1是中文，按2是英文这种IVR的应用时，防止按键信息被G711转没了 作者: dazing 时间: 2008-3-4 05:33 PM
受教了！
就是说FAX relay可以让FAX的流量不经过模拟-数字-模拟的转换？因为我的路由器和PBX都用的E&M接口 如果不配置FAX relay的话现在传真机应该也能使吧？
另外DMTF relay我当时看到了 但是没有看懂 不知道什么时候用 怎么用 作者: eve 时间: 2008-3-4 06:02 PM
不是，FAX relay是确保让FAX的模拟信号不经过G711或者G720采样和编码，因为这2种编码对语音来说是最好的，但对FAX来说会造成失真，导致发出去的文档多几个字或少几个字 作者: dazing 时间: 2008-3-4 06:08 PM
FAX的从PBX出来的时候是模拟的吧？到达路由器不用把它变成数字信号也可以传输吗？ 作者: eve 时间: 2008-3-4 07:26 PM
VOIP啊，不 IP传的是 数字信号啊 作者: dazing 时间: 2008-3-5 01:47 PM
那就是说FAX的模拟信号到达路由器之后，如果用了FAX relay的话，将采用另一种模拟转数字的编码方式来使FAX信息不会发生丢失是吗？ 作者: eve 时间: 2008-3-5 01:55 PM
不用FAX relay的时候，FAX发出的是模拟信号（本来是数字信号，经过T38编码成的模拟信号），到了路由器后，会根据你dial-peer里的配置进行G711或者G729的codec编码，这些编码对语音来说是很好的，基本不会造成声音的失真，但是FAX的模拟信号经过这些编码后，会造成失真，从而丢失一些信息，这不是我们希望的，所以引入FAX relay技术，由于FAX的模拟信号从FAX出来后会经过T38或者其他形式的FAX编码，用了这个技术后，不会对FAX的模拟信号进行G711或G729编码了，而是直接解调T38的编码，还原成数字信号后直接加报头封装进IP 作者: dazing 时间: 2008-3-5 05:00 PM
哦 我明白了 FAX会把信息通过T38转成模拟信号，所以使用FAX relay在用T38给转回数字信号，这样就没有失真了
真是谢谢你了！因为这是我第一次调VOIP，而且是自己翻CISCO文档一点一点做的，所以肯定有些地方做的不够完整，再次谢谢你的帮助！
对了，
voice service voip
fax protocol t38 ls-redundancy 0 hs-redundancy 1 fallback cisco
这句里面
ls-redundancy 0
hs-redundancy 1
fallback cisco
是什么意思呀？"
Q信令和PRI、DSS1、PRA的关系与区别？ - 交换技术 - 通信与网络 - 百思论坛
Q信令和PRI、DSS1、PRA的关系与区别？ - 交换技术 - 通信与网络 - 百思论坛: "Q信令和PRI、DSS1、PRA的关系与区别？
请教高手！
谢谢！
2007-8-27 09:19 #1
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summer
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PRI信令：又称ISDN（30B+D）信令、DSS1信令、PRA信令。在北美和日本，ISDN PRI提供23B+D信道，总速率达1.544Mbps，
其中D信道速率为64kbps。而在欧洲、澳大利亚等国家，ISDN PRI为30B+D，总速率达2.048Mbps。我国为30B+D方式。
ITU-T的I.412建议中规定了两种用户-网络接口结构：基本速率接口（BRI）和基群速率接口（PRI）。ISDN的接口通过时分复用技术，把一个物理接口划分为多个信道（时隙）来使用。ISDN的信道分为B、D两种类型，其中：B信道为用户信道，用来传送数据、话音、图像等用户信息，速率是64kbit/s；D信道为控制信道，用来传送公共信道信令，控制同一接口的B信道上的呼叫，速率是64kbit/s或16kbit/s。正是这样通过B通道和D通道的划分，ISDN接口实现了数据和控制流的分离。
在ITU-T规定的ISDN标准U-N接口中，BRI接口为2B+D，TS0（16kbit/s）为D信道，TS1、TS2（64kbit/s）分别作为两个B信道B1和B2；PRI接口分E1 PRI和T1 PRI两种，B、D信道的带宽均为64kbit/s。E1 PRI为30B+D，分TS0~TS31共32个时隙，TS0用于帧同步，TS16为D信道，一般在中国、亚洲部分国家和地区、欧洲等地使用；T1 PRI为23B+D，分TS0~TS23共24个时隙，TS23为D信道，一般在北美（北美把T1 PRI接口定义为PRA）、加拿大、日本、香港等地使用。
在BRI接口方面，由于各国的用户线特性有差异，各国使用的线路码型有所不同，北美、中国采用2B1Q码，日本、意大利采用AMI码，英国采用3B2T码。ITU-T的 I.430建议对BRI接口电气特性的各项指标均作了详细的规定。BRI接口最常见的配置是用户可以将话机、传真机和数据终端接在一对用户线上，使用户可以同时利用一对电话线通话、传送或接收传真而又进行数据通信。
在PRI接口方面，E1 PRI一般使用AMI（Alternate Mark Inversion)码和HDB3码（High Density Bipolar code with a maximum of 3 zeros)两种线路码型，T1 PRI一般使用AMI、B8ZS两种线路码型。PRI接口主要应用于：（1）数字程控交换（30个64k的B信道接入）+窄带上网业务（128k带宽）；（2）商业机构总部与各分部之间的信息接入直通道；（3）大型企业之间使用专用的会议电视设备，捆绑使用6个B信道（384k）可实现图像实时传送的会议电视业务、捆绑使用2个B信道（128k）可实现图像实时传送的可视电话业务。
2007-8-28 03:09 #2
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根据OSI参考模型，ISDN用户－网络接口协议分为三层：第一层是物理层，它包括基本接入接口和一次群速率接口；第二层是数据链路层；第三层是网络层。ISDN用户-网络接口链路层协议称为LAPD(Link Access Procedure on the D channel: D通路链路接入协议)。
ITU-T的建议Q.933定义在ISDN用户－网络接口上的第二层和第三层协议称为1号数字用户信令(DSS1)。DSS1的全称是Digital subscriber signalling system No.1，是ISDN D信道上的协议。
在ISDN用户－网络接口的T参考点上，第一层采用ITU-T的建议I.430（BRI）和建议I.431（PRI），第二层采用ITU-T的建议Q.921描述的LAPD协议，第三层采用ITU-T的Q.931建议。协议栈如图3所示
2007-8-28 10:23 #3
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hlpsl
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Q.921协议
终端端点标示符TEI（Terminate endpoint identifier）被用来标识ISDN用户侧的一个指定的连接端点。TEI值可以由网络侧分配，也可以是固定使用的。一般情况下，都是采用由网络侧分配的方式。网络侧可以分配给用户侧的TEI值为64～126（0～63为非自动分配的TEI值，127用于广播数据链路连接）。
Q.921的协商过程可以分为分配TEI、建立多帧操作（建链）、维护链路、拆除链路四步，因此，ISDN第二层有如下3个基本状态：TEI未分配、TEI已分配、多帧操作已建立
QSIG协议
QSIG也是ISDN的D信道上的协议，它是专用ISDN网络PBX直接互通的协议，是由国际标准化组织／国际电工技术委员会（ISO／IECJTCI）为专用ISDN电信网颁布的全球标准，最初由ECMA（欧洲计算机制造业者协会）提出，后来被ETSI和ISO收入。QSIG协议不分网络侧和用户侧，进行通信的设备在协议上是对等的。ECMA定义的专用ISDN网络的QSIG协议栈所示
Layers 4 to 7
Application mechanisms
End-to-end protocols
Network transparent
Layer 3
Multiple ECMA standards
Standards for Supplementary Services and Advance Network Features
ECMA-165
QSIG Generic Functional Procedures
ECMA-142/143
QSIG Basic Call
Layer 2
ECMA-141
Interface-dependent protocols
Layer 1
I.430 / I.431
PRI and BRI
QSIG协议栈我们可以看出，QSIG协议实际上是ECMA为ISDN制定的网络层协议，在协议栈中与Q931协议处于对等的位置。与Q.931不同的是，QSIG协议在ISDN第三层上又划分了三个子层：第一个子层为基本呼叫层Basic Call（BC），第二个子层为普通功能层Generic Functional（GF），第三个子层为补充业务层Supplementary Services（SS）。至于第二层的ECMA-141协议，也可以采用ITU-T的Q.921建议描述的LAPD协议的.
QSIG 的基本呼叫功能和普通业务功能是基于ITU-T 建议Q.93x 系列定义，补充业务功能基于ITU-T的建议Q.95x系列，所以QSIG可以确保公用ISDN网络和专用ISDN网络之间的兼容服务。
QSIG协议的基本呼叫过程和Q.931一样，仅是在消息定义方面比Q.931少了几个消息。QSIG协议的基本呼叫过程如图所示。"
请教高手！
谢谢！
2007-8-27 09:19 #1
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summer
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PRI信令：又称ISDN（30B+D）信令、DSS1信令、PRA信令。在北美和日本，ISDN PRI提供23B+D信道，总速率达1.544Mbps，
其中D信道速率为64kbps。而在欧洲、澳大利亚等国家，ISDN PRI为30B+D，总速率达2.048Mbps。我国为30B+D方式。
ITU-T的I.412建议中规定了两种用户-网络接口结构：基本速率接口（BRI）和基群速率接口（PRI）。ISDN的接口通过时分复用技术，把一个物理接口划分为多个信道（时隙）来使用。ISDN的信道分为B、D两种类型，其中：B信道为用户信道，用来传送数据、话音、图像等用户信息，速率是64kbit/s；D信道为控制信道，用来传送公共信道信令，控制同一接口的B信道上的呼叫，速率是64kbit/s或16kbit/s。正是这样通过B通道和D通道的划分，ISDN接口实现了数据和控制流的分离。
在ITU-T规定的ISDN标准U-N接口中，BRI接口为2B+D，TS0（16kbit/s）为D信道，TS1、TS2（64kbit/s）分别作为两个B信道B1和B2；PRI接口分E1 PRI和T1 PRI两种，B、D信道的带宽均为64kbit/s。E1 PRI为30B+D，分TS0~TS31共32个时隙，TS0用于帧同步，TS16为D信道，一般在中国、亚洲部分国家和地区、欧洲等地使用；T1 PRI为23B+D，分TS0~TS23共24个时隙，TS23为D信道，一般在北美（北美把T1 PRI接口定义为PRA）、加拿大、日本、香港等地使用。
在BRI接口方面，由于各国的用户线特性有差异，各国使用的线路码型有所不同，北美、中国采用2B1Q码，日本、意大利采用AMI码，英国采用3B2T码。ITU-T的 I.430建议对BRI接口电气特性的各项指标均作了详细的规定。BRI接口最常见的配置是用户可以将话机、传真机和数据终端接在一对用户线上，使用户可以同时利用一对电话线通话、传送或接收传真而又进行数据通信。
在PRI接口方面，E1 PRI一般使用AMI（Alternate Mark Inversion)码和HDB3码（High Density Bipolar code with a maximum of 3 zeros)两种线路码型，T1 PRI一般使用AMI、B8ZS两种线路码型。PRI接口主要应用于：（1）数字程控交换（30个64k的B信道接入）+窄带上网业务（128k带宽）；（2）商业机构总部与各分部之间的信息接入直通道；（3）大型企业之间使用专用的会议电视设备，捆绑使用6个B信道（384k）可实现图像实时传送的会议电视业务、捆绑使用2个B信道（128k）可实现图像实时传送的可视电话业务。
2007-8-28 03:09 #2
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根据OSI参考模型，ISDN用户－网络接口协议分为三层：第一层是物理层，它包括基本接入接口和一次群速率接口；第二层是数据链路层；第三层是网络层。ISDN用户-网络接口链路层协议称为LAPD(Link Access Procedure on the D channel: D通路链路接入协议)。
ITU-T的建议Q.933定义在ISDN用户－网络接口上的第二层和第三层协议称为1号数字用户信令(DSS1)。DSS1的全称是Digital subscriber signalling system No.1，是ISDN D信道上的协议。
在ISDN用户－网络接口的T参考点上，第一层采用ITU-T的建议I.430（BRI）和建议I.431（PRI），第二层采用ITU-T的建议Q.921描述的LAPD协议，第三层采用ITU-T的Q.931建议。协议栈如图3所示
2007-8-28 10:23 #3
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Q.921协议
终端端点标示符TEI（Terminate endpoint identifier）被用来标识ISDN用户侧的一个指定的连接端点。TEI值可以由网络侧分配，也可以是固定使用的。一般情况下，都是采用由网络侧分配的方式。网络侧可以分配给用户侧的TEI值为64～126（0～63为非自动分配的TEI值，127用于广播数据链路连接）。
Q.921的协商过程可以分为分配TEI、建立多帧操作（建链）、维护链路、拆除链路四步，因此，ISDN第二层有如下3个基本状态：TEI未分配、TEI已分配、多帧操作已建立
QSIG协议
QSIG也是ISDN的D信道上的协议，它是专用ISDN网络PBX直接互通的协议，是由国际标准化组织／国际电工技术委员会（ISO／IECJTCI）为专用ISDN电信网颁布的全球标准，最初由ECMA（欧洲计算机制造业者协会）提出，后来被ETSI和ISO收入。QSIG协议不分网络侧和用户侧，进行通信的设备在协议上是对等的。ECMA定义的专用ISDN网络的QSIG协议栈所示
Layers 4 to 7
Application mechanisms
End-to-end protocols
Network transparent
Layer 3
Multiple ECMA standards
Standards for Supplementary Services and Advance Network Features
ECMA-165
QSIG Generic Functional Procedures
ECMA-142/143
QSIG Basic Call
Layer 2
ECMA-141
Interface-dependent protocols
Layer 1
I.430 / I.431
PRI and BRI
QSIG协议栈我们可以看出，QSIG协议实际上是ECMA为ISDN制定的网络层协议，在协议栈中与Q931协议处于对等的位置。与Q.931不同的是，QSIG协议在ISDN第三层上又划分了三个子层：第一个子层为基本呼叫层Basic Call（BC），第二个子层为普通功能层Generic Functional（GF），第三个子层为补充业务层Supplementary Services（SS）。至于第二层的ECMA-141协议，也可以采用ITU-T的Q.921建议描述的LAPD协议的.
QSIG 的基本呼叫功能和普通业务功能是基于ITU-T 建议Q.93x 系列定义，补充业务功能基于ITU-T的建议Q.95x系列，所以QSIG可以确保公用ISDN网络和专用ISDN网络之间的兼容服务。
QSIG协议的基本呼叫过程和Q.931一样，仅是在消息定义方面比Q.931少了几个消息。QSIG协议的基本呼叫过程如图所示。"
Q信令和PRI、DSS1、PRA的关系与区别？ - 提问讨论区 - 电路交换 - 中国通信网－通信资源 咨询 人才 培训分享 通信技术|通信资源|电信技术|通信资料| 通信培训|移动通信|通信下载|通信培训|通信人才
Q信令和PRI、DSS1、PRA的关系与区别？ - 提问讨论区 - 电路交换 - 中国通信网－通信资源 咨询 人才 培训分享 通信技术|通信资源|电信技术|通信资料| 通信培训|移动通信|通信下载|通信培训|通信人才: "PRI信令
PRI信令：又称ISDN（30B+D）信令、DSS1信令、PRA信令。在北美和日本，ISDN PRI提供23B+D信道，总速率达1.544Mbps，
其中D信道速率为64kbps。而在欧洲、澳大利亚等国家，ISDN PRI为30B+D，总速率达2.048Mbps。我国为30B+D方式。
ITU-T的I.412建议中规定了两种用户-网络接口结构：基本速率接口（BRI）和基群速率接口（PRI）。ISDN的接口通过时分复用技术，把一个物理接口划分为多个信道（时隙）来使用。ISDN的信道分为B、D两种类型，其中：B信道为用户信道，用来传送数据、话音、图像等用户信息，速率是64kbit/s；D信道为控制信道，用来传送公共信道信令，控制同一接口的B信道上的呼叫，速率是64kbit/s或16kbit/s。正是这样通过B通道和D通道的划分，ISDN接口实现了数据和控制流的分离。
在ITU-T规定的ISDN标准U-N接口中，BRI接口为2B+D，TS0（16kbit/s）为D信道，TS1、TS2（64kbit/s）分别作为两个B信道B1和B2；PRI接口分E1 PRI和T1 PRI两种，B、D信道的带宽均为64kbit/s。E1 PRI为30B+D，分TS0~TS31共32个时隙，TS0用于帧同步，TS16为D信道，一般在中国、亚洲部分国家和地区、欧洲等地使用；T1 PRI为23B+D，分TS0~TS23共24个时隙，TS23为D信道，一般在北美（北美把T1 PRI接口定义为PRA）、加拿大、日本、香港等地使用。
在BRI接口方面，由于各国的用户线特性有差异，各国使用的线路码型有所不同，北美、中国采用2B1Q码，日本、意大利采用AMI码，英国采用3B2T码。ITU-T的 I.430建议对BRI接口电气特性的各项指标均作了详细的规定。BRI接口最常见的配置是用户可以将话机、传真机和数据终端接在一对用户线上，使用户可以同时利用一对电话线通话、传送或接收传真而又进行数据通信。
在PRI接口方面，E1 PRI一般使用AMI（Alternate Mark Inversion)码和HDB3码（High Density Bipolar code with a maximum of 3 zeros)两种线路码型，T1 PRI一般使用AMI、B8ZS两种线路码型。PRI接口主要应用于：（1）数字程控交换（30个64k的B信道接入）+窄带上网业务（128k带宽）；（2）商业机构总部与各分部之间的信息接入直通道；（3）大型企业之间使用专用的会议电视设备，捆绑使用6个B信道（384k）可实现图像实时传送的会议电视业务、捆绑使用2个B信道（128k）可实现图像实时传送的可视电话业务。"
PRI信令：又称ISDN（30B+D）信令、DSS1信令、PRA信令。在北美和日本，ISDN PRI提供23B+D信道，总速率达1.544Mbps，
其中D信道速率为64kbps。而在欧洲、澳大利亚等国家，ISDN PRI为30B+D，总速率达2.048Mbps。我国为30B+D方式。
ITU-T的I.412建议中规定了两种用户-网络接口结构：基本速率接口（BRI）和基群速率接口（PRI）。ISDN的接口通过时分复用技术，把一个物理接口划分为多个信道（时隙）来使用。ISDN的信道分为B、D两种类型，其中：B信道为用户信道，用来传送数据、话音、图像等用户信息，速率是64kbit/s；D信道为控制信道，用来传送公共信道信令，控制同一接口的B信道上的呼叫，速率是64kbit/s或16kbit/s。正是这样通过B通道和D通道的划分，ISDN接口实现了数据和控制流的分离。
在ITU-T规定的ISDN标准U-N接口中，BRI接口为2B+D，TS0（16kbit/s）为D信道，TS1、TS2（64kbit/s）分别作为两个B信道B1和B2；PRI接口分E1 PRI和T1 PRI两种，B、D信道的带宽均为64kbit/s。E1 PRI为30B+D，分TS0~TS31共32个时隙，TS0用于帧同步，TS16为D信道，一般在中国、亚洲部分国家和地区、欧洲等地使用；T1 PRI为23B+D，分TS0~TS23共24个时隙，TS23为D信道，一般在北美（北美把T1 PRI接口定义为PRA）、加拿大、日本、香港等地使用。
在BRI接口方面，由于各国的用户线特性有差异，各国使用的线路码型有所不同，北美、中国采用2B1Q码，日本、意大利采用AMI码，英国采用3B2T码。ITU-T的 I.430建议对BRI接口电气特性的各项指标均作了详细的规定。BRI接口最常见的配置是用户可以将话机、传真机和数据终端接在一对用户线上，使用户可以同时利用一对电话线通话、传送或接收传真而又进行数据通信。
在PRI接口方面，E1 PRI一般使用AMI（Alternate Mark Inversion)码和HDB3码（High Density Bipolar code with a maximum of 3 zeros)两种线路码型，T1 PRI一般使用AMI、B8ZS两种线路码型。PRI接口主要应用于：（1）数字程控交换（30个64k的B信道接入）+窄带上网业务（128k带宽）；（2）商业机构总部与各分部之间的信息接入直通道；（3）大型企业之间使用专用的会议电视设备，捆绑使用6个B信道（384k）可实现图像实时传送的会议电视业务、捆绑使用2个B信道（128k）可实现图像实时传送的可视电话业务。"
[问题]什么时候用1号信令什么时候用7号信令呢?(页 1) - 核心网 - 通信人家园 论坛|中国第一通信社区 - Powered by Discuz! Archiver
[问题]什么时候用1号信令什么时候用7号信令呢?(页 1) - 核心网 - 通信人家园 论坛|中国第一通信社区 - Powered by Discuz! Archiver: "通信人家园论坛C114中国通信网
通信人家园 » 核心网 » [问题]什么时候用1号信令什么时候用7号信令呢?
查看完整版本: [问题]什么时候用1号信令什么时候用7号信令呢?
sandy_gz 2006-7-5 10:07
[问题]什么时候用1号信令什么时候用7号信令呢?
是可选的吗?如何依据情况选择?
查看完整版本: [问题]什么时候用1号信令什么时候用7号信令呢?
刚果金 2006-7-5 12:17
７号需要交换机的链路板支持，１号需要ＭＦＣ支持，首先看资源，然后有个不太恰当的比喻：１和７相当于ＧＳＭ和３Ｇ的差别．
switcher 2006-7-5 17:47
可用7号就用7号，没办法才用1号。
woolau 2006-7-5 21:53
顶一个
雪落有声 2006-7-8 23:54
我的理解是，
一号信令：30个话路15出15入，就是说15个出的话路和15个入的话路不能互用
七号信令：30个话路可自由调配出入，每个话路都可能做出或入。
具体用那种信令与组网时的设置有关，需要双方交换机采用同样的信令
不对的地方，请各位批评指正
刚果金 2006-7-9 17:35
[quote][b]以下是引用[i]雪落有声在2006-7-8 23:54:00[/i]的发言：[/b]
我的理解是，
一号信令：30个话路15出15入，就是说15个出的话路和15个入的话路不能互用
七号信令：30个话路可自由调配出入，每个话路都可能做出或入。
具体用那种信令与组网时的设置有关，需要双方交换机采用同样的信令
不对的地方，请各位批评指正
[/quote]关于１号的说法是错误的，没有规定必须１５出１５入，任意调配，甚至可以是双向的（但是有碰头的弊病）．
１和７的主要区别是信令通道的区别，前者是随路信令，后者是共路信令，因此后者有很多优势，学过信令的应该很清楚．
mcg 2006-7-10 17:03
不懂
mlc1072ttd 2006-7-12 15:46
6楼的解释我比较赞同。
calljiji 2006-7-13 15:40
是不是用户线用1号呢,pcm30/32的ts16,而局间就是7号,就不是32字节咯
myswitch 2006-7-14 10:32
什么时候用1号信令什么时候用7号信令呢?是可选的吗?如何依据情况选择?
对于设备来说,什么信令都支持,只要具备相应的软硬件,是可选的.但是具体使用什么信令,需要依据所对接的设备来确定,假如你所对接的设备很老,无法进行改造,它只支持模拟信令,则你也只有使用1号信令或其他模拟信令;反之则无所谓.关键是双方协定.
woolau 2006-7-15 21:17
支持6楼解释
sandy_gz 2006-7-17 16:42
谢谢各位.
ybmgsm 2006-7-29 11:14
现在一号信令主要用于电信，移动一般都用七号信令
sunl71 2006-7-31 15:41
楼上的，谁告诉你电信用一号？电信买那么多STP是用来摆设的啊？
海朗朗 2006-8-2 12:14
13楼明显是移动的卡，带有种族歧视成分！
目前的运营商交换机都支持大部分的信令与协议，并且在自己本网内都采用七号信令，采用MFC方式大部分在市话端局或是极个别的国际局。例如：市话端局与某些小交换机用户之间就采用MFC或PRI方式。至于时隙的使用更是根据具体情况而定，并非一定要前15路出，后15路入。
sxsxxgcsjybc 2006-8-10 15:17
同意15楼
R2信令就是1号信令吧？
myswitch 2006-8-10 16:42
R2信令是1号信令的“父亲”。
mscbschlr 2006-9-24 11:37
6楼好强
peter1028 2006-9-24 21:38
又学到东西了
qiaozhi 2006-9-25 23:14
1，1号是随路信令，7号是共路信令；
2，1号：一个PCM线路对应30个话路，16时隙中的线路信令与话路是相对应的；用户号码
是带内传送的，由记发器进行收发；
7号系统的信令链路与话路逻辑上是分离的，不管是线路信令还是号码信息都通过消息
传送；
3，1号能传送信令内容较少，
7号信令内容丰富；系统较大，分层，分模块很清楚；
[并非原创，只是学习交流]
gyp0196 2006-10-12 11:34
谢谢大家讨论！
tomliang3 2006-11-17 11:07
20 楼的转贴不错
君莫笑 2006-11-17 12:48
呵呵，小弟又学到东西啦
hotplate 2007-1-9 17:52
如果大家需要不同信令之间的匹配，请联系我，我们提供专业设备。
hotplate 2007-1-9 17:53
如果大家需要不同信令之间的匹配，请联系我，我们提供专业设备。
anzi250 2007-1-11 00:54
两个差别很大的,比如说接续速度,对新业务的支持等,现在地方上基本都是7号了.
两者都是局间信令.
不是单方面确定需要使用什么信令的,需要和对端局配合
等待--- 2007-1-12 15:50
学习拉
coffee-5566 2007-2-9 13:59
<p>又学到了。。。</p><p>20楼挺强的哦。。。。。</p><p>支持转贴！！</p>
icerain2005 2007-2-14 15:58
6楼和20楼说的很清楚...不会的话建议先去看看书先
xiong19811209 2007-2-23 10:36
在做一号中继群的时候，在群向的选择中，好象只有入中继和出中继这两个选择，因此一般３０个时隙中１５个做入中继，１５个做出继。而在做７号的时候，就注意主控和非主控
╃→叼嗻煙ヤ 2007-3-26 10:54
阅
haolong153 2007-12-19 10:40
6楼正解
opqning 2008-1-23 21:55
<p>学习学习，又学一点点</p>
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查看完整版本: [问题]什么时候用1号信令什么时候用7号信令呢?
Copyright © 2002 - 2008 通信人家园"
通信人家园 » 核心网 » [问题]什么时候用1号信令什么时候用7号信令呢?
查看完整版本: [问题]什么时候用1号信令什么时候用7号信令呢?
sandy_gz 2006-7-5 10:07
[问题]什么时候用1号信令什么时候用7号信令呢?
是可选的吗?如何依据情况选择?
查看完整版本: [问题]什么时候用1号信令什么时候用7号信令呢?
刚果金 2006-7-5 12:17
７号需要交换机的链路板支持，１号需要ＭＦＣ支持，首先看资源，然后有个不太恰当的比喻：１和７相当于ＧＳＭ和３Ｇ的差别．
switcher 2006-7-5 17:47
可用7号就用7号，没办法才用1号。
woolau 2006-7-5 21:53
顶一个
雪落有声 2006-7-8 23:54
我的理解是，
一号信令：30个话路15出15入，就是说15个出的话路和15个入的话路不能互用
七号信令：30个话路可自由调配出入，每个话路都可能做出或入。
具体用那种信令与组网时的设置有关，需要双方交换机采用同样的信令
不对的地方，请各位批评指正
刚果金 2006-7-9 17:35
[quote][b]以下是引用[i]雪落有声在2006-7-8 23:54:00[/i]的发言：[/b]
我的理解是，
一号信令：30个话路15出15入，就是说15个出的话路和15个入的话路不能互用
七号信令：30个话路可自由调配出入，每个话路都可能做出或入。
具体用那种信令与组网时的设置有关，需要双方交换机采用同样的信令
不对的地方，请各位批评指正
[/quote]关于１号的说法是错误的，没有规定必须１５出１５入，任意调配，甚至可以是双向的（但是有碰头的弊病）．
１和７的主要区别是信令通道的区别，前者是随路信令，后者是共路信令，因此后者有很多优势，学过信令的应该很清楚．
mcg 2006-7-10 17:03
不懂
mlc1072ttd 2006-7-12 15:46
6楼的解释我比较赞同。
calljiji 2006-7-13 15:40
是不是用户线用1号呢,pcm30/32的ts16,而局间就是7号,就不是32字节咯
myswitch 2006-7-14 10:32
什么时候用1号信令什么时候用7号信令呢?是可选的吗?如何依据情况选择?
对于设备来说,什么信令都支持,只要具备相应的软硬件,是可选的.但是具体使用什么信令,需要依据所对接的设备来确定,假如你所对接的设备很老,无法进行改造,它只支持模拟信令,则你也只有使用1号信令或其他模拟信令;反之则无所谓.关键是双方协定.
woolau 2006-7-15 21:17
支持6楼解释
sandy_gz 2006-7-17 16:42
谢谢各位.
ybmgsm 2006-7-29 11:14
现在一号信令主要用于电信，移动一般都用七号信令
sunl71 2006-7-31 15:41
楼上的，谁告诉你电信用一号？电信买那么多STP是用来摆设的啊？
海朗朗 2006-8-2 12:14
13楼明显是移动的卡，带有种族歧视成分！
目前的运营商交换机都支持大部分的信令与协议，并且在自己本网内都采用七号信令，采用MFC方式大部分在市话端局或是极个别的国际局。例如：市话端局与某些小交换机用户之间就采用MFC或PRI方式。至于时隙的使用更是根据具体情况而定，并非一定要前15路出，后15路入。
sxsxxgcsjybc 2006-8-10 15:17
同意15楼
R2信令就是1号信令吧？
myswitch 2006-8-10 16:42
R2信令是1号信令的“父亲”。
mscbschlr 2006-9-24 11:37
6楼好强
peter1028 2006-9-24 21:38
又学到东西了
qiaozhi 2006-9-25 23:14
1，1号是随路信令，7号是共路信令；
2，1号：一个PCM线路对应30个话路，16时隙中的线路信令与话路是相对应的；用户号码
是带内传送的，由记发器进行收发；
7号系统的信令链路与话路逻辑上是分离的，不管是线路信令还是号码信息都通过消息
传送；
3，1号能传送信令内容较少，
7号信令内容丰富；系统较大，分层，分模块很清楚；
[并非原创，只是学习交流]
gyp0196 2006-10-12 11:34
谢谢大家讨论！
tomliang3 2006-11-17 11:07
20 楼的转贴不错
君莫笑 2006-11-17 12:48
呵呵，小弟又学到东西啦
hotplate 2007-1-9 17:52
如果大家需要不同信令之间的匹配，请联系我，我们提供专业设备。
hotplate 2007-1-9 17:53
如果大家需要不同信令之间的匹配，请联系我，我们提供专业设备。
anzi250 2007-1-11 00:54
两个差别很大的,比如说接续速度,对新业务的支持等,现在地方上基本都是7号了.
两者都是局间信令.
不是单方面确定需要使用什么信令的,需要和对端局配合
等待--- 2007-1-12 15:50
学习拉
coffee-5566 2007-2-9 13:59
<p>又学到了。。。</p><p>20楼挺强的哦。。。。。</p><p>支持转贴！！</p>
icerain2005 2007-2-14 15:58
6楼和20楼说的很清楚...不会的话建议先去看看书先
xiong19811209 2007-2-23 10:36
在做一号中继群的时候，在群向的选择中，好象只有入中继和出中继这两个选择，因此一般３０个时隙中１５个做入中继，１５个做出继。而在做７号的时候，就注意主控和非主控
╃→叼嗻煙ヤ 2007-3-26 10:54
阅
haolong153 2007-12-19 10:40
6楼正解
opqning 2008-1-23 21:55
<p>学习学习，又学一点点</p>
页: [1]
查看完整版本: [问题]什么时候用1号信令什么时候用7号信令呢?
Copyright © 2002 - 2008 通信人家园"
ISDN拨号（一号信令）(2)_IT认证_思科认证_GZU521.COM学习网
ISDN拨号（一号信令）(2)_IT认证_思科认证_GZU521.COM学习网: "您的位置： 学习网(Gzu521.com) -> 考试资料 -> IT认证 -> 思科认证 -> 资料正文
学习网考试学习资料
Gzu521.com
ISDN拨号（一号信令）(2)
思科认证 点击：39次 发布时间：2006-8-9 【字体：大 中 小】 来源：Gzu521.com
-
Gzu521.com我的学习网
ppp authentication chap pap
group-range 161 190
!
interface group-async1
ip unnumbered fastethernet4/0
encapsulation ppp
ip tcp header-compression passive
async mode interactive
peer default ip address pool setup_pool
ppp authentication chap pap
group-range 193 222
!
ip local pool setup_pool 10.164.199.100 10.164.199.220
ip classless
no ip http server
!
!
line con 0
transport input none
line 161 190
autoselect during-login
autoselect ppp
login local
modem inout
modem autoconfigure discovery
transport input all
flowcontrol hardware
line 193 222
autoselect during-login
autoselect ppp
login local
modem inout
modem autoconfigure discovery
transport input all
flowcontrol hardware
line aux 0
line vty 0 4
password **********
login
!
end"
学习网考试学习资料
Gzu521.com
ISDN拨号（一号信令）(2)
思科认证 点击：39次 发布时间：2006-8-9 【字体：大 中 小】 来源：Gzu521.com
-
Gzu521.com我的学习网
ppp authentication chap pap
group-range 161 190
!
interface group-async1
ip unnumbered fastethernet4/0
encapsulation ppp
ip tcp header-compression passive
async mode interactive
peer default ip address pool setup_pool
ppp authentication chap pap
group-range 193 222
!
ip local pool setup_pool 10.164.199.100 10.164.199.220
ip classless
no ip http server
!
!
line con 0
transport input none
line 161 190
autoselect during-login
autoselect ppp
login local
modem inout
modem autoconfigure discovery
transport input all
flowcontrol hardware
line 193 222
autoselect during-login
autoselect ppp
login local
modem inout
modem autoconfigure discovery
transport input all
flowcontrol hardware
line aux 0
line vty 0 4
password **********
login
!
end"
ISDN拨号（一号信令）(1) Gzu521.com我的学习网
ISDN拨号（一号信令）(1) Gzu521.com我的学习网: "CISCO 36和朗讯程控交换机连接：
!
version 12.1
no service single-slot-reload-enable
service timestamps debug uptime
service timestamps log uptime
no service password-encryption
!
hostname router
!
logging rate-limit console 10 except errors
enable password *********
!
username test password 0 test
ip subnet-zero
!
!
no ip finger
!
!
!
controller e1 3/0
framing no-crc4
ds0-group 1 timeslots 1-15,17-31 type r2-digital r2-compelled
cas-custom 1
unused-abcd 0 1 1 1
country china use-defaults
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!
controller e1 3/1
framing no-crc4
ds0-group 1 timeslots 1-15,17-31 type r2-digital r2-compelled
cas-custom 1
unused-abcd 0 1 1 1
country china use-defaults
answer-signal group-b 1
!
controller e1 4/0
framing no-crc4
ds0-group 1 timeslots 1-15,17-31 type r2-digital r2-compelled
cas-custom 1
unused-abcd 0 1 1 1
country china use-defaults
answer-signal group-b 1
!
controller e1 4/1
framing no-crc4
ds0-group 1 timeslots 1-15,17-31 type r2-digital r2-compelled
cas-custom 1
unused-abcd 0 1 1 1
country china use-defaults
answer-signal group-b 1
!
!
interface loopback0
ip address 192.168.0.1 255.255.255.0
!
interface fastethernet0/0
no ip address
shutdown
duplex auto
speed auto
!
interface fastethernet3/0
no ip address
shutdown
duplex auto
speed auto
!
interface fastethernet4/0
ip address 10.164.199.221 255.255.255.0
duplex auto
speed auto
!
interface group-async0
ip unnumbered fastethernet4/0
encapsulation ppp
ip tcp header-compression passive
async mode interactive
peer default ip address pool setup_pool"
!
version 12.1
no service single-slot-reload-enable
service timestamps debug uptime
service timestamps log uptime
no service password-encryption
!
hostname router
!
logging rate-limit console 10 except errors
enable password *********
!
username test password 0 test
ip subnet-zero
!
!
no ip finger
!
!
!
controller e1 3/0
framing no-crc4
ds0-group 1 timeslots 1-15,17-31 type r2-digital r2-compelled
cas-custom 1
unused-abcd 0 1 1 1
country china use-defaults
answer-signal group-b 1
!
controller e1 3/1
framing no-crc4
ds0-group 1 timeslots 1-15,17-31 type r2-digital r2-compelled
cas-custom 1
unused-abcd 0 1 1 1
country china use-defaults
answer-signal group-b 1
!
controller e1 4/0
framing no-crc4
ds0-group 1 timeslots 1-15,17-31 type r2-digital r2-compelled
cas-custom 1
unused-abcd 0 1 1 1
country china use-defaults
answer-signal group-b 1
!
controller e1 4/1
framing no-crc4
ds0-group 1 timeslots 1-15,17-31 type r2-digital r2-compelled
cas-custom 1
unused-abcd 0 1 1 1
country china use-defaults
answer-signal group-b 1
!
!
interface loopback0
ip address 192.168.0.1 255.255.255.0
!
interface fastethernet0/0
no ip address
shutdown
duplex auto
speed auto
!
interface fastethernet3/0
no ip address
shutdown
duplex auto
speed auto
!
interface fastethernet4/0
ip address 10.164.199.221 255.255.255.0
duplex auto
speed auto
!
interface group-async0
ip unnumbered fastethernet4/0
encapsulation ppp
ip tcp header-compression passive
async mode interactive
peer default ip address pool setup_pool"
Understanding Basic VoIP with PBX Trunking
emUnderstanding Basic VoIP with PBX Trunking: "The voice network module on the Cisco 3600 series supports up to two voice interface cards (VICs). These VICs come in three types, each having two ports. Each type provides a slightly different interface for connecting to different types of equipment. The basic types of voice interfaces are: foreign exchange service (FXS), foreign exchange office (FXO), and E&M.
*
The FXO interface is an RJ-11 connector that connects local calls to a public switched telephone network (PSTN) central office (CO), or to a PBX that does not support E&M signaling. This is the interface a standard telephone provides. This is the only voice interface card approved to connect to off-premise lines, and this is its primary use. This interface may be used to provide backup over the PSTN or for Centrex-type operations. The FXO ports on the Cisco 3600 series router can support both loop-start and ground-start modes. (In certain situations, a hardware jumper must be used to make an FXO port operate in ground-start mode.) The ground-start signaling method is used primarily on trunk lines or tie lines between private branch exchanges (PBXs).
*
The FXS interface is also an RJ-11 connector that connects directly to a standard telephone, fax machine, or similar device. The FXS interface supplies ringing voltage, dial tone, and similar signals to FXO devices. As the exercises in module 1 demonstrated, you would use the FXS interface when connecting a phone directly to a router. Basically, the FXS interface mimics the PSTN. The FXS ports on the Cisco 3600 series router can provide the battery at 24-volt DC. They can also support both loop-start and ground-start modes of operation according to the software configuration of the router. Loop start, however, is the most common, because most residential telephones are analog loop-start devices.
*
E&M, which stands for 'Ear and Mouth' (or 'recEive and transMit,' or sometimes 'Earth and Magnet'), is a signaling technique for two- and four-wire telephone and trunk interfaces that is used mainly between PBXs or other network-to-network telephony switches, such as Lucent 5 Electronic Switching System [5ESS], Nortel DMS-100, and so on. The E&M analog interface is an RJ-48S connector used to connect remote calls from an IP network to a PBX for local distribution and between PBX trunk lines (tie lines) or other network-to-network telephony switches. Unlike loop-start or ground-start modes of operation on the FXS/FXO ports, E&M uses separate wires for signaling and voice. There are five types of E&M signaling, as well as two wiring methods.
Exercise care when designing a voice network...
Exercise care when designing a voice network to ensure supervisory signals (discussed in module 1 of this tutorial) are transmitted when expected to avoid common problems such as spurious disconnects, undisconnected lines, and continuous rings. For example, a typical disconnect supervision problem can occur when a given line can be released only by the party who originated the phone call. This problem can be fixed by implementing either disconnect supervision or ground start.
Click to close sidebar.
In the first module of this tutorial, you learned about the basics of analog voice internetworking, including the history of telephony, telephone networks and technology, and the various types of telephony signaling. You know how to configure and troubleshoot a Cisco 3640 router using loop-start access signaling across the FXS interface. In this module you will learn more about trunks, analog E&M signaling, and VoIP configuration using E&M. The Cisco VoIP implementation supports E&M types I, II, III, and V, using both two- and four-wire implementations.
Click on any highlighted term see its pop-up glossary definition. For additional information and more definitions of terms used throughout this course, see the full Glossary. You can access the Glossary at any time by clicking Help.
A basic understanding of trunks is necessary to understanding the use of E&M signaling. A line is a communications path between a customer's telephone and a telephone switch, such as a CO switch or a PBX, whereas a trunk is an actual telephone circuit or path between two switches, at least one of which is usually a CO or a switching center. This type of trunk, where voice is carried over a telco's network, is a CO trunk. Another example of a trunk is a private leased line that interconnects two PBXs, forming a corporate voice network. This type of connection is referred to as a PBX trunk.
Central-Office Trunks
Two-Wire Central-Office Trunks
A hunt group is...
A hunt group is a feature supported by voice-capable Cisco routers that involves the configuration of a group of dial peers on the same router with the same destination pattern. With a hunt group, if a call attempt is made to a dial peer on a specific time slot and that time slot is busy, the router hunts for another time slot on that channel until an available time slot is found.
The Cisco 3640 hunts for a group based on matching numbers.
It is possible to configure hunt groups by setting up multiple dial peers having the same destination pattern. Hunt groups are not configured in this tutorial, and the feature is not enabled in the simulation exercises in this module.
Click to close sidebar.
The most common two-wire CO trunks used in PBX networks employ ground start. These trunks may be restricted such that operation may be outgoing only, incoming only, or both. In the case of outgoing calls (direct outward dialing, or DOD) operation additionally may be restricted to local calling only, versus direct distance dialing (DDD). Most incoming calls are routed to an attendant or to a messaging or answering system. The ringback tone is generated from the local exchange carrier's (LEC's) switch that connects to the PBX. This type of trunk is common.
Most of these trunks are assigned the same phone number within the PSTN, and operate in a 'hunt group' fashion.
Direct Inward Dialing Central Office Trunks
Direct inward dialing (DID) is used to determine how the called number is treated for incoming calls. Incoming calls to a PBX often first flow through an attendant position. DID trunks allow users to receive calls directly from the outside without intervention from the attendant.
Use of DID offers three main advantages:
* It allows direct access to stations from outside the PBX.
* It allows users to receive calls even when the attendant switchboard is closed.
* It takes a portion of the load off the attendants.
To accomplish DID on a trunk, the switch at the PSTN (either the local exchange carrier [LEC] or the inter-exchange carriers [IXC]) transmits the number of the called party to the PBX for routing. DID trunks often use existing two-wire circuits, and typically use ground-start signaling. Four-wire circuits using E&M signaling or digital access are sometimes used as well. To use DID, a block of PSTN telephone numbers generally must be reserved from the IXC or LEC.
DID trunk circuits usually employ either Wink Start or Delay-Dial signaling schemes, so that the network knows that the PBX is ready to accept the incoming address supervision. Wink Start and Delay Dial are discussed in more detail later in this section.
The next module in this tutorial, 'Basic Analog-to-Digital Voice over IP,' covers the use of DID in a VoIP configuration.
Wide-Area Telephone Service Central-Office Trunks
Wide-Area Telephone Service (WATS—sometimes referred to as OUTWATS) trunks are used for outgoing calls. They take advantage of interconnection to the IXC. These trunks require that the PBX output digits to the PSTN switch for routing. The network usually provides answer and disconnect supervision. WATS trunks can operate using either ground-start or loop-start signaling, in addition to E&M signaling. Ringback and busy signals are generated from the remote switch or PBX.
WATS trunks do not have addressable PSTN telephone numbers; therefore, these trunks are typically implemented in a hunt-group fashion within the PBX.
Central-Office 800-Service Trunks
Trunks that provide 800-service are sometimes referred to as INWATS service trunks. These trunks are used for incoming calls and take advantage of interconnection to the IXC. With an 800-service trunk, the PSTN does not output digits to the PBX; incoming calls on these trunk numbers are routed to specific individuals or groups by the PBX. These trunks may utilize loop reverse battery or E&M signaling.
Ringback and busy tones are usually generated by the PSTN switch that is interconnected to the PBX. Generally, these trunks are implemented in a hunt group fashion.
Long-Distance Trunks
Long-distance trunks interconnect to an IXC to allow both incoming and outgoing long-distance calling. They usually operate on two-wire ground-start circuits, or two- or four-wire E&M circuits. If access to the PSTN switch is digital (DS1), then conventional robbed-bit signaling (RBS) is used. (RBS is explained in greater detail in the next module of the Voice Internetworking CIM tutorial.)
Ringback and busy tones are usually generated by the IXC's PSTN switch that is interconnected to the PBX. Generally, these types of trunks are implemented in a hunt-group fashion, and incoming calls are typically routed to an attendant or messaging system.
PBX Trunks
A PBX is a private digital or analog telephone switchboard used to originate and answer calls to and from the public network (via CO trunks). PBXs can be combined with PBX trunks between them. When a caller picks up a handset and dials from within the corporation, the PBX connects that user to an idle line or to an idle trunk in an appropriate trunk group, then returns the appropriate call status signal, such as a dial tone or a ringback. A busy or fast-busy signal is returned if the line or the trunk group is busy. An attendant may be present to answer incoming calls and for user assistance.
FX Trunks
Foreign exchange (FX) circuits may consist of either stations or trunks. An FX line is a special line that is run from a local telephone to a CO switch or PBX switch. In this case, the local telephone is assigned a number on a remote switch; for all inbound and outbound calls, the telephone behaves as if it were connected to the remote switch. Generally, an FX circuit uses a two-wire circuit with loop start signaling. The ringback and busy signals are provided by the switch connected to the CO.
An FX trunk operates in a similar fashion, except that a local trunk provides interconnectivity to a remote switch. For outbound calls, local PBX users dial an access code such as 9, receive a dial tone from the remote switch, then output digits to the remote switch for routing. Incoming calls are either barred, or are routed to a PBX attendant or message system.
Because a trunk circuit is involved, features such as call transferring and call forwarding can also be deployed. These types of activities requires disconnect supervision to release the PBX trunk. This supervision cannot be provided with loop-start signaling, so ground start signaling must be used.
Tie Trunks and Tie Lines
Tie trunks interconnect PBX switches within a customer's network. Tie trunks typically connect both trunks and station lines through a network connection. These circuits are usually heavily used and normally support both incoming and outgoing calls.
Tie trunks are typically four-wire E&M-type circuits, but may often interconnect through DS1 facilities using RBS. The use of two-wire trunks is also allowable; these would typically use ground-start signaling.
Although tie lines are generally considered the same as tie trunks, and the terms are often used interchangeably, there are some technical differences:
* Tie trunks can interconnect both trunks and lines. In tie-trunk operation, routing of a call is totally automatic. The remote PBX usually supplies another dial tone, at which the user must enter the necessary address for the called party at that PBX.
* Tie lines connect only lines throughout a network connection. Routing of a call over tie lines is under control of the user. A user would need to dial an access code (9, for example) to gain access to the desired remote PBX. Tie lines are actually tie trunks that use tone-start type of start-dial supervision and cut-through operation. (Details about start-dial supervision can be found at the end of this section.)
Although the next module will include use of a PBX in the network topology, a complete discussion of PBXs and PBX configuration is out of the scope of this tutorial. To learn about PBX configuration, you should refer to the manuals for the particular PBX that you are using in your network. For additional PBX information, however, please see the related document The Private Branch Exchange (PBX) .
As discussed in module 1, 'Basic Analog Voice over IP,' the primary purpose of signaling in a voice network is to establish a connection. Signaling can be classified into four basic functions:
* Supervisory signaling — Informs the telephone or switch of the status of the local loop and any connected trunks. Supervisory signaling is used to:
o Initiate a call request on line or trunks (called line signaling on trunks).
o Indicate that the call has been answered or the call has been disconnected (disconnect and answer supervision).
o Initiate or terminate charging for calls.
o Recall an operator on an established connection.
* Address signaling — Contains information indicating the destination of a call, such as the telephone number and an area code, an access code, or a PBX tie trunk access code.
* Call progress indicators — Convey call-progress or call-failure information to subscribers or operators by the use of usually audible tones (such as the busy signals, the reorder tone, and the ringback, all of which were discussed in module 1).
* Network management signaling — Controls the bulk assignment of circuits or modifies the operating characteristics of switching systems in a network in response to overload conditions. (These signaling functions will not be covered in detail in this tutorial, but are mentioned here for completeness.)
For a more detailed discussion of signaling, including descriptions of supervision, address, call-progress, and network management signaling, read Signaling, which is included in this tutorial. Also search Cisco Connection Online (CCO) at http://www.cisco.com for more information on this specific topic as well as more available information on treansmitting voice over a data network.
Just as the local loop requires supervisory signaling, so does a trunk. Although loop-start signaling, which is generally used on everyday residential telephones, can also be used on trunks, its use can result in problems that make it an ineffective signaling method for a trunk. With loop start, only the device originating the call can release the connection. Also, loop start would not prevent a trunk from being seized by both parties to a call, a condition referred to as glare. Glare can be tolerable on a local loop, but its occurrence on a trunk would be unacceptable. Ttwo-way handshaking methods were developed to coordinate the sequence of events that occur during a call: the request by the calling end for access to the trunk, followed by the acknowledgment by the called end, then the subsequent seizure of the trunk by the calling end.
The basic types of supervisory signaling used on trunks are: ground start, E&M, and start-dial supervision.
Ground-Start Signaling
In module 1 of this course, you configured the Cisco 3640 router FXS port to use loop-start signaling, because calls were being made using a directly connected analog telephone. Ground start is the access signaling method used on trunk lines or tie lines between PBXs to indicate on-hook/off-hook status to the CO. Ground-start signaling works by using ground and current detectors. Ground start behaves like a loop start, however, the PBXs on both sides—the telco and the customer's telephone—have to agree on the status of a line before a call can be placed. This scenario allows the network to indicate off-hook or seizure of an incoming call independent of the ringing signal.
E&M Signaling Interface Types
Analog trunk circuits connect between automated systems (a PBX) and the network (a CO). The most common form of analog trunk is the E&M interface.
E&M signaling is typically used for trunk lines. It provides native support for both disconnect and answer supervision, as well as glare avoidance. In E&M signaling, separate paths are used for voice and signaling. E&M is normally the only way that a CO switch can provide two-way dialing with DID.
The history of the semantics of E&M circuits dates back to the days of telegraphy, where the main office end had a 'key' that grounded the E circuit, and the other end had a sounder with an electromagnet attached to a battery. Descriptions such as 'ear' and 'mouth' were created to help provide a reference to field personnel as to the direction of a signal in a wire. Note that these terms correspond to the inbound and outbound circuits, known as the E lead and M lead, respectively.
There are five standard types of E&M interfaces, which correspond to the five signaling types:
* E&M Type I Interface Model
* E&M Type II Interface Model
* E&M Type III Interface Model
* E&M Type IV Interface Model
* E&M Type V Interface Model
With each signaling type, the PBX supplies one signal, known as the M signal (mouth), and accepts one signal, known as the E signal (ear). Conversely, the tie-line equipment accepts the M signal from the PBX and provides the E signal to the PBX. The M signal accepted by the tie-line equipment at one end of a tie circuit becomes the E signal output by the remote tie-line interface.
Remember learning about tip and ring in module 1?
* When a user tries to place a call by grounding the 'ring' lead, the PBX at the telco senses the flow of current and grounds the 'tip' lead to indicate the PBX is ready to serve.
* The user's equipment perceives the flow of current on its 'tip' and knows that the PBX is ready to serve.
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The signaling part of the interface has five distinct physical configurations (Types I – V) and the audio interface has two (two- and four-wire). Note that the terms two-wire and four-wire are used to reference the communication protocol of the audio path. The difference between a two-wire and four-wire circuit is whether the audio path is full duplex on one pair or two pairs of wires.
Two full-duplex wires (called tip and ring) carry the audio on two-wire E&M. Four-wire E&M uses four half-duplex wires for the audio path (Tip, Ring, Tip1, and Ring1). Note that a four-wire E&M circuit may have from six to eight physical wires.
Signaling Types
* E — Ear or earth: Signal wire from the trunking (CO or network) side to signaling (user) side
* M — Mouth or magnet: Signal wire from signaling (user) side to trunking (CO or network) side
* SG — Signal ground: Used only on certain types of E&M; sometimes grounded, sometimes not>
* SB — Signal battery: Used on certain types of E&M; sometimes provides -48V Direct Current, sometimes ground, sometimes is not used at all
* T/R — Tip/ring: Used only on a four-wire circuit; carries audio from the signaling (user) side to the trunking (CO/network) side; not used on a two-wire circuit
* T1/R1 — Tip1/ring1: On four-wire circuits, carries audio from the trunking (CO/network) side to the signaling (user) side; on a two-wire circuit, this pair carries the full-duplex audio path
Types I and II are the most popular E&M signaling types in the Americas. Type V is used in the United States, and is very popular in Europe. (SSDC5 is most often found in the United Kingdom.) Similar to Type V, SSDC5A differs in that on- and off-hook states are backward to allow for fail-safe operation: if the line breaks, the interface defaults to off hook (busy). Of all the types, only Types II and V are symmetrical (can be back to back using a crossover cable).
The Cisco 2600 and 3600 series routers currently support types I, II, III, and V utilizing both two- and four-wire implementations. The E&M interface of the Cisco 3600 series presents the 'channel bank,' 'tie equipment,' or 'CO' side of a trunk interface to a PBX. Each E&M signaling type has a unique circuit model and connection diagram. All Cisco analog voice gateway routers have 24V DC designed on the FXS ports. According to EIA 464 standard, any voltage from DC 24 volts to DC 52 volts is acceptable.
Characteristics of E&M Lead Signaling
Type
M Lead
E Lead
Outbound Direction Inbound Direction
Off Hook
On Hook
Off Hook
On Hook
I
Battery
Ground
Ground
Open
II
Battery
Open
Ground
Open
III
Loop current
Ground
Ground
Open
IV
Ground
Open
Ground
Open
V
Ground
Open
Ground
Open
SSDC5
Earth on
Earth off
Earth on
Earth off
Cisco routers expect to see off-hook conditions on the M lead and signal off hook to the remote device on the E lead.
See the following sections and illustrations for individual descriptions of the different types. Each of the following illustrations shows the E&M interface of the PBX on the left, and the corresponding tie-line equipment interface on the right. The symbol V refers to battery voltage, which can be 25 VDC to 65 VDC, and is usually (nominally) -48 VDC.
E&M Type I
In E&M Type I signaling, the battery PBX provides the battery for both the E and M leads. During an off-hook condition, the PBX generates the E signal by grounding the E lead. The PBX detects the E signal by sensing the increase in current through a resistive load. Similarly, in an on-hook condition, the PBX generates the M signal by sourcing a current to the tie-line equipment, which detects it via a resistive load.
The four-wire Type I interface from the PBX has the following characteristics:
* E (pin 7) detector 'floats' at -48V below ground
* M (pin 2) contact has low ohms to ground on hook, and is -48V below ground when off hook
* Approximately 30-150 ohms between T/R (pins 6/3) sometimes in series with 2.2uf of capacitance
* Approximately 30-150 ohms between T1/R1 (pins 5/4) sometimes in series with 2.2uf of capacitance
E&M Type I
The Type I interface requires that the PBX and tie-line equipment share a common signaling ground reference. This setup can be achieved by connecting signal ground from the PBX to the signal-ground (SG) lead (pin 8) of the RJ-48S connector.
Note that the voltage on the E and M leads may not be the same (the E lead may have lower voltage). This asymmetrical signaling scheme is a potential source of interference, which could cause high return current through the grounding system. If two PBXs were not correctly grounded, current could flow down the M lead, causing the remote PBX to erroneously detect a current on the E lead, thus resulting in false seizure of a trunk. Despite this potential problem, E & M signaling is the most common four-wire trunk interface used in North America.
E&M Type II
The E&M Type II interface provides almost complete isolation of signaling power systems by the addition of two additional signaling leads: signal battery (SB) and signal ground (SG). In Type II, each of the two signals has its own return. For the E signal, the E lead works with the SSG lead to allow current to flow from the PBX while the M lead is strapped to the SB lead. This scenario results in the trunk being grounded at each end, eliminating the potential problem associated with Type I, discussed above.
The four-wire Type II interface from the PBX has the following characteristics:
* E lead (pin 7) detector 'floats' at -48V below ground
* SG lead (pin 8) has a low ohms to ground
* M lead (pin 2) contact between M and SB is open when on-hook, and closed when off hook
* M lead floats
* SB lead (pin 1) floats
* Approximately 30-150 ohms between T/R (pins 6/3) sometimes in series with 2.2uF of capacitance
* Approximately 30-150 ohms between T1/R1 (pins 5/4) sometimes in series with 2.2uF of capacitance
E&M Type II
Type II is seen only occasionally in North America, usually on Centrex trunk circuits or Nortel PBX systems.
E&M Type III
E&M Type III signaling is very similar to Type I, except that the battery and ground source for the M lead is supplied by the transmission equipment. Complete power isolation is provided with the M lead, and the facility can establish and control the amount of E lead current. Type III uses the SG lead to provide common ground. The PBX drops the M signal by grounding it, rather than by opening a current loop.
The four-wire Type III interface from the PBX has the following characteristics:
* E lead (pin 7) detector 'floats' at -48V below ground
* M lead (pin 2) contact between M and SG when on hook, and between M and SB when off hook
* SG lead (pin 8) floats
* M lead (pin 2) floats
* SB lead (pin 1) floats
* Approximately 30-150 ohms between T/R (pins 6/3) sometimes in series with 2.2uF of capacitance
* Approximately 30-150 ohms between T1/R1 (pins 5/4) sometimes in series with 2.2uf of capacitance
E&M Type III
There is no evidence that the unbalanced E lead of Type III has caused any interference problems, but one drawback of this interface is its inability to operate in a 'back-to-back' configuration.
This interface is most often used in older CO equipment such as 1/1AESS, 2/2BESS, and 3ESS switches. It is not often seen now because most of these older switches have been replaced.
E&M Type IV
Type IV is symmetric and requires no common ground. Each side closes a current loop to signal; the flow of current is detected via a resistive load to indicate the presence of the signal. The Type IV interface is similar to Type II, with the difference in the operation of the M lead—in Type II, the M lead states are 'open' and 'battery;' Type IV states are 'ground' and 'open.'
The four-wire Type IV interface from the PBX has the following characteristics:
* E lead (pin 7) detector 'floats' at -48V below ground
* SG lead (pin 8) has low ohms to ground
* M lead (pin 2) contact between M and SB is open when on-hook, and closed when off hook
* M lead (pin 2) floats
* SB lead (pin 1) floats
* Approximately 30-150 ohms between T/R (pins 6/3) sometimes in series with 2.2uF of capacitance
* Approximately 30-150 ohms between T1/R1 (pins 5/4) sometimes in series with 2.2uf of capacitance
E&M Type IV
The advantages of Type IV include:
* Accidental shorting of the SB lead (during cable wiring, for example) will not result in an excessive current flow.
* The interface can interconnect to a Type II device.
* The interface can operate in a 'back-to-back' configuration.
Because it can be difficult for an external monitor to distinguish between 'open' and 'ground' states, it can be difficult to obtain test and supporting equipment for a Type IV interface. Type IV is not currently supported by the Cisco 3600 series routers.
E&M Type V
E&M Type V interface is a simplified version of Type IV. It is also a symmetric interface, using only two wires. In the Type V interface, both the switch and the transmission equipment supply battery. The battery for the M lead is located in the signaling equipment, and the battery for the E lead is located in the PBX. Type V requires a common ground between the PBX and the tie line equipment, which is provided via the SG leads.
The Type V interface from the PBX has the following characteristics:
* E lead (pin 7) detector 'floats' at -48V below ground
* M lead (pin 2) contact ground is open when on-hook, and closed when off hook
* Approximately 30-150 ohms between T/R (pins 6/3) sometimes in series with 2.2uF of capacitance
* Approximately 30-150 ohms between T1/R1 (pins 5/4) sometimes in series with 2.2uf of capacitance
E&M Type V
Although this interface does not provide isolation between power systems, there is minimal (or no) return currents in this symmetrical signaling scheme. Type V is the most popular interface outside North America.
E&M Start-Dial Supervision Signaling Protocol
Start-dial supervision is the line protocol used between equipment that takes place after the initial off-hook condition, up to the passing of dial digits to the connected device.
Three principal protocols are used on E&M circuits:
* E&M Immediate Start Signaling
* E&M Wink-Start Signaling
* E&M Delay-Dial Signaling
Wink start is used to notify the remote side it can send dialed number identification service (DNIS) information. Wink acknowledgment is a second wink that is sent to acknowledge the receipt of the DNIS information. Immediate start does not send any winks at all.
The following sections explain how each protocol works. It is important to understand what the protocol is supposed to do when debugging, because call progress anomalies provide clues to the cause.
Immediate Start
E&M Immediate Start is the simplest of protocols. Here, the originating switch goes off hook, waits for a finite period of time (say, 200 ms) then sends the address digits without regard to the remote. Because there is no acknowledgment, or handshaking, between switches, this type of trunk signaling should be used only when there is a dedicated physical or logical trunk between switches. Cisco routers support the use of the Immediate Start protocol.
Immediate Start
Wink Start
E&M Wink Start is the most common of protocols. Wink start is an in-band technique in which the calling switch waits from 140 to 290 ms for an off-hook wink pulse from the called switch before sending the dialed digits. Wink start was developed to minimize glare, which occurs when both ends attempt to seize the trunk at the same time. If there is no wink pulse, an error condition is caused by both ends attempting to place a call on the same trunk.
* In the original Wink Start protocol, the remote responds to an off hook from the originating device with a short wink (transition from on hook to off hook and back again). This wink tells the originating device that the remote device is ready to receive digits. After receiving the addressing digits, the remote then goes off hook for the duration of the call. The originating device maintains off hook for the duration of the call.
* In Wink Start with Wink Acknowledge protocol (sometimes referred to as double wink), the remote responds to an off hook from the originating device with a short wink (transition from on hook to off hook and back again), just as in the original Wink Start. This wink tells the originating side that the remote is ready to receive digits. After receiving the addressing digits, the remote then provides another wink (called an Acknowledgment Wink) that tells the originating side that the terminating side has received the dialed digits. The remote then goes off hook to indicate connection when the ultimate called endpoint has answered. The originating device maintains off-hook status for the duration of the call.
Wink Start normally is not used on trunks that are controlled with message-oriented signaling schemes such as ISDN or Signaling System 7 (SS7). Cisco routers support both wink-start (fgb) and wink-start with wink-acknowledge or double-wink (fgd).
Wink Start
Delay Dial
In E&M Delay Dial mode, the originating device goes off hook and waits for about 200 ms, then checks to see if the remote end is on hook. If so, it then outputs addressing digits. If the remote is off hook, the calling device waits until the remote goes back on hook before transmitting digits. The delay signal says, in effect 'hold on, I'm not ready to receive digits.' This protocol was invented for use with systems that have fewer digit collectors than trunk interfaces. Delay dial is not currently supported by Cisco 3600 series routers.
Delay
Start Dial Supervision Mismatches
Although both ends of a call do not need Start Dial Supervision, mismatches can be a source of problems. Sometimes a PBX has a different Start Dial Supervision protocol for inbound and outbound calls. This setup can lead to erratic behavior if the remote is not configured to properly handle this condition. The following general rule set applies:
* An Immediate-Start interface can usually originate a call to a Wink-Start interface.
* An Immediate-Start interface can usually place a call to a Delay-Dial interface if the delay pulse is shorter than the immediate-start delay. Otherwise, operation is erratic.
* A Wink-Start interface can usually originate a call into a Delay-Dial interface if there is a delay pulse. Otherwise, the call will hang, with only a 50-percent chance of working.
* A Delay-Dial interface can, for the most part, originate a call into an Immediate-Start or Wink-Start interface.
Go on to Configuration of VoIP with E&M Signaling."
*
The FXO interface is an RJ-11 connector that connects local calls to a public switched telephone network (PSTN) central office (CO), or to a PBX that does not support E&M signaling. This is the interface a standard telephone provides. This is the only voice interface card approved to connect to off-premise lines, and this is its primary use. This interface may be used to provide backup over the PSTN or for Centrex-type operations. The FXO ports on the Cisco 3600 series router can support both loop-start and ground-start modes. (In certain situations, a hardware jumper must be used to make an FXO port operate in ground-start mode.) The ground-start signaling method is used primarily on trunk lines or tie lines between private branch exchanges (PBXs).
*
The FXS interface is also an RJ-11 connector that connects directly to a standard telephone, fax machine, or similar device. The FXS interface supplies ringing voltage, dial tone, and similar signals to FXO devices. As the exercises in module 1 demonstrated, you would use the FXS interface when connecting a phone directly to a router. Basically, the FXS interface mimics the PSTN. The FXS ports on the Cisco 3600 series router can provide the battery at 24-volt DC. They can also support both loop-start and ground-start modes of operation according to the software configuration of the router. Loop start, however, is the most common, because most residential telephones are analog loop-start devices.
*
E&M, which stands for 'Ear and Mouth' (or 'recEive and transMit,' or sometimes 'Earth and Magnet'), is a signaling technique for two- and four-wire telephone and trunk interfaces that is used mainly between PBXs or other network-to-network telephony switches, such as Lucent 5 Electronic Switching System [5ESS], Nortel DMS-100, and so on. The E&M analog interface is an RJ-48S connector used to connect remote calls from an IP network to a PBX for local distribution and between PBX trunk lines (tie lines) or other network-to-network telephony switches. Unlike loop-start or ground-start modes of operation on the FXS/FXO ports, E&M uses separate wires for signaling and voice. There are five types of E&M signaling, as well as two wiring methods.
Exercise care when designing a voice network...
Exercise care when designing a voice network to ensure supervisory signals (discussed in module 1 of this tutorial) are transmitted when expected to avoid common problems such as spurious disconnects, undisconnected lines, and continuous rings. For example, a typical disconnect supervision problem can occur when a given line can be released only by the party who originated the phone call. This problem can be fixed by implementing either disconnect supervision or ground start.
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In the first module of this tutorial, you learned about the basics of analog voice internetworking, including the history of telephony, telephone networks and technology, and the various types of telephony signaling. You know how to configure and troubleshoot a Cisco 3640 router using loop-start access signaling across the FXS interface. In this module you will learn more about trunks, analog E&M signaling, and VoIP configuration using E&M. The Cisco VoIP implementation supports E&M types I, II, III, and V, using both two- and four-wire implementations.
Click on any highlighted term see its pop-up glossary definition. For additional information and more definitions of terms used throughout this course, see the full Glossary. You can access the Glossary at any time by clicking Help.
A basic understanding of trunks is necessary to understanding the use of E&M signaling. A line is a communications path between a customer's telephone and a telephone switch, such as a CO switch or a PBX, whereas a trunk is an actual telephone circuit or path between two switches, at least one of which is usually a CO or a switching center. This type of trunk, where voice is carried over a telco's network, is a CO trunk. Another example of a trunk is a private leased line that interconnects two PBXs, forming a corporate voice network. This type of connection is referred to as a PBX trunk.
Central-Office Trunks
Two-Wire Central-Office Trunks
A hunt group is...
A hunt group is a feature supported by voice-capable Cisco routers that involves the configuration of a group of dial peers on the same router with the same destination pattern. With a hunt group, if a call attempt is made to a dial peer on a specific time slot and that time slot is busy, the router hunts for another time slot on that channel until an available time slot is found.
The Cisco 3640 hunts for a group based on matching numbers.
It is possible to configure hunt groups by setting up multiple dial peers having the same destination pattern. Hunt groups are not configured in this tutorial, and the feature is not enabled in the simulation exercises in this module.
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The most common two-wire CO trunks used in PBX networks employ ground start. These trunks may be restricted such that operation may be outgoing only, incoming only, or both. In the case of outgoing calls (direct outward dialing, or DOD) operation additionally may be restricted to local calling only, versus direct distance dialing (DDD). Most incoming calls are routed to an attendant or to a messaging or answering system. The ringback tone is generated from the local exchange carrier's (LEC's) switch that connects to the PBX. This type of trunk is common.
Most of these trunks are assigned the same phone number within the PSTN, and operate in a 'hunt group' fashion.
Direct Inward Dialing Central Office Trunks
Direct inward dialing (DID) is used to determine how the called number is treated for incoming calls. Incoming calls to a PBX often first flow through an attendant position. DID trunks allow users to receive calls directly from the outside without intervention from the attendant.
Use of DID offers three main advantages:
* It allows direct access to stations from outside the PBX.
* It allows users to receive calls even when the attendant switchboard is closed.
* It takes a portion of the load off the attendants.
To accomplish DID on a trunk, the switch at the PSTN (either the local exchange carrier [LEC] or the inter-exchange carriers [IXC]) transmits the number of the called party to the PBX for routing. DID trunks often use existing two-wire circuits, and typically use ground-start signaling. Four-wire circuits using E&M signaling or digital access are sometimes used as well. To use DID, a block of PSTN telephone numbers generally must be reserved from the IXC or LEC.
DID trunk circuits usually employ either Wink Start or Delay-Dial signaling schemes, so that the network knows that the PBX is ready to accept the incoming address supervision. Wink Start and Delay Dial are discussed in more detail later in this section.
The next module in this tutorial, 'Basic Analog-to-Digital Voice over IP,' covers the use of DID in a VoIP configuration.
Wide-Area Telephone Service Central-Office Trunks
Wide-Area Telephone Service (WATS—sometimes referred to as OUTWATS) trunks are used for outgoing calls. They take advantage of interconnection to the IXC. These trunks require that the PBX output digits to the PSTN switch for routing. The network usually provides answer and disconnect supervision. WATS trunks can operate using either ground-start or loop-start signaling, in addition to E&M signaling. Ringback and busy signals are generated from the remote switch or PBX.
WATS trunks do not have addressable PSTN telephone numbers; therefore, these trunks are typically implemented in a hunt-group fashion within the PBX.
Central-Office 800-Service Trunks
Trunks that provide 800-service are sometimes referred to as INWATS service trunks. These trunks are used for incoming calls and take advantage of interconnection to the IXC. With an 800-service trunk, the PSTN does not output digits to the PBX; incoming calls on these trunk numbers are routed to specific individuals or groups by the PBX. These trunks may utilize loop reverse battery or E&M signaling.
Ringback and busy tones are usually generated by the PSTN switch that is interconnected to the PBX. Generally, these trunks are implemented in a hunt group fashion.
Long-Distance Trunks
Long-distance trunks interconnect to an IXC to allow both incoming and outgoing long-distance calling. They usually operate on two-wire ground-start circuits, or two- or four-wire E&M circuits. If access to the PSTN switch is digital (DS1), then conventional robbed-bit signaling (RBS) is used. (RBS is explained in greater detail in the next module of the Voice Internetworking CIM tutorial.)
Ringback and busy tones are usually generated by the IXC's PSTN switch that is interconnected to the PBX. Generally, these types of trunks are implemented in a hunt-group fashion, and incoming calls are typically routed to an attendant or messaging system.
PBX Trunks
A PBX is a private digital or analog telephone switchboard used to originate and answer calls to and from the public network (via CO trunks). PBXs can be combined with PBX trunks between them. When a caller picks up a handset and dials from within the corporation, the PBX connects that user to an idle line or to an idle trunk in an appropriate trunk group, then returns the appropriate call status signal, such as a dial tone or a ringback. A busy or fast-busy signal is returned if the line or the trunk group is busy. An attendant may be present to answer incoming calls and for user assistance.
FX Trunks
Foreign exchange (FX) circuits may consist of either stations or trunks. An FX line is a special line that is run from a local telephone to a CO switch or PBX switch. In this case, the local telephone is assigned a number on a remote switch; for all inbound and outbound calls, the telephone behaves as if it were connected to the remote switch. Generally, an FX circuit uses a two-wire circuit with loop start signaling. The ringback and busy signals are provided by the switch connected to the CO.
An FX trunk operates in a similar fashion, except that a local trunk provides interconnectivity to a remote switch. For outbound calls, local PBX users dial an access code such as 9, receive a dial tone from the remote switch, then output digits to the remote switch for routing. Incoming calls are either barred, or are routed to a PBX attendant or message system.
Because a trunk circuit is involved, features such as call transferring and call forwarding can also be deployed. These types of activities requires disconnect supervision to release the PBX trunk. This supervision cannot be provided with loop-start signaling, so ground start signaling must be used.
Tie Trunks and Tie Lines
Tie trunks interconnect PBX switches within a customer's network. Tie trunks typically connect both trunks and station lines through a network connection. These circuits are usually heavily used and normally support both incoming and outgoing calls.
Tie trunks are typically four-wire E&M-type circuits, but may often interconnect through DS1 facilities using RBS. The use of two-wire trunks is also allowable; these would typically use ground-start signaling.
Although tie lines are generally considered the same as tie trunks, and the terms are often used interchangeably, there are some technical differences:
* Tie trunks can interconnect both trunks and lines. In tie-trunk operation, routing of a call is totally automatic. The remote PBX usually supplies another dial tone, at which the user must enter the necessary address for the called party at that PBX.
* Tie lines connect only lines throughout a network connection. Routing of a call over tie lines is under control of the user. A user would need to dial an access code (9, for example) to gain access to the desired remote PBX. Tie lines are actually tie trunks that use tone-start type of start-dial supervision and cut-through operation. (Details about start-dial supervision can be found at the end of this section.)
Although the next module will include use of a PBX in the network topology, a complete discussion of PBXs and PBX configuration is out of the scope of this tutorial. To learn about PBX configuration, you should refer to the manuals for the particular PBX that you are using in your network. For additional PBX information, however, please see the related document The Private Branch Exchange (PBX) .
As discussed in module 1, 'Basic Analog Voice over IP,' the primary purpose of signaling in a voice network is to establish a connection. Signaling can be classified into four basic functions:
* Supervisory signaling — Informs the telephone or switch of the status of the local loop and any connected trunks. Supervisory signaling is used to:
o Initiate a call request on line or trunks (called line signaling on trunks).
o Indicate that the call has been answered or the call has been disconnected (disconnect and answer supervision).
o Initiate or terminate charging for calls.
o Recall an operator on an established connection.
* Address signaling — Contains information indicating the destination of a call, such as the telephone number and an area code, an access code, or a PBX tie trunk access code.
* Call progress indicators — Convey call-progress or call-failure information to subscribers or operators by the use of usually audible tones (such as the busy signals, the reorder tone, and the ringback, all of which were discussed in module 1).
* Network management signaling — Controls the bulk assignment of circuits or modifies the operating characteristics of switching systems in a network in response to overload conditions. (These signaling functions will not be covered in detail in this tutorial, but are mentioned here for completeness.)
For a more detailed discussion of signaling, including descriptions of supervision, address, call-progress, and network management signaling, read Signaling, which is included in this tutorial. Also search Cisco Connection Online (CCO) at http://www.cisco.com for more information on this specific topic as well as more available information on treansmitting voice over a data network.
Just as the local loop requires supervisory signaling, so does a trunk. Although loop-start signaling, which is generally used on everyday residential telephones, can also be used on trunks, its use can result in problems that make it an ineffective signaling method for a trunk. With loop start, only the device originating the call can release the connection. Also, loop start would not prevent a trunk from being seized by both parties to a call, a condition referred to as glare. Glare can be tolerable on a local loop, but its occurrence on a trunk would be unacceptable. Ttwo-way handshaking methods were developed to coordinate the sequence of events that occur during a call: the request by the calling end for access to the trunk, followed by the acknowledgment by the called end, then the subsequent seizure of the trunk by the calling end.
The basic types of supervisory signaling used on trunks are: ground start, E&M, and start-dial supervision.
Ground-Start Signaling
In module 1 of this course, you configured the Cisco 3640 router FXS port to use loop-start signaling, because calls were being made using a directly connected analog telephone. Ground start is the access signaling method used on trunk lines or tie lines between PBXs to indicate on-hook/off-hook status to the CO. Ground-start signaling works by using ground and current detectors. Ground start behaves like a loop start, however, the PBXs on both sides—the telco and the customer's telephone—have to agree on the status of a line before a call can be placed. This scenario allows the network to indicate off-hook or seizure of an incoming call independent of the ringing signal.
E&M Signaling Interface Types
Analog trunk circuits connect between automated systems (a PBX) and the network (a CO). The most common form of analog trunk is the E&M interface.
E&M signaling is typically used for trunk lines. It provides native support for both disconnect and answer supervision, as well as glare avoidance. In E&M signaling, separate paths are used for voice and signaling. E&M is normally the only way that a CO switch can provide two-way dialing with DID.
The history of the semantics of E&M circuits dates back to the days of telegraphy, where the main office end had a 'key' that grounded the E circuit, and the other end had a sounder with an electromagnet attached to a battery. Descriptions such as 'ear' and 'mouth' were created to help provide a reference to field personnel as to the direction of a signal in a wire. Note that these terms correspond to the inbound and outbound circuits, known as the E lead and M lead, respectively.
There are five standard types of E&M interfaces, which correspond to the five signaling types:
* E&M Type I Interface Model
* E&M Type II Interface Model
* E&M Type III Interface Model
* E&M Type IV Interface Model
* E&M Type V Interface Model
With each signaling type, the PBX supplies one signal, known as the M signal (mouth), and accepts one signal, known as the E signal (ear). Conversely, the tie-line equipment accepts the M signal from the PBX and provides the E signal to the PBX. The M signal accepted by the tie-line equipment at one end of a tie circuit becomes the E signal output by the remote tie-line interface.
Remember learning about tip and ring in module 1?
* When a user tries to place a call by grounding the 'ring' lead, the PBX at the telco senses the flow of current and grounds the 'tip' lead to indicate the PBX is ready to serve.
* The user's equipment perceives the flow of current on its 'tip' and knows that the PBX is ready to serve.
Click to close sidebar.
The signaling part of the interface has five distinct physical configurations (Types I – V) and the audio interface has two (two- and four-wire). Note that the terms two-wire and four-wire are used to reference the communication protocol of the audio path. The difference between a two-wire and four-wire circuit is whether the audio path is full duplex on one pair or two pairs of wires.
Two full-duplex wires (called tip and ring) carry the audio on two-wire E&M. Four-wire E&M uses four half-duplex wires for the audio path (Tip, Ring, Tip1, and Ring1). Note that a four-wire E&M circuit may have from six to eight physical wires.
Signaling Types
* E — Ear or earth: Signal wire from the trunking (CO or network) side to signaling (user) side
* M — Mouth or magnet: Signal wire from signaling (user) side to trunking (CO or network) side
* SG — Signal ground: Used only on certain types of E&M; sometimes grounded, sometimes not>
* SB — Signal battery: Used on certain types of E&M; sometimes provides -48V Direct Current, sometimes ground, sometimes is not used at all
* T/R — Tip/ring: Used only on a four-wire circuit; carries audio from the signaling (user) side to the trunking (CO/network) side; not used on a two-wire circuit
* T1/R1 — Tip1/ring1: On four-wire circuits, carries audio from the trunking (CO/network) side to the signaling (user) side; on a two-wire circuit, this pair carries the full-duplex audio path
Types I and II are the most popular E&M signaling types in the Americas. Type V is used in the United States, and is very popular in Europe. (SSDC5 is most often found in the United Kingdom.) Similar to Type V, SSDC5A differs in that on- and off-hook states are backward to allow for fail-safe operation: if the line breaks, the interface defaults to off hook (busy). Of all the types, only Types II and V are symmetrical (can be back to back using a crossover cable).
The Cisco 2600 and 3600 series routers currently support types I, II, III, and V utilizing both two- and four-wire implementations. The E&M interface of the Cisco 3600 series presents the 'channel bank,' 'tie equipment,' or 'CO' side of a trunk interface to a PBX. Each E&M signaling type has a unique circuit model and connection diagram. All Cisco analog voice gateway routers have 24V DC designed on the FXS ports. According to EIA 464 standard, any voltage from DC 24 volts to DC 52 volts is acceptable.
Characteristics of E&M Lead Signaling
Type
M Lead
E Lead
Outbound Direction Inbound Direction
Off Hook
On Hook
Off Hook
On Hook
I
Battery
Ground
Ground
Open
II
Battery
Open
Ground
Open
III
Loop current
Ground
Ground
Open
IV
Ground
Open
Ground
Open
V
Ground
Open
Ground
Open
SSDC5
Earth on
Earth off
Earth on
Earth off
Cisco routers expect to see off-hook conditions on the M lead and signal off hook to the remote device on the E lead.
See the following sections and illustrations for individual descriptions of the different types. Each of the following illustrations shows the E&M interface of the PBX on the left, and the corresponding tie-line equipment interface on the right. The symbol V refers to battery voltage, which can be 25 VDC to 65 VDC, and is usually (nominally) -48 VDC.
E&M Type I
In E&M Type I signaling, the battery PBX provides the battery for both the E and M leads. During an off-hook condition, the PBX generates the E signal by grounding the E lead. The PBX detects the E signal by sensing the increase in current through a resistive load. Similarly, in an on-hook condition, the PBX generates the M signal by sourcing a current to the tie-line equipment, which detects it via a resistive load.
The four-wire Type I interface from the PBX has the following characteristics:
* E (pin 7) detector 'floats' at -48V below ground
* M (pin 2) contact has low ohms to ground on hook, and is -48V below ground when off hook
* Approximately 30-150 ohms between T/R (pins 6/3) sometimes in series with 2.2uf of capacitance
* Approximately 30-150 ohms between T1/R1 (pins 5/4) sometimes in series with 2.2uf of capacitance
E&M Type I
The Type I interface requires that the PBX and tie-line equipment share a common signaling ground reference. This setup can be achieved by connecting signal ground from the PBX to the signal-ground (SG) lead (pin 8) of the RJ-48S connector.
Note that the voltage on the E and M leads may not be the same (the E lead may have lower voltage). This asymmetrical signaling scheme is a potential source of interference, which could cause high return current through the grounding system. If two PBXs were not correctly grounded, current could flow down the M lead, causing the remote PBX to erroneously detect a current on the E lead, thus resulting in false seizure of a trunk. Despite this potential problem, E & M signaling is the most common four-wire trunk interface used in North America.
E&M Type II
The E&M Type II interface provides almost complete isolation of signaling power systems by the addition of two additional signaling leads: signal battery (SB) and signal ground (SG). In Type II, each of the two signals has its own return. For the E signal, the E lead works with the SSG lead to allow current to flow from the PBX while the M lead is strapped to the SB lead. This scenario results in the trunk being grounded at each end, eliminating the potential problem associated with Type I, discussed above.
The four-wire Type II interface from the PBX has the following characteristics:
* E lead (pin 7) detector 'floats' at -48V below ground
* SG lead (pin 8) has a low ohms to ground
* M lead (pin 2) contact between M and SB is open when on-hook, and closed when off hook
* M lead floats
* SB lead (pin 1) floats
* Approximately 30-150 ohms between T/R (pins 6/3) sometimes in series with 2.2uF of capacitance
* Approximately 30-150 ohms between T1/R1 (pins 5/4) sometimes in series with 2.2uF of capacitance
E&M Type II
Type II is seen only occasionally in North America, usually on Centrex trunk circuits or Nortel PBX systems.
E&M Type III
E&M Type III signaling is very similar to Type I, except that the battery and ground source for the M lead is supplied by the transmission equipment. Complete power isolation is provided with the M lead, and the facility can establish and control the amount of E lead current. Type III uses the SG lead to provide common ground. The PBX drops the M signal by grounding it, rather than by opening a current loop.
The four-wire Type III interface from the PBX has the following characteristics:
* E lead (pin 7) detector 'floats' at -48V below ground
* M lead (pin 2) contact between M and SG when on hook, and between M and SB when off hook
* SG lead (pin 8) floats
* M lead (pin 2) floats
* SB lead (pin 1) floats
* Approximately 30-150 ohms between T/R (pins 6/3) sometimes in series with 2.2uF of capacitance
* Approximately 30-150 ohms between T1/R1 (pins 5/4) sometimes in series with 2.2uf of capacitance
E&M Type III
There is no evidence that the unbalanced E lead of Type III has caused any interference problems, but one drawback of this interface is its inability to operate in a 'back-to-back' configuration.
This interface is most often used in older CO equipment such as 1/1AESS, 2/2BESS, and 3ESS switches. It is not often seen now because most of these older switches have been replaced.
E&M Type IV
Type IV is symmetric and requires no common ground. Each side closes a current loop to signal; the flow of current is detected via a resistive load to indicate the presence of the signal. The Type IV interface is similar to Type II, with the difference in the operation of the M lead—in Type II, the M lead states are 'open' and 'battery;' Type IV states are 'ground' and 'open.'
The four-wire Type IV interface from the PBX has the following characteristics:
* E lead (pin 7) detector 'floats' at -48V below ground
* SG lead (pin 8) has low ohms to ground
* M lead (pin 2) contact between M and SB is open when on-hook, and closed when off hook
* M lead (pin 2) floats
* SB lead (pin 1) floats
* Approximately 30-150 ohms between T/R (pins 6/3) sometimes in series with 2.2uF of capacitance
* Approximately 30-150 ohms between T1/R1 (pins 5/4) sometimes in series with 2.2uf of capacitance
E&M Type IV
The advantages of Type IV include:
* Accidental shorting of the SB lead (during cable wiring, for example) will not result in an excessive current flow.
* The interface can interconnect to a Type II device.
* The interface can operate in a 'back-to-back' configuration.
Because it can be difficult for an external monitor to distinguish between 'open' and 'ground' states, it can be difficult to obtain test and supporting equipment for a Type IV interface. Type IV is not currently supported by the Cisco 3600 series routers.
E&M Type V
E&M Type V interface is a simplified version of Type IV. It is also a symmetric interface, using only two wires. In the Type V interface, both the switch and the transmission equipment supply battery. The battery for the M lead is located in the signaling equipment, and the battery for the E lead is located in the PBX. Type V requires a common ground between the PBX and the tie line equipment, which is provided via the SG leads.
The Type V interface from the PBX has the following characteristics:
* E lead (pin 7) detector 'floats' at -48V below ground
* M lead (pin 2) contact ground is open when on-hook, and closed when off hook
* Approximately 30-150 ohms between T/R (pins 6/3) sometimes in series with 2.2uF of capacitance
* Approximately 30-150 ohms between T1/R1 (pins 5/4) sometimes in series with 2.2uf of capacitance
E&M Type V
Although this interface does not provide isolation between power systems, there is minimal (or no) return currents in this symmetrical signaling scheme. Type V is the most popular interface outside North America.
E&M Start-Dial Supervision Signaling Protocol
Start-dial supervision is the line protocol used between equipment that takes place after the initial off-hook condition, up to the passing of dial digits to the connected device.
Three principal protocols are used on E&M circuits:
* E&M Immediate Start Signaling
* E&M Wink-Start Signaling
* E&M Delay-Dial Signaling
Wink start is used to notify the remote side it can send dialed number identification service (DNIS) information. Wink acknowledgment is a second wink that is sent to acknowledge the receipt of the DNIS information. Immediate start does not send any winks at all.
The following sections explain how each protocol works. It is important to understand what the protocol is supposed to do when debugging, because call progress anomalies provide clues to the cause.
Immediate Start
E&M Immediate Start is the simplest of protocols. Here, the originating switch goes off hook, waits for a finite period of time (say, 200 ms) then sends the address digits without regard to the remote. Because there is no acknowledgment, or handshaking, between switches, this type of trunk signaling should be used only when there is a dedicated physical or logical trunk between switches. Cisco routers support the use of the Immediate Start protocol.
Immediate Start
Wink Start
E&M Wink Start is the most common of protocols. Wink start is an in-band technique in which the calling switch waits from 140 to 290 ms for an off-hook wink pulse from the called switch before sending the dialed digits. Wink start was developed to minimize glare, which occurs when both ends attempt to seize the trunk at the same time. If there is no wink pulse, an error condition is caused by both ends attempting to place a call on the same trunk.
* In the original Wink Start protocol, the remote responds to an off hook from the originating device with a short wink (transition from on hook to off hook and back again). This wink tells the originating device that the remote device is ready to receive digits. After receiving the addressing digits, the remote then goes off hook for the duration of the call. The originating device maintains off hook for the duration of the call.
* In Wink Start with Wink Acknowledge protocol (sometimes referred to as double wink), the remote responds to an off hook from the originating device with a short wink (transition from on hook to off hook and back again), just as in the original Wink Start. This wink tells the originating side that the remote is ready to receive digits. After receiving the addressing digits, the remote then provides another wink (called an Acknowledgment Wink) that tells the originating side that the terminating side has received the dialed digits. The remote then goes off hook to indicate connection when the ultimate called endpoint has answered. The originating device maintains off-hook status for the duration of the call.
Wink Start normally is not used on trunks that are controlled with message-oriented signaling schemes such as ISDN or Signaling System 7 (SS7). Cisco routers support both wink-start (fgb) and wink-start with wink-acknowledge or double-wink (fgd).
Wink Start
Delay Dial
In E&M Delay Dial mode, the originating device goes off hook and waits for about 200 ms, then checks to see if the remote end is on hook. If so, it then outputs addressing digits. If the remote is off hook, the calling device waits until the remote goes back on hook before transmitting digits. The delay signal says, in effect 'hold on, I'm not ready to receive digits.' This protocol was invented for use with systems that have fewer digit collectors than trunk interfaces. Delay dial is not currently supported by Cisco 3600 series routers.
Delay
Start Dial Supervision Mismatches
Although both ends of a call do not need Start Dial Supervision, mismatches can be a source of problems. Sometimes a PBX has a different Start Dial Supervision protocol for inbound and outbound calls. This setup can lead to erratic behavior if the remote is not configured to properly handle this condition. The following general rule set applies:
* An Immediate-Start interface can usually originate a call to a Wink-Start interface.
* An Immediate-Start interface can usually place a call to a Delay-Dial interface if the delay pulse is shorter than the immediate-start delay. Otherwise, operation is erratic.
* A Wink-Start interface can usually originate a call into a Delay-Dial interface if there is a delay pulse. Otherwise, the call will hang, with only a 50-percent chance of working.
* A Delay-Dial interface can, for the most part, originate a call into an Immediate-Start or Wink-Start interface.
Go on to Configuration of VoIP with E&M Signaling."
Cisco - Signaling
Cisco - Signaling: "Signaling
Signaling is defined by Consultative Committee for International Telegraph and Telephone (CCITT) Recommendation Q.9 as 'the exchange of information (other than speech) specifically concerned with the establishment, release, and control of calls, and network management in automatic telecommunications operations.'
In the broadest sense, there are two signaling realms:
* Subscriber signaling
* Trunk signaling (interswitch and/or interoffice)
Signaling is also traditionally classified into four basic functions:
* Supervision
* Address
* Call progress
* Network management
Supervision signaling is used to:
* Initiate a call request on line or trunks (called line signaling on trunks).
* Hold or release an established connection.
* Initiate or terminate charging.
* Recall an operator on an established connection.
Address signaling conveys such information as the calling or called subscriber's telephone number and an area code, an access code, or a Private Automatic Branch Exchange (PABX) tie trunk access code. An address signal contains information indicating the destination of a call initiated by a customer, network facility, and so forth.
Call progress signals are usually audible tones or recorded announcements that convey call-progress or call-failure information to subscribers or operators. These call-progress signals are fully described below.
Network management signals are used to control the bulk assignment of circuits or to modify the operating characteristics of switching systems in a network in response to overload conditions.
There are about 25 recognized interregister signaling systems worldwide, in addition to some subscriber signaling techniques. CCITT Signaling System Number 7 (SSN7) is fast becoming the international/national standard interregister signaling system.
Most installations will probably involve recEive and transMit (E&M) signaling; however, for reference, single frequency (SF) signaling on Tip and Ring loops, Tip and Ring reverse battery loops, loop start, and ground start are also included.
Types I and II are the most popular E&M signaling in the Americas. Type V is used in the United States, but is very popular in Europe. Similar to type V, SSDC5A differs in that on- and off-hook states are reversed to allow for fail-safe operation: if the line breaks, the interface defaults to off-hook (busy). Of all the types, only II and V are symmetrical (can be back-to-back using a cross-over cable). SSDC5 is most often found in England.
Other signaling techniques often used are delay, immediate, and wink start. Wink start is an in-band technique where the originating device waits for an indication from the called switch before sending the dialed digits. Wink start normally is not used on trunks that are controlled with message-oriented signaling schemes such as Integrated Services Digital Network (ISDN) or Signaling System 7 (SS7).
Summary of Signaling System Applications and Interfaces
Signaling System Application/Interface Characteristics
Station Loop
Loop signaling
Basic Station
DC signaling.
Origination at station.
Ringing from Central Office.
Coin Station
DC signaling.
Loop-start or ground-start origination at station.
Ground and simplex paths may be used in addition to the line for coin collection and return.
Interoffice Trunk
Loop Reverse Battery
One-way call origination.
Directly applicable to metallic facilities.
Both current and polarity are sensed.
Can be used on carrier facilities with appropriate facility signaling system.
E&M Lead
Two way call origination.
Requires facility signaling system for all applications.
Facility Signaling System
Metallic DX
Analog SF
Digital Bits in information
Special Service
Loop Type
Standard station loop and trunk arrangement as above.
Ground-start format similar to coin service for PBX-CO trunks.
E & M Lead
E&M for PBX dial tie trunks. E&M for carrier system channels in special service circuits.
North American Practices
The typical North American touchtone set provides a 12-tone set. Some custom sets provide 16-tone signals of which the extra digits are identified by the A-D pushbuttons.
Dual Tone Multifrequency (DTMF) Frequency Pairs
Low Frequency Group (Hz) High Frequency Group (Hz)
1209 1336 1477 1633
697 1 2 3 A
770 4 5 6 B
852 7 8 9 C
941 * 0 # D
Audible tones commonly used in North America
Tone Frequencies (Hz) Cadence
Dial 350 + 440 Continuous
Busy (station) 480 + 620 0.5 sec on, 0.5 sec off
Busy (network) 480 + 620 0.2 sec on, 0.3 sec off
Ring return 440 + 480 2 sec on, 4 sec off
Off-hook alert Multifreq howl 1 sec on, 1 sec off
Recording warning 1400 0.5 sec on, 15 sec off
Call waiting 440 0.3 sec on, 9.7 sec off
Call Progress Tones Used in North America
Name Frequencies (Hz) Pattern Levels
Low tone 480 + 620
600 x 120
600 x 133
600 x 140
600 x 160 Various -24 dBm0
61 to 71 dBmC
61 to 71 dBmC
61 to 71 dBmC
61 to 71 dBmC
High tone 480
400
500 Various -17 dBmC
61 to 71 dBmC
61 to 71 dBmC
Dial tone 350 + 440 Steady -13 dBm0
Audible ring tone 440 + 480
440 + 40
500 + 40 2 sec on, 4 sec off
2 sec on, 4 sec off
2 sec on, 4 sec off -19 dBmC
61 to 71 dBmC
61 to 71 dBmC
Line Busy Tone 480 + 620
600 x 120
600 x 133
600 x 140
600 x 160 0.5 sec on, 0.5 sec off
Reorder 480 + 620
600 x 120
600 x 133
600 x 140
600 x 160 0.3 sec on, 0.2 sec off
6A alerting tone 440 2 sec on, followed by 0.5 sec on, every 10 sec
Recorder warning tone 1400 0.5 sec burst every 15 sec
Reverting tone 480 + 620
600 x 120
600 x 133
600 x 140
600 x 160 0.5 sec on, 0.5 sec off -24 dBmC
Deposit coin tone 480 + 620
600 x 120
600 x 133
600 x 140
600 x 160 Steady
Receiver off-hook (analog) 1400 + 2060 + 2450 + 2600 0.1 sec on, 0.1 sec off +5 vu
Receiver off-hook 1400 + 2060 + 2450 + 2600 0.1 sec on, 0.1 sec off +3.9 to -6.0 dBm
Howler 480 Incremented in level
Every 1 sec for 10 sec Up to 40 vu
No such number (crybaby) 200 to 400 Freq. modulated at 1 Hz interrupted every 6 sec for 0.5 sec
Vacant code 480 + 620
600 x 120
600 x 133
600 x 140
600 x 160 0.5 sec on, 0.5 sec off, 0.5 sec on, 1.5 sec off?
Busy verification Tone (Centrex) 440 Initial 1.5 sec followed 0.3 sec every 7.5 to 10 sec -13 dBm0
Busy verification Tone (TSPS) 440 Initial 2 sec followed 0.5 sec every 10 sec -13 dBm0
Call waiting tone 440 Two bursts of 300 ms separated by 10 sec -13 dBm0
Confirmation tone 350 + 440 3 bursts of 300 ms separated by 10 sec -13 dBm0
Indication of camp-on 440 1 sec every attendant releases from loop -13 dBm0
Recall dial tone 350 + 440 3 bursts, 0.1 sec on, sec off then steady -13 dBm0
Data set answer back tone 2025 Steady -13 dBm
Calling card prompt tone 941 + 1477 followed by 440 + 350 60ms -10 dBm0
Class of service 480
400
500 0.5 to 1 sec once
Order tones
Single
480
400
500 0.5 sec
Double
480
400
500 2 short bursts
Triple
480
400
500 3 short bursts
Quad
480
400
500 4 short bursts
Number checking tone 135 Steady
Coin denomination
3 5 cents
1050-1100 (bell) One tap
slot 10 cents
1050-1100 (bell) Two taps
stations 25 cents
800 (gong) One tap
Coin collect tone 480 + 620
600 x 120
600 x 133
600 x 140
600 x 160 Steady
Coin return tone 480
400
500 0.5 to 1 sec once
Coin return test tone 480
400
500 0.5 to 1 sec once
Group busy tone 480 + 620
600 x 120
600 x 133
600 x 140
600 x 160 Steady
Vacant position 480 + 620
600 x 120
600 x 133
600 x 140
600 x 160 Steady
Dial off normal 480 + 620
600 x 120
600 x 133
600 x 140
600 x 160 Steady
Permanent signal 480
400
500 Steady
Warning tone 480
400
500 Steady
Service observing 135 Steady
Proceed to send Tone (IDDD) 480 Steady -22 dBm0
Centralized intercept 1850 500 ms -17 dBm0
ONI order tone 700 + 1100 95 to 250 ms -25 dBm0
Note: Three dots in the pattern mean that the pattern is repeated indefinitely.
Single Frequency In-Band Signaling
SF in-band signaling is widely used in North America. Its most common application is for supervision, such as idle-busy, also called line signaling. It also can be used for dial pulse signaling on trunks. The dynamics of SF signaling requires an understanding of the signal durations and configurations of the E&M circuits, as well as the lead interface arrangements. The following tables show the characteristics of SF signaling, E&M lead configurations, and interface arrangements.
Typical Single Frequency Signaling Characteristics
General
Signaling frequency (tone)
2600 Hz
Idle state transmission
Cut
Idle/break
Tone
Busy/make
No tone
Receiver
Detector bandwidth
+/- 50 Hz @ -7 dBm for E type
+/- 30 Hz @ -7 dBm
Pulsing rate
7.5 to 122 pps
E/M unit
Minimum time for on-hook
33 ms
Minimum no tone for off-hook
55 ms
Input percent break (tone)
38-85 (10 pps)
E lead - open
Idle
- ground
Busy
Originating (loop reverse battery) unit
Minimum tone for idle
40 ms
Minimum no tone for off-hook
43 ms
Minimum output for on-hook
69 ms
Voltage on R lead (-48 V on ring and ground on tip)
On-hook
Voltage on T lead (-48 V on tip and ground on ring)
Off-hook
Terminating (loop reverse battery) unit
Minimum tone for on-hook
90 ms
Minimum no tone for off-hook
60 ms
Minimum output (tone-on)
56 ms
Loop open
On-hook
Loop closed
Off-hook
Transmitter
Low level tone
-36 dBm
High level tone
-24 dBm
High level tone duration
400 ms
Precut
8 ms
Holdover cut
125 ms
Crosscut
625 ms
On hook cut
625 ms
E/M unit
Voltage on M lead
Off-hook (no tone)
Open/ground on M lead
On-hook (tone)
Minimum ground on M lead
21 ms
Minimum voltage on M lead
21 ms
Minimum output tone
21 ms
Minimum no tone
21 ms
Originating (loop reverse battery) unit
Loop current to no tone
19 ms
No loop current to tone
19 ms
Minimum input for tone out
20 ms
Minimum input for no tone out
14 ms
Minimum tone out
51 ms
Minimum no tone out
26 ms
Loop open
On-hook
Loop closed
Off-hook
Terminating (loop) unit
Reverse battery to no tone
19 ms
Normal battery to tone
19 ms
Minimum battery for tone out
25 ms
Minimum reverse battery for no tone
14 ms
Minimum tone out
51 ms
Minimum no tone out
26 ms
Battery on R lead (-48 v)
On-hook
Battery on TY lead (-48 on tip)
Off-hook
Single Frequency Signals Used in E&M Lead Signaling
Calling End Called End
Signal M-Lead E-Lead 2600 Hz 2600 Hz E-Lead M-Lead Signal
Idle Ground Open On On Open Ground Idle
Connect Battery Open Off On Ground Ground Connect
Stop dialing Battery Ground Off Off Ground Battery Stop dialing
Start dialing Battery Open Off On Ground Ground Start dialing
Dial pulsing Ground Open On On Open Ground Dial pulsing
Battery
Off
Ground
Off -hook Battery Ground Off Off Ground Battery Off-hook (answer)
Ring forward Ground Ground On Off Open Battery Ring forward
Battery
Off
Ground
Ringback Battery Open Off On Ground Ground Ringback
Ground
Off
Battery
Flashing Battery Open Off On Ground Ground Flashing
Ground
Off
Battery
On-hook Battery Open Off On Ground Ground On-hook
Disconnect Ground Open On On Open Ground Disconnect
Single Frequency Signals Used in Reverse Battery Tip and Ring Loop Signaling
Calling End Called End
Signal T/R - SF SF - T/R 2600 Hz 2600 Hz T/R - SF SF - T/R Signal
Idle Open Batt-gnd On On Open Batt-gnd Idle
Connect Closure Batt-gnd Off On Closure Batt-gnd Connect
Stop dialing Closure Rev batt-gnd Off Off Closure Rev batt-gnd Stop dialing
Start dialing Closure Batt-gnd Off On Closure Batt-gnd Start dialing
Dial pulsing Open Batt-gnd On On Open Batt-gnd Dial pulsing
Closure
Off
Closure
Off -hook Closure Rev batt-gnd Off Off Closure Rev batt-gnd Off-hook (answer)
Ring forward Open Rev batt-gnd On Off Open Rev batt-gnd Ring forward
Closure
Off
Closure
Ringback Closure Batt-gnd Off On Closure Batt-gnd Ringback
Rev batt-gnd
Off
Rev batt-gnd
Flashing Closure Batt-gnd Off On Closure Batt-gnd Flashing
Rev batt-gnd
Off
Rev batt-gnd
On-hook Closure Batt-gnd Off On Closure Batt-gnd On-hook
Disconnect Open Batt-gnd On On Open Batt-gnd Disconnect
Single Frequency Signals Used for Ringing and Loop-Start Signaling Using Tip and Ring Leads
Call Originating at Central Office End
Signal T/R - SF SF - T/R 2600 Hz 2600 Hz T/R - SF SF - T/R Signal
Idle Gnd-batt Open Off On Gnd-batt Open Idle
Seizure Gnd-batt Open Off On Gnd-batt Open Idle
Ringing Gnd-batt and 20 Hz Open On-off On Gnd-batt and 20 Hz Open Ringing
Off-hook (ring-trip and talk) Gnd-batt Closure Off Off Gnd-batt Closure Off-hook (ring-trip and answer)
On-hook Gnd-batt Closure Off Off Gnd-batt Closure Off-hook
On-hook (hang-up) Gnd-batt Open Off On Gnd-batt Open On-hook (hang-up)
Note: 20 Hz ringing (2 sec on, 4 sec off)
Call Originating at Station End
Signal T/R - SF SF - T/R 2600 Hz 2600 Hz T/R - SF SF - T/R Signal
Idle Open Gnd-batt On Off Open Gnd-batt Idle
Off-hook (seizure) Closure Gnd-batt Off Off Closure Gnd-batt Idle
Start dial Closure Dial tone and gnd-batt Off Off Closure Dial tone and gnd-batt Start dial
Dial pulsing Open-closure Gnd-batt On-off Off Open-closure Gnd-batt Dial pulsing
Waiting answer Closure Audible ring and gnd-batt Off Off Closure Audible ring and gnd-batt Waiting answer
On-hook (talk) Closure Gnd-batt Off Off Closure Gnd-batt Off-hook (answered)
On-hook (hang-up) Open Gnd-batt
Closure On Off Open Gnd-batt On-hook (disconnected)
Off-hook
Single Frequency Signals Used for Ringing and Ground-Start Signaling Using Tip and Ring Leads
Call Originating at Central Office End
Signal T/R - SF SF - T/R 2600 Hz 2600 Hz T/R - SF SF - T/R Signal
Idle Open-batt Batt-batt On On Open-batt
Idle
Seizure Gnd-batt Open On On Gnd-batt
Make-busy
Ringing Gnd-batt and 20 Hz Open On and 20 Hz On Gnd-batt and 20 Hz Open Ringing
Off-hook (ring-trip and talk) Gnd-batt Closure Off Off Gnd-batt Closure Off-hook (ring-trip and answer)
On-hook Gnd-batt Closure On Off Open-batt Closure On-hook
On-hook (hang-up) Gnd-batt Open Off On Gnd-batt Open On-hook (hang-up)
Note: 20 Hz ringing (2 sec on, 4 sec off)
Call Originating at Station End
Signal T/R - SF SF - T/R 2600 Hz 2600 Hz T/R - SF SF - T/R Signal
Idle
Open-batt On On Batt-batt Open-batt Idle
Off-hook (seizure) Ground Open-batt Off On Batt-gnd Open-batt Seizure
Start dial Closure Dial tone and gnd-batt Off Off Closure Dial tone and gnd-batt Start dial
Dial pulsing Open-closure Gnd-batt On-off Off Open-closure Gnd-batt Dial pulsing
Waiting answer Closure Audible ring and gnd-batt Off Off Closure Audible ring and gnd-batt Waiting answer
Off-hook (talk) Closure Gnd-batt Off Off Closure Gnd-batt Off-hook (answered)
On-hook Closure Open-batt On On Batt-batt Open-batt On-hook (disconnected)
On-hook (disconnected)
Closure On Off Open-batt Open-batt On-hook"
Signaling is defined by Consultative Committee for International Telegraph and Telephone (CCITT) Recommendation Q.9 as 'the exchange of information (other than speech) specifically concerned with the establishment, release, and control of calls, and network management in automatic telecommunications operations.'
In the broadest sense, there are two signaling realms:
* Subscriber signaling
* Trunk signaling (interswitch and/or interoffice)
Signaling is also traditionally classified into four basic functions:
* Supervision
* Address
* Call progress
* Network management
Supervision signaling is used to:
* Initiate a call request on line or trunks (called line signaling on trunks).
* Hold or release an established connection.
* Initiate or terminate charging.
* Recall an operator on an established connection.
Address signaling conveys such information as the calling or called subscriber's telephone number and an area code, an access code, or a Private Automatic Branch Exchange (PABX) tie trunk access code. An address signal contains information indicating the destination of a call initiated by a customer, network facility, and so forth.
Call progress signals are usually audible tones or recorded announcements that convey call-progress or call-failure information to subscribers or operators. These call-progress signals are fully described below.
Network management signals are used to control the bulk assignment of circuits or to modify the operating characteristics of switching systems in a network in response to overload conditions.
There are about 25 recognized interregister signaling systems worldwide, in addition to some subscriber signaling techniques. CCITT Signaling System Number 7 (SSN7) is fast becoming the international/national standard interregister signaling system.
Most installations will probably involve recEive and transMit (E&M) signaling; however, for reference, single frequency (SF) signaling on Tip and Ring loops, Tip and Ring reverse battery loops, loop start, and ground start are also included.
Types I and II are the most popular E&M signaling in the Americas. Type V is used in the United States, but is very popular in Europe. Similar to type V, SSDC5A differs in that on- and off-hook states are reversed to allow for fail-safe operation: if the line breaks, the interface defaults to off-hook (busy). Of all the types, only II and V are symmetrical (can be back-to-back using a cross-over cable). SSDC5 is most often found in England.
Other signaling techniques often used are delay, immediate, and wink start. Wink start is an in-band technique where the originating device waits for an indication from the called switch before sending the dialed digits. Wink start normally is not used on trunks that are controlled with message-oriented signaling schemes such as Integrated Services Digital Network (ISDN) or Signaling System 7 (SS7).
Summary of Signaling System Applications and Interfaces
Signaling System Application/Interface Characteristics
Station Loop
Loop signaling
Basic Station
DC signaling.
Origination at station.
Ringing from Central Office.
Coin Station
DC signaling.
Loop-start or ground-start origination at station.
Ground and simplex paths may be used in addition to the line for coin collection and return.
Interoffice Trunk
Loop Reverse Battery
One-way call origination.
Directly applicable to metallic facilities.
Both current and polarity are sensed.
Can be used on carrier facilities with appropriate facility signaling system.
E&M Lead
Two way call origination.
Requires facility signaling system for all applications.
Facility Signaling System
Metallic DX
Analog SF
Digital Bits in information
Special Service
Loop Type
Standard station loop and trunk arrangement as above.
Ground-start format similar to coin service for PBX-CO trunks.
E & M Lead
E&M for PBX dial tie trunks. E&M for carrier system channels in special service circuits.
North American Practices
The typical North American touchtone set provides a 12-tone set. Some custom sets provide 16-tone signals of which the extra digits are identified by the A-D pushbuttons.
Dual Tone Multifrequency (DTMF) Frequency Pairs
Low Frequency Group (Hz) High Frequency Group (Hz)
1209 1336 1477 1633
697 1 2 3 A
770 4 5 6 B
852 7 8 9 C
941 * 0 # D
Audible tones commonly used in North America
Tone Frequencies (Hz) Cadence
Dial 350 + 440 Continuous
Busy (station) 480 + 620 0.5 sec on, 0.5 sec off
Busy (network) 480 + 620 0.2 sec on, 0.3 sec off
Ring return 440 + 480 2 sec on, 4 sec off
Off-hook alert Multifreq howl 1 sec on, 1 sec off
Recording warning 1400 0.5 sec on, 15 sec off
Call waiting 440 0.3 sec on, 9.7 sec off
Call Progress Tones Used in North America
Name Frequencies (Hz) Pattern Levels
Low tone 480 + 620
600 x 120
600 x 133
600 x 140
600 x 160 Various -24 dBm0
61 to 71 dBmC
61 to 71 dBmC
61 to 71 dBmC
61 to 71 dBmC
High tone 480
400
500 Various -17 dBmC
61 to 71 dBmC
61 to 71 dBmC
Dial tone 350 + 440 Steady -13 dBm0
Audible ring tone 440 + 480
440 + 40
500 + 40 2 sec on, 4 sec off
2 sec on, 4 sec off
2 sec on, 4 sec off -19 dBmC
61 to 71 dBmC
61 to 71 dBmC
Line Busy Tone 480 + 620
600 x 120
600 x 133
600 x 140
600 x 160 0.5 sec on, 0.5 sec off
Reorder 480 + 620
600 x 120
600 x 133
600 x 140
600 x 160 0.3 sec on, 0.2 sec off
6A alerting tone 440 2 sec on, followed by 0.5 sec on, every 10 sec
Recorder warning tone 1400 0.5 sec burst every 15 sec
Reverting tone 480 + 620
600 x 120
600 x 133
600 x 140
600 x 160 0.5 sec on, 0.5 sec off -24 dBmC
Deposit coin tone 480 + 620
600 x 120
600 x 133
600 x 140
600 x 160 Steady
Receiver off-hook (analog) 1400 + 2060 + 2450 + 2600 0.1 sec on, 0.1 sec off +5 vu
Receiver off-hook 1400 + 2060 + 2450 + 2600 0.1 sec on, 0.1 sec off +3.9 to -6.0 dBm
Howler 480 Incremented in level
Every 1 sec for 10 sec Up to 40 vu
No such number (crybaby) 200 to 400 Freq. modulated at 1 Hz interrupted every 6 sec for 0.5 sec
Vacant code 480 + 620
600 x 120
600 x 133
600 x 140
600 x 160 0.5 sec on, 0.5 sec off, 0.5 sec on, 1.5 sec off?
Busy verification Tone (Centrex) 440 Initial 1.5 sec followed 0.3 sec every 7.5 to 10 sec -13 dBm0
Busy verification Tone (TSPS) 440 Initial 2 sec followed 0.5 sec every 10 sec -13 dBm0
Call waiting tone 440 Two bursts of 300 ms separated by 10 sec -13 dBm0
Confirmation tone 350 + 440 3 bursts of 300 ms separated by 10 sec -13 dBm0
Indication of camp-on 440 1 sec every attendant releases from loop -13 dBm0
Recall dial tone 350 + 440 3 bursts, 0.1 sec on, sec off then steady -13 dBm0
Data set answer back tone 2025 Steady -13 dBm
Calling card prompt tone 941 + 1477 followed by 440 + 350 60ms -10 dBm0
Class of service 480
400
500 0.5 to 1 sec once
Order tones
Single
480
400
500 0.5 sec
Double
480
400
500 2 short bursts
Triple
480
400
500 3 short bursts
Quad
480
400
500 4 short bursts
Number checking tone 135 Steady
Coin denomination
3 5 cents
1050-1100 (bell) One tap
slot 10 cents
1050-1100 (bell) Two taps
stations 25 cents
800 (gong) One tap
Coin collect tone 480 + 620
600 x 120
600 x 133
600 x 140
600 x 160 Steady
Coin return tone 480
400
500 0.5 to 1 sec once
Coin return test tone 480
400
500 0.5 to 1 sec once
Group busy tone 480 + 620
600 x 120
600 x 133
600 x 140
600 x 160 Steady
Vacant position 480 + 620
600 x 120
600 x 133
600 x 140
600 x 160 Steady
Dial off normal 480 + 620
600 x 120
600 x 133
600 x 140
600 x 160 Steady
Permanent signal 480
400
500 Steady
Warning tone 480
400
500 Steady
Service observing 135 Steady
Proceed to send Tone (IDDD) 480 Steady -22 dBm0
Centralized intercept 1850 500 ms -17 dBm0
ONI order tone 700 + 1100 95 to 250 ms -25 dBm0
Note: Three dots in the pattern mean that the pattern is repeated indefinitely.
Single Frequency In-Band Signaling
SF in-band signaling is widely used in North America. Its most common application is for supervision, such as idle-busy, also called line signaling. It also can be used for dial pulse signaling on trunks. The dynamics of SF signaling requires an understanding of the signal durations and configurations of the E&M circuits, as well as the lead interface arrangements. The following tables show the characteristics of SF signaling, E&M lead configurations, and interface arrangements.
Typical Single Frequency Signaling Characteristics
General
Signaling frequency (tone)
2600 Hz
Idle state transmission
Cut
Idle/break
Tone
Busy/make
No tone
Receiver
Detector bandwidth
+/- 50 Hz @ -7 dBm for E type
+/- 30 Hz @ -7 dBm
Pulsing rate
7.5 to 122 pps
E/M unit
Minimum time for on-hook
33 ms
Minimum no tone for off-hook
55 ms
Input percent break (tone)
38-85 (10 pps)
E lead - open
Idle
- ground
Busy
Originating (loop reverse battery) unit
Minimum tone for idle
40 ms
Minimum no tone for off-hook
43 ms
Minimum output for on-hook
69 ms
Voltage on R lead (-48 V on ring and ground on tip)
On-hook
Voltage on T lead (-48 V on tip and ground on ring)
Off-hook
Terminating (loop reverse battery) unit
Minimum tone for on-hook
90 ms
Minimum no tone for off-hook
60 ms
Minimum output (tone-on)
56 ms
Loop open
On-hook
Loop closed
Off-hook
Transmitter
Low level tone
-36 dBm
High level tone
-24 dBm
High level tone duration
400 ms
Precut
8 ms
Holdover cut
125 ms
Crosscut
625 ms
On hook cut
625 ms
E/M unit
Voltage on M lead
Off-hook (no tone)
Open/ground on M lead
On-hook (tone)
Minimum ground on M lead
21 ms
Minimum voltage on M lead
21 ms
Minimum output tone
21 ms
Minimum no tone
21 ms
Originating (loop reverse battery) unit
Loop current to no tone
19 ms
No loop current to tone
19 ms
Minimum input for tone out
20 ms
Minimum input for no tone out
14 ms
Minimum tone out
51 ms
Minimum no tone out
26 ms
Loop open
On-hook
Loop closed
Off-hook
Terminating (loop) unit
Reverse battery to no tone
19 ms
Normal battery to tone
19 ms
Minimum battery for tone out
25 ms
Minimum reverse battery for no tone
14 ms
Minimum tone out
51 ms
Minimum no tone out
26 ms
Battery on R lead (-48 v)
On-hook
Battery on TY lead (-48 on tip)
Off-hook
Single Frequency Signals Used in E&M Lead Signaling
Calling End Called End
Signal M-Lead E-Lead 2600 Hz 2600 Hz E-Lead M-Lead Signal
Idle Ground Open On On Open Ground Idle
Connect Battery Open Off On Ground Ground Connect
Stop dialing Battery Ground Off Off Ground Battery Stop dialing
Start dialing Battery Open Off On Ground Ground Start dialing
Dial pulsing Ground Open On On Open Ground Dial pulsing
Battery
Off
Ground
Off -hook Battery Ground Off Off Ground Battery Off-hook (answer)
Ring forward Ground Ground On Off Open Battery Ring forward
Battery
Off
Ground
Ringback Battery Open Off On Ground Ground Ringback
Ground
Off
Battery
Flashing Battery Open Off On Ground Ground Flashing
Ground
Off
Battery
On-hook Battery Open Off On Ground Ground On-hook
Disconnect Ground Open On On Open Ground Disconnect
Single Frequency Signals Used in Reverse Battery Tip and Ring Loop Signaling
Calling End Called End
Signal T/R - SF SF - T/R 2600 Hz 2600 Hz T/R - SF SF - T/R Signal
Idle Open Batt-gnd On On Open Batt-gnd Idle
Connect Closure Batt-gnd Off On Closure Batt-gnd Connect
Stop dialing Closure Rev batt-gnd Off Off Closure Rev batt-gnd Stop dialing
Start dialing Closure Batt-gnd Off On Closure Batt-gnd Start dialing
Dial pulsing Open Batt-gnd On On Open Batt-gnd Dial pulsing
Closure
Off
Closure
Off -hook Closure Rev batt-gnd Off Off Closure Rev batt-gnd Off-hook (answer)
Ring forward Open Rev batt-gnd On Off Open Rev batt-gnd Ring forward
Closure
Off
Closure
Ringback Closure Batt-gnd Off On Closure Batt-gnd Ringback
Rev batt-gnd
Off
Rev batt-gnd
Flashing Closure Batt-gnd Off On Closure Batt-gnd Flashing
Rev batt-gnd
Off
Rev batt-gnd
On-hook Closure Batt-gnd Off On Closure Batt-gnd On-hook
Disconnect Open Batt-gnd On On Open Batt-gnd Disconnect
Single Frequency Signals Used for Ringing and Loop-Start Signaling Using Tip and Ring Leads
Call Originating at Central Office End
Signal T/R - SF SF - T/R 2600 Hz 2600 Hz T/R - SF SF - T/R Signal
Idle Gnd-batt Open Off On Gnd-batt Open Idle
Seizure Gnd-batt Open Off On Gnd-batt Open Idle
Ringing Gnd-batt and 20 Hz Open On-off On Gnd-batt and 20 Hz Open Ringing
Off-hook (ring-trip and talk) Gnd-batt Closure Off Off Gnd-batt Closure Off-hook (ring-trip and answer)
On-hook Gnd-batt Closure Off Off Gnd-batt Closure Off-hook
On-hook (hang-up) Gnd-batt Open Off On Gnd-batt Open On-hook (hang-up)
Note: 20 Hz ringing (2 sec on, 4 sec off)
Call Originating at Station End
Signal T/R - SF SF - T/R 2600 Hz 2600 Hz T/R - SF SF - T/R Signal
Idle Open Gnd-batt On Off Open Gnd-batt Idle
Off-hook (seizure) Closure Gnd-batt Off Off Closure Gnd-batt Idle
Start dial Closure Dial tone and gnd-batt Off Off Closure Dial tone and gnd-batt Start dial
Dial pulsing Open-closure Gnd-batt On-off Off Open-closure Gnd-batt Dial pulsing
Waiting answer Closure Audible ring and gnd-batt Off Off Closure Audible ring and gnd-batt Waiting answer
On-hook (talk) Closure Gnd-batt Off Off Closure Gnd-batt Off-hook (answered)
On-hook (hang-up) Open Gnd-batt
Closure On Off Open Gnd-batt On-hook (disconnected)
Off-hook
Single Frequency Signals Used for Ringing and Ground-Start Signaling Using Tip and Ring Leads
Call Originating at Central Office End
Signal T/R - SF SF - T/R 2600 Hz 2600 Hz T/R - SF SF - T/R Signal
Idle Open-batt Batt-batt On On Open-batt
Idle
Seizure Gnd-batt Open On On Gnd-batt
Make-busy
Ringing Gnd-batt and 20 Hz Open On and 20 Hz On Gnd-batt and 20 Hz Open Ringing
Off-hook (ring-trip and talk) Gnd-batt Closure Off Off Gnd-batt Closure Off-hook (ring-trip and answer)
On-hook Gnd-batt Closure On Off Open-batt Closure On-hook
On-hook (hang-up) Gnd-batt Open Off On Gnd-batt Open On-hook (hang-up)
Note: 20 Hz ringing (2 sec on, 4 sec off)
Call Originating at Station End
Signal T/R - SF SF - T/R 2600 Hz 2600 Hz T/R - SF SF - T/R Signal
Idle
Open-batt On On Batt-batt Open-batt Idle
Off-hook (seizure) Ground Open-batt Off On Batt-gnd Open-batt Seizure
Start dial Closure Dial tone and gnd-batt Off Off Closure Dial tone and gnd-batt Start dial
Dial pulsing Open-closure Gnd-batt On-off Off Open-closure Gnd-batt Dial pulsing
Waiting answer Closure Audible ring and gnd-batt Off Off Closure Audible ring and gnd-batt Waiting answer
Off-hook (talk) Closure Gnd-batt Off Off Closure Gnd-batt Off-hook (answered)
On-hook Closure Open-batt On On Batt-batt Open-batt On-hook (disconnected)
On-hook (disconnected)
Closure On Off Open-batt Open-batt On-hook"
CTI论坛: 破解传统语音网
CTI论坛: 破解传统语音网: "局间信令
局间传输通常在数字中继链路上进行，有时也把局间信令称数字中继信令。目前最通用的信令为R2和SS7；国内通常称为１号信令和７号信令。
R2信令
R2信令为一种随路信令（CAS: Channel Associated Signaling）。R2信令是一种基于E1数字网络的国际标准信令，Timeslot 16被预留用来传递其话音通道的信令。但是R2信令并不统一，ITU-T的标准Q.400-Q.490定义了R2信令标准，但不同的国家和地区都有自己的实现方式。
SS7信令
SS7信令为一种共路信令（CCS: Common Channel Signaling）。SS7信令是将呼叫控制信息和其他业务信息通过一张独立的信令网络传输。它比R2信令更高效，更可靠。SS7信令的标准化程度要比R2信令好，但依然存在标准兼容问题。如国内的SS7 称为中国7号信令。
用户端信令
电话用户通常采用两芯双绞线连接到用户交换机。所以用户端信令为面向用户的模拟接口信令，通常也称为模拟接口信令。电话机不同于计算系统，它几乎没有什么智能，只能通过简单的模拟信号来沟通。下面介绍模拟话音的信令。
模拟话音的信令有三种，FXS、FXO和E&M。
FXS（Foreign Exchange Station）
可以理解为计算机通信中的DCE（数据通信设备）接口。通常为电话交换机的用户端口，用来连接具有FXO端口的端设备，如电话机或集团电话。
FXO（Foreign Exchange Office）
可以理解为计算机通信中的DTE（数据终端设备）接口。通常为端设备，连接具有FXS端口的电话交换机设备，如PBX或市话局。
FXS和FXO通过两芯电缆构成环路的断开和闭合与电流信号来完成电话的呼叫控制。
E&M（RecEive and TransMit）
E&M是一种模拟中继信令，主要用于PBX到PBX和PBX到市话局的互连。与FXS/FXO不同，E&M端口之间直接互连，将两台PBX连在一起，通常我们也称为捆绑中继接口（Tie Trunk）。
E&M信令采用４对电缆（８芯）通信。E&M信令有５种类型：TYPE1、TYPE2、 TYPE3、TYPE4和TYPE5，每一种类型有自己的通信方式。
当了解现有的电话网络的连接信令后，企业可以自主的采用多种技术组建自己的电话网，只需要在与市话提供商互连时准备相应的接口。"
局间传输通常在数字中继链路上进行，有时也把局间信令称数字中继信令。目前最通用的信令为R2和SS7；国内通常称为１号信令和７号信令。
R2信令
R2信令为一种随路信令（CAS: Channel Associated Signaling）。R2信令是一种基于E1数字网络的国际标准信令，Timeslot 16被预留用来传递其话音通道的信令。但是R2信令并不统一，ITU-T的标准Q.400-Q.490定义了R2信令标准，但不同的国家和地区都有自己的实现方式。
SS7信令
SS7信令为一种共路信令（CCS: Common Channel Signaling）。SS7信令是将呼叫控制信息和其他业务信息通过一张独立的信令网络传输。它比R2信令更高效，更可靠。SS7信令的标准化程度要比R2信令好，但依然存在标准兼容问题。如国内的SS7 称为中国7号信令。
用户端信令
电话用户通常采用两芯双绞线连接到用户交换机。所以用户端信令为面向用户的模拟接口信令，通常也称为模拟接口信令。电话机不同于计算系统，它几乎没有什么智能，只能通过简单的模拟信号来沟通。下面介绍模拟话音的信令。
模拟话音的信令有三种，FXS、FXO和E&M。
FXS（Foreign Exchange Station）
可以理解为计算机通信中的DCE（数据通信设备）接口。通常为电话交换机的用户端口，用来连接具有FXO端口的端设备，如电话机或集团电话。
FXO（Foreign Exchange Office）
可以理解为计算机通信中的DTE（数据终端设备）接口。通常为端设备，连接具有FXS端口的电话交换机设备，如PBX或市话局。
FXS和FXO通过两芯电缆构成环路的断开和闭合与电流信号来完成电话的呼叫控制。
E&M（RecEive and TransMit）
E&M是一种模拟中继信令，主要用于PBX到PBX和PBX到市话局的互连。与FXS/FXO不同，E&M端口之间直接互连，将两台PBX连在一起，通常我们也称为捆绑中继接口（Tie Trunk）。
E&M信令采用４对电缆（８芯）通信。E&M信令有５种类型：TYPE1、TYPE2、 TYPE3、TYPE4和TYPE5，每一种类型有自己的通信方式。
当了解现有的电话网络的连接信令后，企业可以自主的采用多种技术组建自己的电话网，只需要在与市话提供商互连时准备相应的接口。"
出售 供应信令网关 信令转换器 | [网络设备信息安全-转换器] 中国金融机具网——努力为您带来订单!国内最大的金融设备及安防产品网上市场！
出售 供应信令网关 信令转换器 | [网络设备信息安全-转换器] 中国金融机具网——努力为您带来订单!国内最大的金融设备及安防产品网上市场！: "由于很多厂家的用户交换机不提供七号信令接口，只提供ISDN PRI或中国一号信令接口，因此与局方进行七号信令对接时，便会存在信令接口的问题。FL-300D支持七号信令（SS7）、数字一号信令（DSS1）ISDN PRI、QSIG、中国一号信令（SS1）和V5.2以及环路信令之间的自由互转，从而使得普通的PRI、中国一号信令接口设备能够实现与局方的七号信令（TUP/ISUP）实现对接。
它采用模块化结构，实时嵌入式操作系统，无阻塞交换，最大可处理8路E1。硬件采用进口通信专用器件,功耗低，可靠性高，结构配置灵活, 扩充方便，安装维护简单，是低成本高效率信令转换的最佳方案。
信令网关设备FL-300D已经成功与华为HuaWei、贝尔BELL、阿尔卡特Alcatel、北电Nortel、西门子Siemens、朗讯Lucent、Avaya、中兴ZTE、爱立信Ericsson、奥迪坚Altigen、思科Cisco 等公司的设备实现对接，与国外同类产品相比，有很高的性价比。目前，我公司信令网关设备拥有大量的成功应用案例，已广泛应用于电信、移动、联通等主要通信网络中。
设备特点
提供8个E1接口，可同时4进4出
支持七号信令（ITU-T SS7），完全按照《CCITT七号信令技术规范》和GF001-9001《中国国内电话网No.7信号方式技术规范》。
提供七号信令的第一、第二、第三级MTP及第四级TUP、SCCP、ISUP功能
支持数字一号信令（DSS1），符合ITU-TQ.920-Q.921,Q.930-Q.940
支持中国一号信令（SS1），符合国际GF002-9002的DL信令和MFC信令标准（也可支持不含MFC的中国一号信令，即E&M信令）
支持V5.2信令
全数字时分交换方式，提供512 x 512无阻塞交换
分组、分群交换管理
E1接口支持不定长接续
可任意处理主被叫号码
每个E1接口均可同时拨入和拨出
支持输出原始计费话单功能"
它采用模块化结构，实时嵌入式操作系统，无阻塞交换，最大可处理8路E1。硬件采用进口通信专用器件,功耗低，可靠性高，结构配置灵活, 扩充方便，安装维护简单，是低成本高效率信令转换的最佳方案。
信令网关设备FL-300D已经成功与华为HuaWei、贝尔BELL、阿尔卡特Alcatel、北电Nortel、西门子Siemens、朗讯Lucent、Avaya、中兴ZTE、爱立信Ericsson、奥迪坚Altigen、思科Cisco 等公司的设备实现对接，与国外同类产品相比，有很高的性价比。目前，我公司信令网关设备拥有大量的成功应用案例，已广泛应用于电信、移动、联通等主要通信网络中。
设备特点
提供8个E1接口，可同时4进4出
支持七号信令（ITU-T SS7），完全按照《CCITT七号信令技术规范》和GF001-9001《中国国内电话网No.7信号方式技术规范》。
提供七号信令的第一、第二、第三级MTP及第四级TUP、SCCP、ISUP功能
支持数字一号信令（DSS1），符合ITU-TQ.920-Q.921,Q.930-Q.940
支持中国一号信令（SS1），符合国际GF002-9002的DL信令和MFC信令标准（也可支持不含MFC的中国一号信令，即E&M信令）
支持V5.2信令
全数字时分交换方式，提供512 x 512无阻塞交换
分组、分群交换管理
E1接口支持不定长接续
可任意处理主被叫号码
每个E1接口均可同时拨入和拨出
支持输出原始计费话单功能"
cisco路由器上语音卡fxs、fxo、E&M接口分别可以接什么设备，用的什么连接？ - 中国主流IT|思科华为3COM微软Juniper|认证网络技术专业站-无兄弟不技术(56Cto.Com)
cisco路由器上语音卡fxs、fxo、E&M接口分别可以接什么设备，用的什么连接？ - 中国主流IT|思科华为3COM微软Juniper|认证网络技术专业站-无兄弟不技术(56Cto.Com): "cisco路由器上语音卡fxs、fxo、E&M接口分别可以接什么设备，用的什么连接？
来源：作者： 发布时间：2008-04-28 阅读次数226
在应用中可以简单理解为：
FXO为普通电话机接口，需要远程馈电；
FXS接口为PBX的内线分机接口，向远程馈电；
E&M为专用的一般用在PBX中继线接口。
CISCO提供的说法是：
FXO用于连接PSTN，二线（因为PSTN向用户馈电）
FXS用于连接POT普通电话机，二线（因为电话机需要FXS提供馈电信号）
E&M用于连接PBX，CISCO语音路由器可以设置二线或四线（因为PBX上可能配置E&M接口板）
2/4 线E&M 接口：E&M 中继接口是国际上广泛采用的一种中继接口类型，其信令通道包括发信令的M 线（源于transMit 或Mouth）和收信令的E 线（源于rEceive 或Ear），即通过与话音分开的信令通道E 线和M 线实现交换机与传输设备之间信令转换的一种信令方式。2 线E&M 指音频接口即业务承载通道采用2 线方式，收发在一对平衡线上；4 线E&M 指音频接口采用4 线方式，收发分开，各占用一对平衡线。两种方式简称为2/4 线E&M 中继接口或2/4 E&M 接口。Metro Wave TM MSTP综合接入系统的2/4 线E&M 接口板提供6路2/4 线E&M 接口，每路支持2/4 线音频通道，并提供一根E 信令线和一根M 信令线，利用MSTP传输系统实现异地模拟中继的转接，功能上起业务传输和转发的作用，对信令和话路作透明传送处理。由系统完成话路半永久连接，两侧的2/4 E&M 接口板通过内部定义的协议利用话路时隙来传送E&M 线路信令
F X S是Foreign eXchange Station（外部交换站）的缩写,是标准模拟电话接口。F X S接口用于连接到电话、调制解调器、传真、键盘系统和模拟P B X这样的基本电话设备。F X S接口使用标准R J - 11 2-线接头~~
F X O是Foreign eXchange Off i c e（外部交换局）的缩写，标准电话上的端口是一个F X O端口，与交换机通信。带语音功能的路由器使用P X O端口与P S T N接口。F X O端口就像是一个标准电话一样，要求拨号音来进行呼叫。它使用标准R J - 11模块插口~~
E & M提供挂机/摘机信号，并使干扰最小。它常常用于P B X主干或连接线。有几种类型的E & M存在，每一种规定不同的方法表示P B X和C O交换机之间的摘机条件。E & M信号在2 -线和4 -线实现上支持~~"
来源：作者： 发布时间：2008-04-28 阅读次数226
在应用中可以简单理解为：
FXO为普通电话机接口，需要远程馈电；
FXS接口为PBX的内线分机接口，向远程馈电；
E&M为专用的一般用在PBX中继线接口。
CISCO提供的说法是：
FXO用于连接PSTN，二线（因为PSTN向用户馈电）
FXS用于连接POT普通电话机，二线（因为电话机需要FXS提供馈电信号）
E&M用于连接PBX，CISCO语音路由器可以设置二线或四线（因为PBX上可能配置E&M接口板）
2/4 线E&M 接口：E&M 中继接口是国际上广泛采用的一种中继接口类型，其信令通道包括发信令的M 线（源于transMit 或Mouth）和收信令的E 线（源于rEceive 或Ear），即通过与话音分开的信令通道E 线和M 线实现交换机与传输设备之间信令转换的一种信令方式。2 线E&M 指音频接口即业务承载通道采用2 线方式，收发在一对平衡线上；4 线E&M 指音频接口采用4 线方式，收发分开，各占用一对平衡线。两种方式简称为2/4 线E&M 中继接口或2/4 E&M 接口。Metro Wave TM MSTP综合接入系统的2/4 线E&M 接口板提供6路2/4 线E&M 接口，每路支持2/4 线音频通道，并提供一根E 信令线和一根M 信令线，利用MSTP传输系统实现异地模拟中继的转接，功能上起业务传输和转发的作用，对信令和话路作透明传送处理。由系统完成话路半永久连接，两侧的2/4 E&M 接口板通过内部定义的协议利用话路时隙来传送E&M 线路信令
F X S是Foreign eXchange Station（外部交换站）的缩写,是标准模拟电话接口。F X S接口用于连接到电话、调制解调器、传真、键盘系统和模拟P B X这样的基本电话设备。F X S接口使用标准R J - 11 2-线接头~~
F X O是Foreign eXchange Off i c e（外部交换局）的缩写，标准电话上的端口是一个F X O端口，与交换机通信。带语音功能的路由器使用P X O端口与P S T N接口。F X O端口就像是一个标准电话一样，要求拨号音来进行呼叫。它使用标准R J - 11模块插口~~
E & M提供挂机/摘机信号，并使干扰最小。它常常用于P B X主干或连接线。有几种类型的E & M存在，每一种规定不同的方法表示P B X和C O交换机之间的摘机条件。E & M信号在2 -线和4 -线实现上支持~~"
请问EM接口中的信令接口的实现方式 爱问知识人
请问EM接口中的信令接口的实现方式 爱问知识人: "朋友你好，说白了EM中继接口就是把话音通道和信号通道分开,你没必要去测它！！因此,交换机也常用EM中继接口以适应不同的复用方式和不同信号方式的传输线路.概括而言是为了适应不同信号方式的一种硬件机制.所以你把它们了解清楚了，问题就可以解决了。。EM接口一种常见的交换机接口.它分为以下几种模式:
模式
信令接口
双音频发号(DTMF)
双频互控发号(MFC)
两线EM
1E1M
2E2M
1E1M
2E2M
四线EM
1E1M
2E2M
1E1M
2E2M
EM接口是一种模拟接口.在它上面,信息与接续信令是分开的.信令采用20 mA电流环的方式.其中,信息线可约定为AB线(两线EM),ABCD线(四线EM);信令线为E线/M线(两线EM),E0 E1线/ M0 M1线(四线EM).在进行EM对接时必须采用相同的发号方式,相同的信令接口方式.而且,E线与对方M线对接.
通利的EM板
通利的EM板分为两种类型,DTMF和MFC.对两线/四线以及1E1M/2E2M的选择在板上设置即可.
通利EM(DTMF)示意图如下:
说明:
①LED1:通讯电源指示灯.
②RS232监视接口:使用19200kB/s的波特率与计算机的串行口相连.
③EM1,EM2,EM3,EM4:4乘八线EM接口.
④J10:EM板的设置开关.使用见说明.
⑤J1:源控与非源控选择.
⑥J103,J104,……J403,J404:两线四线的跳线选择.
⑦Line1,Line2,Line3,Line4:EM接续指示灯.当使用为1E1M时,平常灭,占用长亮;当使用
为2E2M时,平常长亮,占用时灭.
通利EM板具体使用说明:
LED1:通讯电源指示灯
当通讯正常时灯快闪(与初始上电时相比).
RS232监视接口
RS232监视接口输出的是TTL电平,计算机串行口使用的是RS232电平,当需要监视时,需用一个电平'转换器'.在电脑中可以使用超级终端进行监视.超级终端的设置为:19200KB/S,数据位8位,奇偶校验无,停止位1,流量控制无,Windows98中如下图所示:
设置后可跟踪到如下数据:
-->Hold On TS=72 在72时隙收到摘机命令
<++Station:2 TS=72 发摘机到对方EM板
<++Mode: Hold On TS=72 发摘机到对方EM板
Ear:fe 收到对方的状态改变
++>Station:2 TS=72 收到对方的状态改变
-->CMD: 9 72 b9 0 收到CPU板的主叫号码
CPU Bureau Code=9 TS=72
CMD: 9 72 b1 0 收到CPU板的主叫号码
CPU Bureau Code=1 TS=72
-->CMD: 9 72 b1 0
CPU Bureau Code=1 TS=72 收到CPU板的主叫号码
-->CMD: 9 72 b1 0
CPU Bureau Code=1 TS=72 收到CPU板的主叫号码
-->CMD: 9 72 b2 0
CPU Bureau Code=2 TS=72 收到CPU板的主叫号码
-->CMD: 9 72 ff 0 收到主叫号码结束
-->CMD: 9 72 39 0
CPU Call Code=9 TS=72 Code_Count=1 Code_Send=0
Send Code=9 72
-->CMD: 9 72 31 0 收到CPU板'呼叫号码'
CPU Call Code=1 TS=72 Code_Count=2 Code_Send=1
Send Code=1 72
-->CMD: 9 72 31 0 收到CPU板'呼叫号码'
CPU Call Code=1 TS=72 Code_Count=3 Code_Send=2
Send Code=1 72
-->CMD: 9 72 31 0 收到CPU板'呼叫号码'
CPU Call Code=1 TS=72 Code_Count=4 Code_Send=3
Send Code=1 72
-->CMD: 9 72 33 0 收到CPU板'呼叫号码'
CPU Call Code=3 TS=72 Code_Count=5 Code_Send=4
Send Code=3 72
<--CMD: 14 72 1 0 中继容错
CMD: 8 72 0 0
-->Hang Off TS=72
Ear:ff
++>Station:3 TS=72
<--CMD: 11 72 ff ff
表示:CPU下发的命令
表示:EM板收到对方EM板的命令
<++表示:EM板发到对方EM板的命令
八线EM接口
EM1,EM2,EM3,EM4为4个八线EM接口.每个EM接口如下图:
其中,AB线为语音发送线;CD为语音接收线;(两线EM使用CD线作为语音线).E0E1为信令接收,M0M1为信令发送.对RJ45头来说,1到8芯为E1 M1 E0M0 CDAB.一般四线1E1M接E0M0 CDAB.根据对方提供(必须由对方提供)对接情况为:
我方:(接)对方:
语音发————语音收
语音收————语音发
M0 ————E0
E0 ————M0
M1 ————E1
E1 ————M1
J10-DIP开关设置
J10-DIP为软件开关设置,为了正常的工作,必需正确设置.规则:当DIP开关打到ON为'0',相对位置为'1'.开关靠近EM接口端为第一个开关.
位置
功能说明
选择说明
1
设置参数扫描时间
0:3s扫描一次
1:120s扫描一次(正常使用应选择该选项)
2
3
线路使用情况选择
00:使用1路
01:使用2路
10:使用3路
11:使用4路
4
RS232监视输出选择
0:状态不输出(不监视时建议采用,提EM接续高效率)
1:状态输出
5
1E1M/2E2M选择
0:选择1E1M
1:选择2E2M
6
1E1M 时:
Cisco/Philips
0:选择Cisco
1:选择Philips
7
NC
未使用
8
NC
未使用
源控与非源控选择
对于大多数交换机,都可采用非源控的方式,即:接地为'0',悬空为'1'( 接地为有信号,悬空为无信号).个别的交换机需要源控的方式,即:接地为'0',接-48V为'1'( 接地为有信号,接-48V为无信号).
通利EM板的J1便是源控/非源控选择,当短路为源控,断开为非源控.
四线/两线EM选择
通常EM多为四线,在语音上收发分开(如接卫星连路,2M分离连路).也有语音收发和在一起的情况(如近距离自组内网).当采用四线EM时参见下图:(一路)
当采用两线EM时参见下图:(一路)
系统设置
对3.36的CPU板,需设置以下一指令:
**00*000000**拍叉
**45*1*EM中继所在内码1* EM中继所在内码2*......
为保证交换机与对方设备共地,应把交换机备电地与对方EM接口设备地线对连."
模式
信令接口
双音频发号(DTMF)
双频互控发号(MFC)
两线EM
1E1M
2E2M
1E1M
2E2M
四线EM
1E1M
2E2M
1E1M
2E2M
EM接口是一种模拟接口.在它上面,信息与接续信令是分开的.信令采用20 mA电流环的方式.其中,信息线可约定为AB线(两线EM),ABCD线(四线EM);信令线为E线/M线(两线EM),E0 E1线/ M0 M1线(四线EM).在进行EM对接时必须采用相同的发号方式,相同的信令接口方式.而且,E线与对方M线对接.
通利的EM板
通利的EM板分为两种类型,DTMF和MFC.对两线/四线以及1E1M/2E2M的选择在板上设置即可.
通利EM(DTMF)示意图如下:
说明:
①LED1:通讯电源指示灯.
②RS232监视接口:使用19200kB/s的波特率与计算机的串行口相连.
③EM1,EM2,EM3,EM4:4乘八线EM接口.
④J10:EM板的设置开关.使用见说明.
⑤J1:源控与非源控选择.
⑥J103,J104,……J403,J404:两线四线的跳线选择.
⑦Line1,Line2,Line3,Line4:EM接续指示灯.当使用为1E1M时,平常灭,占用长亮;当使用
为2E2M时,平常长亮,占用时灭.
通利EM板具体使用说明:
LED1:通讯电源指示灯
当通讯正常时灯快闪(与初始上电时相比).
RS232监视接口
RS232监视接口输出的是TTL电平,计算机串行口使用的是RS232电平,当需要监视时,需用一个电平'转换器'.在电脑中可以使用超级终端进行监视.超级终端的设置为:19200KB/S,数据位8位,奇偶校验无,停止位1,流量控制无,Windows98中如下图所示:
设置后可跟踪到如下数据:
-->Hold On TS=72 在72时隙收到摘机命令
<++Station:2 TS=72 发摘机到对方EM板
<++Mode: Hold On TS=72 发摘机到对方EM板
Ear:fe 收到对方的状态改变
++>Station:2 TS=72 收到对方的状态改变
-->CMD: 9 72 b9 0 收到CPU板的主叫号码
CPU Bureau Code=9 TS=72
CMD: 9 72 b1 0 收到CPU板的主叫号码
CPU Bureau Code=1 TS=72
-->CMD: 9 72 b1 0
CPU Bureau Code=1 TS=72 收到CPU板的主叫号码
-->CMD: 9 72 b1 0
CPU Bureau Code=1 TS=72 收到CPU板的主叫号码
-->CMD: 9 72 b2 0
CPU Bureau Code=2 TS=72 收到CPU板的主叫号码
-->CMD: 9 72 ff 0 收到主叫号码结束
-->CMD: 9 72 39 0
CPU Call Code=9 TS=72 Code_Count=1 Code_Send=0
Send Code=9 72
-->CMD: 9 72 31 0 收到CPU板'呼叫号码'
CPU Call Code=1 TS=72 Code_Count=2 Code_Send=1
Send Code=1 72
-->CMD: 9 72 31 0 收到CPU板'呼叫号码'
CPU Call Code=1 TS=72 Code_Count=3 Code_Send=2
Send Code=1 72
-->CMD: 9 72 31 0 收到CPU板'呼叫号码'
CPU Call Code=1 TS=72 Code_Count=4 Code_Send=3
Send Code=1 72
-->CMD: 9 72 33 0 收到CPU板'呼叫号码'
CPU Call Code=3 TS=72 Code_Count=5 Code_Send=4
Send Code=3 72
<--CMD: 14 72 1 0 中继容错
CMD: 8 72 0 0
-->Hang Off TS=72
Ear:ff
++>Station:3 TS=72
<--CMD: 11 72 ff ff
表示:CPU下发的命令
表示:EM板收到对方EM板的命令
<++表示:EM板发到对方EM板的命令
八线EM接口
EM1,EM2,EM3,EM4为4个八线EM接口.每个EM接口如下图:
其中,AB线为语音发送线;CD为语音接收线;(两线EM使用CD线作为语音线).E0E1为信令接收,M0M1为信令发送.对RJ45头来说,1到8芯为E1 M1 E0M0 CDAB.一般四线1E1M接E0M0 CDAB.根据对方提供(必须由对方提供)对接情况为:
我方:(接)对方:
语音发————语音收
语音收————语音发
M0 ————E0
E0 ————M0
M1 ————E1
E1 ————M1
J10-DIP开关设置
J10-DIP为软件开关设置,为了正常的工作,必需正确设置.规则:当DIP开关打到ON为'0',相对位置为'1'.开关靠近EM接口端为第一个开关.
位置
功能说明
选择说明
1
设置参数扫描时间
0:3s扫描一次
1:120s扫描一次(正常使用应选择该选项)
2
3
线路使用情况选择
00:使用1路
01:使用2路
10:使用3路
11:使用4路
4
RS232监视输出选择
0:状态不输出(不监视时建议采用,提EM接续高效率)
1:状态输出
5
1E1M/2E2M选择
0:选择1E1M
1:选择2E2M
6
1E1M 时:
Cisco/Philips
0:选择Cisco
1:选择Philips
7
NC
未使用
8
NC
未使用
源控与非源控选择
对于大多数交换机,都可采用非源控的方式,即:接地为'0',悬空为'1'( 接地为有信号,悬空为无信号).个别的交换机需要源控的方式,即:接地为'0',接-48V为'1'( 接地为有信号,接-48V为无信号).
通利EM板的J1便是源控/非源控选择,当短路为源控,断开为非源控.
四线/两线EM选择
通常EM多为四线,在语音上收发分开(如接卫星连路,2M分离连路).也有语音收发和在一起的情况(如近距离自组内网).当采用四线EM时参见下图:(一路)
当采用两线EM时参见下图:(一路)
系统设置
对3.36的CPU板,需设置以下一指令:
**00*000000**拍叉
**45*1*EM中继所在内码1* EM中继所在内码2*......
为保证交换机与对方设备共地,应把交换机备电地与对方EM接口设备地线对连."
E&M线序问题 - VoIP - H3C全球技术服务论坛 - Powered by Discuz!
E&M线序问题 - VoIP - H3C全球技术服务论坛 - Powered by Discuz!: "E&M线序问题
针对E&M模块的简要说明：
1.在实际连接中，要保证路由器和PBX交换机共地，即双方的地线要接在一起接地。
2.E&M模块用的是RJ-45标准的线，所以对线序有要求：
E&M模块支持BELL I ，II，III，V类交换机类型，支持2线，4线音频两种方式。一般多为4 线音频方式。
模块上RJ-45插座管脚顺序从插座往里看，缺口朝下，从左往右为1-8脚。
路由器模块侧：RJ-45管脚信号： 对应交换机信号:
1--------------------SB(负电源)----------------------空
2--------------------E-----------------------------------M
3--------------------RING0----------------------------RING0
4--------------------RING1----------------------------RING1
5--------------------TIP1------------------------------TIP1
6--------------------TIP0------------------------------TIP0
7--------------------M----------------------------------E
8--------------------SG(负电源地)-------------------空
3,6 为一对线，为发信号，对应对方的收
4,5为一对线，为收信号，对应对方的发
2对应对方的M信号
7对应对方的E信号"
针对E&M模块的简要说明：
1.在实际连接中，要保证路由器和PBX交换机共地，即双方的地线要接在一起接地。
2.E&M模块用的是RJ-45标准的线，所以对线序有要求：
E&M模块支持BELL I ，II，III，V类交换机类型，支持2线，4线音频两种方式。一般多为4 线音频方式。
模块上RJ-45插座管脚顺序从插座往里看，缺口朝下，从左往右为1-8脚。
路由器模块侧：RJ-45管脚信号： 对应交换机信号:
1--------------------SB(负电源)----------------------空
2--------------------E-----------------------------------M
3--------------------RING0----------------------------RING0
4--------------------RING1----------------------------RING1
5--------------------TIP1------------------------------TIP1
6--------------------TIP0------------------------------TIP0
7--------------------M----------------------------------E
8--------------------SG(负电源地)-------------------空
3,6 为一对线，为发信号，对应对方的收
4,5为一对线，为收信号，对应对方的发
2对应对方的M信号
7对应对方的E信号"
E1/CE1/T1/PRI/BRI知识介绍和配置(页 1) - 思科技术 - 51CTO技术论坛
E1/CE1/T1/PRI/BRI知识介绍和配置(页 1) - 思科技术 - 51CTO技术论坛: "E1/CE1/T1/PRI/BRI知识介绍和配置
E1简介：
① 一条E1是2.048M的链路，用PCM编码。
② 一个E1的帧长为256个bit,分为32个时隙，一个时隙为8个bit。
③ 每秒有8k个E1的帧通过接口，即8K*256=2048kbps。
④ 每个时隙在E1帧中占8bit，8*8k=64k，即一条E1中含有32个64K。
E1帧结构
E1分为有成帧，成复帧与不成帧三种方式，在成帧的E1中第0时隙用于传输帧同步数据，其余31个时隙可以用于传输有效数据；在成复帧的E1中，除了第0时隙外，第16时隙是用于传输信令的，只有第1到15，第17到第31共30个时隙可用于传输有效数据；而在不成帧的E1中，所有32个时隙都可用于传输有效数据。
E1信道的帧结构简述
在E1信道中，8bit组成一个时隙（TS），由32个时隙组成了一个帧（F），16个帧组成一个复帧（MF）。在一个帧中，TS0 主要用于传送帧定位信号（FAS）、CRC-4（循环冗余校验）和对端告警指示，TS16主要传送随路信令（CAS）、复帧定位信号和复帧对端告警指示，TS1至TS15和TS17至TS31共30个时隙传送话音或数据等信息。我们称TS1至TS15和TS17至TS31为“净荷”，TS0和TS16为“开销”。如果采用带外公共信道信令（CCS），TS16就失去了传送信令的用途，该时隙也可用来传送信息信号，这时帧结构的净荷为TS1至TS31，开销只有TS0了。
由PCM编码介绍E1：
由PCM编码中E1的时隙特征可知，E1共分32个时隙TS0-TS31。 每个时隙为64K，其中TS0为被帧同步码，Si、Sa4、Sa5、sa6、Sa7、A比特占用，若系统运用了CRC校验，则Si比特位置改传CRC校验码。TS16为信令时隙，当使用到信令(共路信令或随路信令)时，该时隙用来传输信令，用户不可用来传输数据。所以2M的PCM码型有
① PCM30：PCM30用户可用时隙为30个，TS1-TS15,TS17-TS31。TS16传送信令，无CRC校验。
② PCM31：PCM30用户可用时隙为31个，TS1-TS15,TS16-TS31。TS16不传送信令，无CRC校验。
③ PCM30C：PCM30用户可用时隙为30个，TS1-TS15,TS17-TS31。TS16传送信令，有CRC校验。
④ PCM31C：PCM30用户可用时隙为31个，TS1-TS15,TS16-TS31。TS16不传送信令，有CRC校验。
CE1,就是把2M的传输分成了30个64K的时隙，一般写成N*64，你可以利用其中的几个时隙，也就是只利用n个64K，必须接在ce1/pri上。
CE1----最多可有31个信道承载数据 timeslots 1----31 timeslots 0 传同步。
E1接口：
G．703非平衡的75 ohm，平衡的120 ohm2种接口
使用E1的三种方法：
1.将整个2M用作一条链路，如DDN 2M；
2.将2M用作若干个64k及其组合，如128K，256K等，这就是CE1；
3.在用作语音交换机的数字中继时，这也是E1最本来的用法，是把一条E1作为32个64K来用，但是时隙0和时隙15是用作signaling即信令的，所以一条E1可以传30路话音。PRI就是其中的最常用的一种接入方式，标准叫PRA信令。
用2611等的广域网接口卡，经V.35-G.703转换器接E1线。这样的成本应该比E1卡低的。目前DDN的2M速率线路通常是经HDSL线路拉至用户侧。E1可由传输设备出的光纤拉至用户侧的光端机提供E1服务。
E1的使用注意事项：
E1接口对接时，双方的E1不能有信号丢失/帧失步/复帧失步/滑码告警，但是双方在E1接口参数上必须完全一致，因为个别特性参数的不一致，不会在指示灯或者告警台上有任何告警，但是会造成数据通道的不通/误码/滑码/失步等情况。这些特性参数主要有；阻抗/ 帧结构/CRC4校验，阻有75ohm和120ohm两种，帧结构有PCM31/PCM30/不成帧三种；在新桥节点机中将PCM31和PCM30分别描述为CCS和CAS，对接时要告诉网管人员选择CCS，是否进行CRC校验可以灵活选择，关键要双方一致，这样采可保证物理层的正常。
E1常见问题
E1 与 CE1是由谁控制，电信还是互连的两侧的用户设备？用户侧肯定要求支持他们，电信又是如何分别实现的。
答：首先由电信决定，电信可提供E1和CE1两种线路，但一般用户的E1线路都是CE1，除非你特别要只用E1，然后才由你的设备所决定，CE1可以当E1用，但E1却不可以作CE1。
CE1 是32个时隙都可用是吧？
答：CE1的0和16时隙不用,0是传送同步号,16传送控制命令,实际能用的只有30个时隙1-15，16-30
E1/CE1/PRI又是如何区分的和通常说的2M的关系。和DDN的2M又如何关联啊？
答：E1 和CE1 都是E1线路标准，PRI是ISDN主干线咱，30B+D，DDN的2M是透明线路你可以他上面跑任何协议。
E1和CE1的区别，当然可不可分时隙了。
E1/CE1/PRI与信令、时隙的关系
答：E1，CE1，都是32时隙，30时隙，0、16分别传送同步信号和控制信今，PRI采用30B+D ，30B传数据，D信道传送信令， E1都是CAS结构，叫带内信令，PRI信令与数据分开传送，即带外信令。
CE1可否接E1。
答：CE1 和E1 当然可以互联。但CE1必需当E1用，即不可分时隙使用。
为实现利用CE1实现一点对多点互连，此时中心肯定是2M了，各分支速率是N*64K28k
56k Modem >56k
Fractional T1/ISDN 128 K
Frame Relay 1.5 M
DSL / ADSL 1.5 M
T-1/DS-1 America 1.5 M
SDSL 2.0 M
E-1/DS-1 Europe 2.0 M
Fractional T3 3.0 M
Wireless 4.0 M
T-2/DS-2 6.3 M
E-2 Europe 8.4 M
Cable 10.0 M
Thin Ethernet 10.0 M
E-3 Europe 34.3 M
T-3/DS-3 America 44.7 M
OC-1/STS-1 51.8 M
CDDI,FDDI 100.0 M
OC-3/STS-3 155.5 M
OC-12/STS-12 622.0 M
OC-24 1.2 G
OC-48/STS-48 2.4 G
OC-192 10 G
OC-255 13.2 G
路由器CE1/PRI接口配置命令
1. channel-group
将 CE1/PRI 捆绑为channel-group。
channel-group channel-group timeslots { number | number1-number2 } [,number | number1-number2 ... ]
no channel-group channel-group
【参数说明】
channel-group为该CE1/PRI接口上的新接口的索引号，范围0~30。
指定捆绑时隙时，可以指定单个时隙number，也可以指定时隙范围number1 - number2，number为时隙编号，范围1~31。
【缺省情况】
没有捆绑任何channel-group。
【命令模式】
CE1/PRI接口配置模式
【使用指南】
CE1/PRI接口在物理上分为31个时隙，对应编号为0~30，可以任意地将全部时隙分成若干组，每组时隙捆绑以后作为一个接口（channel-group）使用，其逻辑特性与同步串口相同，支持PPP、帧中继、LAPB和X.25等链路层协议，支持IP和IPX等网络协议。
no channel-group 命令取消相应的时隙捆绑。
【举例】
将E1接口的1、2、5、10-15和18时隙捆绑为1号channel-group。
Quidway(config-if-e0)#channel-group 1 timeslots 1,2,5,10-15,18
【相关命令】
pri-group
2. clock
设置CE1/PRI接口的时钟方式。
clock { DCE | lineclock }
no clock
【缺省情况】
CE1/PRI接口的缺省时钟方式为DTE方式（lineclock）。
【命令模式】
CE1/PRI 接口配置模式
【使用指南】
当CE1/PRI作为同步接口使用时，同样有DTE和DCE两种工作方式，也需要选择线路时钟。当两台路由器的CE1/PRI接口直接相连时，必须使两端分别工作在DTE和DCE方式；当路由器的CE1/PRI接口与交换机连接时，交换机为DCE设备，而路由器的CE1/PRI接口需工作在DTE方式（lineclock）。
no clock命令恢复缺省的时钟方式。
【举例】
设置CE1/PRI接口的时钟方式为DCE。
Quidway(config-if-e0)#clock DCE
【相关命令】
channel-group，pri-group
3. framing
设置CE1/PRI接口的帧校验方式。
framing { crc4 | no- 4 }
no framing
【缺省情况】
CE1/PRI接口缺省为不校验。
【命令模式】
CE1/PRI接口配置模式
【使用指南】
CE1/PRI接口支持对物理帧的4字节CRC校验，no framing命令恢复缺省不校验。
【举例】
设置CE1/PRI接口对物理帧进行4字节CRC校验。
Quidway(config-if-e0)#framing crc4
4. linecode
设置CE1/PRI接口的线路编解码格式。
linecode { ami | hdb3 }
no linecode
【缺省情况】
CE1/PRI接口的缺省编解码格式为HDB3格式。
【命令模式】
CE1/PRI接口配置模式
【使用指南】
no linecode命令恢复缺省的编解码格式。
【举例】
设置CE1/PRI接口的缺省编解码格式为AMI格式。
Quidway#(config-if-E0)#linecode ami
5. loopback
允许或禁止CE1/PRI对内自环和对外回波。
[ no ] loopback
【缺省情况】
禁止CE1/PRI接口对内自环和对外回波。
【命令模式】
CE1/PRI接口配置模式
【使用指南】
只有在进行某些特殊功能测试时，才将CE1/PRI 接口设为对内自环和对外回波。
【举例】
允许CE1/PRI接口e0对内自环和对外回波。
Quidway(config-if-e0)#loopback
6. pri-group
将CE1/PRI接口时隙捆绑为pri-group。
pri-group [ timeslots { number | number1-number2 } [,number | number1-number2 ... ] ]
no pri-group
【参数说明】
捆绑时隙时，既可以指定单个时隙number，也可以指定时隙范围number1-number2 。
注意：由于时隙16作为D信道用于传输信令，不能作为捆绑时隙。
【缺省情况】
缺省没有创建PRI捆绑。
【命令模式】
CE1/PRI接口配置模式
【使用指南】
CE1/PRI接口只能捆绑生成一个pri-group使用，其逻辑特性与ISDN拨号口相同，支持PPP链路层协议，支持IP和IPX等网络协议，可以配置DDR等参数。
no pri-group取消相应的时隙捆绑。
pri-group的索引号固定为15。
【举例】
将CE1/PRI接口的1，2，8~12等时隙捆绑为pri-group。
Quidway(config-if-E0)#pri-group timeslots 1,2,8-12
【相关命令】
channel-group
破解传统语音网
随着IP网络成为企业经营不可缺少的部分，越来越多的企业将电话网络迁移到IP网络之上。通常我们把通过IP网络传输的电话称为IP电话。本文从企业IT管理者的视角，全面介绍从传统电话网络开始、企业内部电话由PBX向IP电话发展的过程，同时也展示了未来企业电话发展的前景。
传统电话网路通常称作公共电话交换网（PSTN: Public Switch Telephone Network）。传统电话网络从最原始的依靠接线员的人工交换网络到今天的数字程控电话网络已经发展了一百多年了。传统电话网络的基本原理就是采用某种连接技术在两部电话之间建立一条可传输话音的电路。随着数字通信技术的发展，电话网发展到今天内部结构已经数字化了。现代数字电话交换网络就是基于TDM（时分复用）技术通过呼叫控制信令在两个电话机之间建立一条独享的电路。
为达到扩展性和可靠性，现代电话网络有严格的层次化结构：通常称为用户端局和汇接局。用户端局通常配置用户交换机，主要连接家庭用户和企业用户，并上连到汇接局。用户接入的方式通常采用模拟双绞线；部分大客户也有采用数字链路接入，通常是n×E1链路。汇接局是汇聚用户局的连接，并与其他汇接局连接，连接链路通常是n×E1或STM1。
从传统上讲，一个企业的电话网络由一个个孤立的电话系统。如下图所示。企业总部同下面分公司的电话通过电话服务商来提供。企业要付昂贵的电话费，并且企业能否有一些特殊的电话需求很大程度受服务商的限制。
为突破这种限制，企业完全可以组建自己的电话网络系统。首先我们要了解现有的电话系统。同计算机网络一样，电话网络也有自己的网络通信描述名词。为了更好地理解电话网络，首先介绍传统电话网络的一些信令。
基于层次化结构，电话的呼叫控制信令可分为局间信令和用户信令。
局间信令
局间传输通常在数字中继链路上进行，有时也把局间信令称数字中继信令。目前最通用的信令为R2和SS7；国内通常称为１号信令和７号信令。
R2信令
R2信令为一种随路信令（CAS: Channel Associated Signaling）。R2信令是一种基于E1数字网络的国际标准信令，Timeslot 16被预留用来传递其话音通道的信令。但是R2信令并不统一，ITU-T的标准Q.400-Q.490定义了R2信令标准，但不同的国家和地区都有自己的实现方式。
SS7信令
SS7信令为一种共路信令（CCS: Common Channel Signaling）。SS7信令是将呼叫控制信息和其他业务信息通过一张独立的信令网络传输。它比R2信令更高效，更可靠。SS7信令的标准化程度要比R2信令好，但依然存在标准兼容问题。如国内的SS7 称为中国7号信令。
用户端信令
电话用户通常采用两芯双绞线连接到用户交换机。所以用户端信令为面向用户的模拟接口信令，通常也称为模拟接口信令。电话机不同于计算系统，它几乎没有什么智能，只能通过简单的模拟信号来沟通。下面介绍模拟话音的信令。
模拟话音的信令有三种，FXS、FXO和E&M。
FXS（Foreign Exchange Station）
可以理解为计算机通信中的DCE（数据通信设备）接口。通常为电话交换机的用户端口，用来连接具有FXO端口的端设备，如电话机或集团电话。
FXO（Foreign Exchange Office）
可以理解为计算机通信中的DTE（数据终端设备）接口。通常为端设备，连接具有FXS端口的电话交换机设备，如PBX或市话局。
FXS和FXO通过两芯电缆构成环路的断开和闭合与电流信号来完成电话的呼叫控制。
E&M（RecEive and TransMit）
E&M是一种模拟中继信令，主要用于PBX到PBX和PBX到市话局的互连。与FXS/FXO不同，E&M端口之间直接互连，将两台PBX连在一起，通常我们也称为捆绑中继接口（Tie Trunk）。
E&M信令采用４对电缆（８芯）通信。E&M信令有５种类型：TYPE1、TYPE2、 TYPE3、TYPE4和TYPE5，每一种类型有自己的通信方式。
当了解现有的电话网络的连接信令后，企业可以自主的采用多种技术组建自己的电话网，只需要在与市话提供商互连时准备相应的接口。"
E1简介：
① 一条E1是2.048M的链路，用PCM编码。
② 一个E1的帧长为256个bit,分为32个时隙，一个时隙为8个bit。
③ 每秒有8k个E1的帧通过接口，即8K*256=2048kbps。
④ 每个时隙在E1帧中占8bit，8*8k=64k，即一条E1中含有32个64K。
E1帧结构
E1分为有成帧，成复帧与不成帧三种方式，在成帧的E1中第0时隙用于传输帧同步数据，其余31个时隙可以用于传输有效数据；在成复帧的E1中，除了第0时隙外，第16时隙是用于传输信令的，只有第1到15，第17到第31共30个时隙可用于传输有效数据；而在不成帧的E1中，所有32个时隙都可用于传输有效数据。
E1信道的帧结构简述
在E1信道中，8bit组成一个时隙（TS），由32个时隙组成了一个帧（F），16个帧组成一个复帧（MF）。在一个帧中，TS0 主要用于传送帧定位信号（FAS）、CRC-4（循环冗余校验）和对端告警指示，TS16主要传送随路信令（CAS）、复帧定位信号和复帧对端告警指示，TS1至TS15和TS17至TS31共30个时隙传送话音或数据等信息。我们称TS1至TS15和TS17至TS31为“净荷”，TS0和TS16为“开销”。如果采用带外公共信道信令（CCS），TS16就失去了传送信令的用途，该时隙也可用来传送信息信号，这时帧结构的净荷为TS1至TS31，开销只有TS0了。
由PCM编码介绍E1：
由PCM编码中E1的时隙特征可知，E1共分32个时隙TS0-TS31。 每个时隙为64K，其中TS0为被帧同步码，Si、Sa4、Sa5、sa6、Sa7、A比特占用，若系统运用了CRC校验，则Si比特位置改传CRC校验码。TS16为信令时隙，当使用到信令(共路信令或随路信令)时，该时隙用来传输信令，用户不可用来传输数据。所以2M的PCM码型有
① PCM30：PCM30用户可用时隙为30个，TS1-TS15,TS17-TS31。TS16传送信令，无CRC校验。
② PCM31：PCM30用户可用时隙为31个，TS1-TS15,TS16-TS31。TS16不传送信令，无CRC校验。
③ PCM30C：PCM30用户可用时隙为30个，TS1-TS15,TS17-TS31。TS16传送信令，有CRC校验。
④ PCM31C：PCM30用户可用时隙为31个，TS1-TS15,TS16-TS31。TS16不传送信令，有CRC校验。
CE1,就是把2M的传输分成了30个64K的时隙，一般写成N*64，你可以利用其中的几个时隙，也就是只利用n个64K，必须接在ce1/pri上。
CE1----最多可有31个信道承载数据 timeslots 1----31 timeslots 0 传同步。
E1接口：
G．703非平衡的75 ohm，平衡的120 ohm2种接口
使用E1的三种方法：
1.将整个2M用作一条链路，如DDN 2M；
2.将2M用作若干个64k及其组合，如128K，256K等，这就是CE1；
3.在用作语音交换机的数字中继时，这也是E1最本来的用法，是把一条E1作为32个64K来用，但是时隙0和时隙15是用作signaling即信令的，所以一条E1可以传30路话音。PRI就是其中的最常用的一种接入方式，标准叫PRA信令。
用2611等的广域网接口卡，经V.35-G.703转换器接E1线。这样的成本应该比E1卡低的。目前DDN的2M速率线路通常是经HDSL线路拉至用户侧。E1可由传输设备出的光纤拉至用户侧的光端机提供E1服务。
E1的使用注意事项：
E1接口对接时，双方的E1不能有信号丢失/帧失步/复帧失步/滑码告警，但是双方在E1接口参数上必须完全一致，因为个别特性参数的不一致，不会在指示灯或者告警台上有任何告警，但是会造成数据通道的不通/误码/滑码/失步等情况。这些特性参数主要有；阻抗/ 帧结构/CRC4校验，阻有75ohm和120ohm两种，帧结构有PCM31/PCM30/不成帧三种；在新桥节点机中将PCM31和PCM30分别描述为CCS和CAS，对接时要告诉网管人员选择CCS，是否进行CRC校验可以灵活选择，关键要双方一致，这样采可保证物理层的正常。
E1常见问题
E1 与 CE1是由谁控制，电信还是互连的两侧的用户设备？用户侧肯定要求支持他们，电信又是如何分别实现的。
答：首先由电信决定，电信可提供E1和CE1两种线路，但一般用户的E1线路都是CE1，除非你特别要只用E1，然后才由你的设备所决定，CE1可以当E1用，但E1却不可以作CE1。
CE1 是32个时隙都可用是吧？
答：CE1的0和16时隙不用,0是传送同步号,16传送控制命令,实际能用的只有30个时隙1-15，16-30
E1/CE1/PRI又是如何区分的和通常说的2M的关系。和DDN的2M又如何关联啊？
答：E1 和CE1 都是E1线路标准，PRI是ISDN主干线咱，30B+D，DDN的2M是透明线路你可以他上面跑任何协议。
E1和CE1的区别，当然可不可分时隙了。
E1/CE1/PRI与信令、时隙的关系
答：E1，CE1，都是32时隙，30时隙，0、16分别传送同步信号和控制信今，PRI采用30B+D ，30B传数据，D信道传送信令， E1都是CAS结构，叫带内信令，PRI信令与数据分开传送，即带外信令。
CE1可否接E1。
答：CE1 和E1 当然可以互联。但CE1必需当E1用，即不可分时隙使用。
为实现利用CE1实现一点对多点互连，此时中心肯定是2M了，各分支速率是N*64K28k
56k Modem >56k
Fractional T1/ISDN 128 K
Frame Relay 1.5 M
DSL / ADSL 1.5 M
T-1/DS-1 America 1.5 M
SDSL 2.0 M
E-1/DS-1 Europe 2.0 M
Fractional T3 3.0 M
Wireless 4.0 M
T-2/DS-2 6.3 M
E-2 Europe 8.4 M
Cable 10.0 M
Thin Ethernet 10.0 M
E-3 Europe 34.3 M
T-3/DS-3 America 44.7 M
OC-1/STS-1 51.8 M
CDDI,FDDI 100.0 M
OC-3/STS-3 155.5 M
OC-12/STS-12 622.0 M
OC-24 1.2 G
OC-48/STS-48 2.4 G
OC-192 10 G
OC-255 13.2 G
路由器CE1/PRI接口配置命令
1. channel-group
将 CE1/PRI 捆绑为channel-group。
channel-group channel-group timeslots { number | number1-number2 } [,number | number1-number2 ... ]
no channel-group channel-group
【参数说明】
channel-group为该CE1/PRI接口上的新接口的索引号，范围0~30。
指定捆绑时隙时，可以指定单个时隙number，也可以指定时隙范围number1 - number2，number为时隙编号，范围1~31。
【缺省情况】
没有捆绑任何channel-group。
【命令模式】
CE1/PRI接口配置模式
【使用指南】
CE1/PRI接口在物理上分为31个时隙，对应编号为0~30，可以任意地将全部时隙分成若干组，每组时隙捆绑以后作为一个接口（channel-group）使用，其逻辑特性与同步串口相同，支持PPP、帧中继、LAPB和X.25等链路层协议，支持IP和IPX等网络协议。
no channel-group 命令取消相应的时隙捆绑。
【举例】
将E1接口的1、2、5、10-15和18时隙捆绑为1号channel-group。
Quidway(config-if-e0)#channel-group 1 timeslots 1,2,5,10-15,18
【相关命令】
pri-group
2. clock
设置CE1/PRI接口的时钟方式。
clock { DCE | lineclock }
no clock
【缺省情况】
CE1/PRI接口的缺省时钟方式为DTE方式（lineclock）。
【命令模式】
CE1/PRI 接口配置模式
【使用指南】
当CE1/PRI作为同步接口使用时，同样有DTE和DCE两种工作方式，也需要选择线路时钟。当两台路由器的CE1/PRI接口直接相连时，必须使两端分别工作在DTE和DCE方式；当路由器的CE1/PRI接口与交换机连接时，交换机为DCE设备，而路由器的CE1/PRI接口需工作在DTE方式（lineclock）。
no clock命令恢复缺省的时钟方式。
【举例】
设置CE1/PRI接口的时钟方式为DCE。
Quidway(config-if-e0)#clock DCE
【相关命令】
channel-group，pri-group
3. framing
设置CE1/PRI接口的帧校验方式。
framing { crc4 | no- 4 }
no framing
【缺省情况】
CE1/PRI接口缺省为不校验。
【命令模式】
CE1/PRI接口配置模式
【使用指南】
CE1/PRI接口支持对物理帧的4字节CRC校验，no framing命令恢复缺省不校验。
【举例】
设置CE1/PRI接口对物理帧进行4字节CRC校验。
Quidway(config-if-e0)#framing crc4
4. linecode
设置CE1/PRI接口的线路编解码格式。
linecode { ami | hdb3 }
no linecode
【缺省情况】
CE1/PRI接口的缺省编解码格式为HDB3格式。
【命令模式】
CE1/PRI接口配置模式
【使用指南】
no linecode命令恢复缺省的编解码格式。
【举例】
设置CE1/PRI接口的缺省编解码格式为AMI格式。
Quidway#(config-if-E0)#linecode ami
5. loopback
允许或禁止CE1/PRI对内自环和对外回波。
[ no ] loopback
【缺省情况】
禁止CE1/PRI接口对内自环和对外回波。
【命令模式】
CE1/PRI接口配置模式
【使用指南】
只有在进行某些特殊功能测试时，才将CE1/PRI 接口设为对内自环和对外回波。
【举例】
允许CE1/PRI接口e0对内自环和对外回波。
Quidway(config-if-e0)#loopback
6. pri-group
将CE1/PRI接口时隙捆绑为pri-group。
pri-group [ timeslots { number | number1-number2 } [,number | number1-number2 ... ] ]
no pri-group
【参数说明】
捆绑时隙时，既可以指定单个时隙number，也可以指定时隙范围number1-number2 。
注意：由于时隙16作为D信道用于传输信令，不能作为捆绑时隙。
【缺省情况】
缺省没有创建PRI捆绑。
【命令模式】
CE1/PRI接口配置模式
【使用指南】
CE1/PRI接口只能捆绑生成一个pri-group使用，其逻辑特性与ISDN拨号口相同，支持PPP链路层协议，支持IP和IPX等网络协议，可以配置DDR等参数。
no pri-group取消相应的时隙捆绑。
pri-group的索引号固定为15。
【举例】
将CE1/PRI接口的1，2，8~12等时隙捆绑为pri-group。
Quidway(config-if-E0)#pri-group timeslots 1,2,8-12
【相关命令】
channel-group
破解传统语音网
随着IP网络成为企业经营不可缺少的部分，越来越多的企业将电话网络迁移到IP网络之上。通常我们把通过IP网络传输的电话称为IP电话。本文从企业IT管理者的视角，全面介绍从传统电话网络开始、企业内部电话由PBX向IP电话发展的过程，同时也展示了未来企业电话发展的前景。
传统电话网路通常称作公共电话交换网（PSTN: Public Switch Telephone Network）。传统电话网络从最原始的依靠接线员的人工交换网络到今天的数字程控电话网络已经发展了一百多年了。传统电话网络的基本原理就是采用某种连接技术在两部电话之间建立一条可传输话音的电路。随着数字通信技术的发展，电话网发展到今天内部结构已经数字化了。现代数字电话交换网络就是基于TDM（时分复用）技术通过呼叫控制信令在两个电话机之间建立一条独享的电路。
为达到扩展性和可靠性，现代电话网络有严格的层次化结构：通常称为用户端局和汇接局。用户端局通常配置用户交换机，主要连接家庭用户和企业用户，并上连到汇接局。用户接入的方式通常采用模拟双绞线；部分大客户也有采用数字链路接入，通常是n×E1链路。汇接局是汇聚用户局的连接，并与其他汇接局连接，连接链路通常是n×E1或STM1。
从传统上讲，一个企业的电话网络由一个个孤立的电话系统。如下图所示。企业总部同下面分公司的电话通过电话服务商来提供。企业要付昂贵的电话费，并且企业能否有一些特殊的电话需求很大程度受服务商的限制。
为突破这种限制，企业完全可以组建自己的电话网络系统。首先我们要了解现有的电话系统。同计算机网络一样，电话网络也有自己的网络通信描述名词。为了更好地理解电话网络，首先介绍传统电话网络的一些信令。
基于层次化结构，电话的呼叫控制信令可分为局间信令和用户信令。
局间信令
局间传输通常在数字中继链路上进行，有时也把局间信令称数字中继信令。目前最通用的信令为R2和SS7；国内通常称为１号信令和７号信令。
R2信令
R2信令为一种随路信令（CAS: Channel Associated Signaling）。R2信令是一种基于E1数字网络的国际标准信令，Timeslot 16被预留用来传递其话音通道的信令。但是R2信令并不统一，ITU-T的标准Q.400-Q.490定义了R2信令标准，但不同的国家和地区都有自己的实现方式。
SS7信令
SS7信令为一种共路信令（CCS: Common Channel Signaling）。SS7信令是将呼叫控制信息和其他业务信息通过一张独立的信令网络传输。它比R2信令更高效，更可靠。SS7信令的标准化程度要比R2信令好，但依然存在标准兼容问题。如国内的SS7 称为中国7号信令。
用户端信令
电话用户通常采用两芯双绞线连接到用户交换机。所以用户端信令为面向用户的模拟接口信令，通常也称为模拟接口信令。电话机不同于计算系统，它几乎没有什么智能，只能通过简单的模拟信号来沟通。下面介绍模拟话音的信令。
模拟话音的信令有三种，FXS、FXO和E&M。
FXS（Foreign Exchange Station）
可以理解为计算机通信中的DCE（数据通信设备）接口。通常为电话交换机的用户端口，用来连接具有FXO端口的端设备，如电话机或集团电话。
FXO（Foreign Exchange Office）
可以理解为计算机通信中的DTE（数据终端设备）接口。通常为端设备，连接具有FXS端口的电话交换机设备，如PBX或市话局。
FXS和FXO通过两芯电缆构成环路的断开和闭合与电流信号来完成电话的呼叫控制。
E&M（RecEive and TransMit）
E&M是一种模拟中继信令，主要用于PBX到PBX和PBX到市话局的互连。与FXS/FXO不同，E&M端口之间直接互连，将两台PBX连在一起，通常我们也称为捆绑中继接口（Tie Trunk）。
E&M信令采用４对电缆（８芯）通信。E&M信令有５种类型：TYPE1、TYPE2、 TYPE3、TYPE4和TYPE5，每一种类型有自己的通信方式。
当了解现有的电话网络的连接信令后，企业可以自主的采用多种技术组建自己的电话网，只需要在与市话提供商互连时准备相应的接口。"
R2信令就是中国一号信令吗？ - Cisco网络技术论坛
R2信令就是中国一号信令吗？ - Cisco网络技术论坛: "回复: R2信令就是中国一号信令吗？
对，r2就是一号信令。
controller E1 1/0==>设置E1接口
framing NO-CRC4==>成帧不效验
service-type cas-voice==>语音中继
ds0-group 0 timeslots 1-15,17-31 type r2-digital r2-compelled ani==>定义16信道为控制信道，这是r2信令的标准
cas-custom 0==>自定义控制方式
unused-abcd 1 0 1 1==>信道控制信令值，r2的标准
country china==>中国标准
answer-signal group-b 1==>中继应答标准
voice-port 1/0:0
cptone CN==>音序为中国标准
!
!
!
dial-peer voice 30 pots
destination-pattern 30...==>目的电话为30开头的
direct-inward-dial==>定义进入方式为直拨,即该电话号码为本地的
port 1/0:0==>定义会话脚
prefix 30==>前缀为30的电话
!
dial-peer voice 20 voip
destination-pattern 20...==>目的电话为30开头的
session target ipv4:10.33.0.164==>定义目的ip地址，即voip的远端
__________________"
对，r2就是一号信令。
controller E1 1/0==>设置E1接口
framing NO-CRC4==>成帧不效验
service-type cas-voice==>语音中继
ds0-group 0 timeslots 1-15,17-31 type r2-digital r2-compelled ani==>定义16信道为控制信道，这是r2信令的标准
cas-custom 0==>自定义控制方式
unused-abcd 1 0 1 1==>信道控制信令值，r2的标准
country china==>中国标准
answer-signal group-b 1==>中继应答标准
voice-port 1/0:0
cptone CN==>音序为中国标准
!
!
!
dial-peer voice 30 pots
destination-pattern 30...==>目的电话为30开头的
direct-inward-dial==>定义进入方式为直拨,即该电话号码为本地的
port 1/0:0==>定义会话脚
prefix 30==>前缀为30的电话
!
dial-peer voice 20 voip
destination-pattern 20...==>目的电话为30开头的
session target ipv4:10.33.0.164==>定义目的ip地址，即voip的远端
__________________"
一个典型的R2(中国一号)信令上车网关的配置（基于AS5300）－2008年计算机等级考试成绩查询,计算机类考试,计算机等级考试,水平考试,微软认证,思科认证,Oracle认证,Linux认证
一个典型的R2(中国一号)信令上车网关的配置（基于AS5300）－2008年计算机等级考试成绩查询,计算机类考试,计算机等级考试,水平考试,微软认证,思科认证,Oracle认证,Linux认证: "sunspirit0(sunspirit)
W#sh run
Building configuration...
Current configuration:
!
!
version 12.0
no service pad
service timestamps debug datetime msec
service timestamps log datetime msec
service password-encryption
!
hostname GW
!
logging buffered 512000 debugging
aaa new-model
aaa accounting connection h323 start-stop group radius
enable secret 5 $1$MxX.$c79QYkvpIvxGTkTnqy1en0
!
username xxxx password 7 107A0C1A0D
!
!
resource-pool disable
!
!
!
!
!
clock calendar-valid
ip subnet-zero
no ip domain-lookup
ip domain-name x.net.cn
ip name-server x.x.x.x
!
isdn voice-call-failure 0
mta receive maximum-recipients 1
!
!
controller E1 0
framing NO-CRC4
clock source line primary
ds0-group 0 timeslots 1-15,17-31 type r2-digital r2-compelled ani
cas-custom 0
unused-abcd 0 1 1 1
country china
answer-signal group-b 1
dnis-digits min 5 max 5
!
controller E1 1
framing NO-CRC4
clock source line secondary 1
ds0-group 0 timeslots 1-15,17-31 type r2-digital r2-compelled ani
cas-custom 0
unused-abcd 0 1 1 1
country china
answer-signal group-b 1
dnis-digits min 5 max 5
!
controller E1 2
framing NO-CRC4
clock source line secondary 2
ds0-group 0 timeslots 1-15,17-31 type r2-digital r2-compelled ani
cas-custom 0
unused-abcd 0 1 1 1
country china
answer-signal group-b 1
dnis-digits min 5 max 5
!
controller E1 3
framing NO-CRC4
clock source line secondary 3
ds0-group 0 timeslots 1-15,17-31 type r2-digital r2-compelled ani
cas-custom 0
unused-abcd 0 1 1 1
country china
answer-signal group-b 1
dnis-digits min 5 max 5
!
!
voice-port 0:0
cptone GB
timeouts call-disconnect 0
!
voice-port 1:0
cptone GB
timeouts call-disconnect 0
!
voice-port 2:0
cptone GB
timeouts call-disconnect 0
!
voice-port 3:0
cptone GB
timeouts call-disconnect 0
gw-accounting h323 vsa
!
voice class codec 88
codec preference 1 g729br8 bytes 50
codec preference 3 g729r8 bytes 50
codec preference 5 g723ar53
codec preference 6 g723ar63 bytes 144
codec preference 7 g723r53
codec preference 8 g723r63 bytes 120
!
!
dial-peer voice 80001 voip
destination-pattern 01T
voice-class codec 88
dtmf-relay cisco-rtp h245-signal h245-alphanumeric
fax-rate 9600
session target ras
!
dial-peer voice 80002 voip
destination-pattern 02T
voice-class codec 88
dtmf-relay cisco-rtp h245-signal h245-alphanumeric
fax-rate 9600
session target ras
!
dial-peer voice 80003 voip
destination-pattern 03T
voice-class codec 88
dtmf-relay cisco-rtp h245-signal h245-alphanumeric
fax-rate 9600
session target ras
!
dial-peer voice 80003 voip
destination-pattern 04T
voice-class codec 88
dtmf-relay cisco-rtp h245-signal h245-alphanumeric
fax-rate 9600
session target ras
dial-peer voice 80003 voip
destination-pattern 05T
voice-class codec 88
dtmf-relay cisco-rtp h245-signal h245-alphanumeric
fax-rate 9600
session target ras
dial-peer voice 80003 voip
destination-pattern 06T
voice-class codec 88
dtmf-relay cisco-rtp h245-signal h245-alphanumeric
fax-rate 9600
session target ras
dial-peer voice 80003 voip
destination-pattern 07T
voice-class codec 88
dtmf-relay cisco-rtp h245-signal h245-alphanumeric
fax-rate 9600
session target ras
!
dial-peer voice 80003 voip
destination-pattern 08T
voice-class codec 88
dtmf-relay cisco-rtp h245-signal h245-alphanumeric
fax-rate 9600
session target ras
!
dial-peer voice 80003 voip
destination-pattern 09T
voice-class codec 88
dtmf-relay cisco-rtp h245-signal h245-alphanumeric
fax-rate 9600
session target ras
!
dial-peer voice 80003 voip
destination-pattern 00T
voice-class codec 88
dtmf-relay cisco-rtp h245-signal h245-alphanumeric
fax-rate 9600
session target ras
dial-peer voice 500000 pots
incoming called-number 0T
answer-address xxxxxxxxxxxx
direct-inward-dial
prefix 0
!
interface Loopback0
ip address x.x.x.x 255.255.255.255
no ip directed-broadcast
h323-gateway voip interface
h323-gateway voip id haiyun ipaddr x.x.x.x 1719
h323-gateway voip h323-id xxxx
h323-gateway voip tech-prefix 1#
!
interface Ethernet0
ip address 210.53.232.78 255.255.255.192
no ip directed-broadcast
no ip route-cache
no ip mroute-cache
!
interface Serial0
no ip address
no ip directed-broadcast
no ip route-cache
no ip mroute-cache
shutdown
no fair-queue
clockrate 2015232
!
interface Serial1
no ip address
no ip directed-broadcast
no ip route-cache
no ip mroute-cache
shutdown
no fair-queue
clockrate 2015232
!
interface Serial2
no ip address
no ip directed-broadcast
no ip route-cache
no ip mroute-cache
shutdown
no fair-queue
clockrate 2015232
!
interface Serial3
no ip address
no ip directed-broadcast
no ip route-cache
no ip mroute-cache
shutdown
no fair-queue
clockrate 2015232
!
interface FastEthernet0
no ip address
no ip directed-broadcast
no ip route-cache
no ip mroute-cache
shutdown
duplex auto
speed auto
!
ip classless
no ip http server
!
!
!
line con 0
transport input none
line aux 0
line vty 0 4
!
end"
W#sh run
Building configuration...
Current configuration:
!
!
version 12.0
no service pad
service timestamps debug datetime msec
service timestamps log datetime msec
service password-encryption
!
hostname GW
!
logging buffered 512000 debugging
aaa new-model
aaa accounting connection h323 start-stop group radius
enable secret 5 $1$MxX.$c79QYkvpIvxGTkTnqy1en0
!
username xxxx password 7 107A0C1A0D
!
!
resource-pool disable
!
!
!
!
!
clock calendar-valid
ip subnet-zero
no ip domain-lookup
ip domain-name x.net.cn
ip name-server x.x.x.x
!
isdn voice-call-failure 0
mta receive maximum-recipients 1
!
!
controller E1 0
framing NO-CRC4
clock source line primary
ds0-group 0 timeslots 1-15,17-31 type r2-digital r2-compelled ani
cas-custom 0
unused-abcd 0 1 1 1
country china
answer-signal group-b 1
dnis-digits min 5 max 5
!
controller E1 1
framing NO-CRC4
clock source line secondary 1
ds0-group 0 timeslots 1-15,17-31 type r2-digital r2-compelled ani
cas-custom 0
unused-abcd 0 1 1 1
country china
answer-signal group-b 1
dnis-digits min 5 max 5
!
controller E1 2
framing NO-CRC4
clock source line secondary 2
ds0-group 0 timeslots 1-15,17-31 type r2-digital r2-compelled ani
cas-custom 0
unused-abcd 0 1 1 1
country china
answer-signal group-b 1
dnis-digits min 5 max 5
!
controller E1 3
framing NO-CRC4
clock source line secondary 3
ds0-group 0 timeslots 1-15,17-31 type r2-digital r2-compelled ani
cas-custom 0
unused-abcd 0 1 1 1
country china
answer-signal group-b 1
dnis-digits min 5 max 5
!
!
voice-port 0:0
cptone GB
timeouts call-disconnect 0
!
voice-port 1:0
cptone GB
timeouts call-disconnect 0
!
voice-port 2:0
cptone GB
timeouts call-disconnect 0
!
voice-port 3:0
cptone GB
timeouts call-disconnect 0
gw-accounting h323 vsa
!
voice class codec 88
codec preference 1 g729br8 bytes 50
codec preference 3 g729r8 bytes 50
codec preference 5 g723ar53
codec preference 6 g723ar63 bytes 144
codec preference 7 g723r53
codec preference 8 g723r63 bytes 120
!
!
dial-peer voice 80001 voip
destination-pattern 01T
voice-class codec 88
dtmf-relay cisco-rtp h245-signal h245-alphanumeric
fax-rate 9600
session target ras
!
dial-peer voice 80002 voip
destination-pattern 02T
voice-class codec 88
dtmf-relay cisco-rtp h245-signal h245-alphanumeric
fax-rate 9600
session target ras
!
dial-peer voice 80003 voip
destination-pattern 03T
voice-class codec 88
dtmf-relay cisco-rtp h245-signal h245-alphanumeric
fax-rate 9600
session target ras
!
dial-peer voice 80003 voip
destination-pattern 04T
voice-class codec 88
dtmf-relay cisco-rtp h245-signal h245-alphanumeric
fax-rate 9600
session target ras
dial-peer voice 80003 voip
destination-pattern 05T
voice-class codec 88
dtmf-relay cisco-rtp h245-signal h245-alphanumeric
fax-rate 9600
session target ras
dial-peer voice 80003 voip
destination-pattern 06T
voice-class codec 88
dtmf-relay cisco-rtp h245-signal h245-alphanumeric
fax-rate 9600
session target ras
dial-peer voice 80003 voip
destination-pattern 07T
voice-class codec 88
dtmf-relay cisco-rtp h245-signal h245-alphanumeric
fax-rate 9600
session target ras
!
dial-peer voice 80003 voip
destination-pattern 08T
voice-class codec 88
dtmf-relay cisco-rtp h245-signal h245-alphanumeric
fax-rate 9600
session target ras
!
dial-peer voice 80003 voip
destination-pattern 09T
voice-class codec 88
dtmf-relay cisco-rtp h245-signal h245-alphanumeric
fax-rate 9600
session target ras
!
dial-peer voice 80003 voip
destination-pattern 00T
voice-class codec 88
dtmf-relay cisco-rtp h245-signal h245-alphanumeric
fax-rate 9600
session target ras
dial-peer voice 500000 pots
incoming called-number 0T
answer-address xxxxxxxxxxxx
direct-inward-dial
prefix 0
!
interface Loopback0
ip address x.x.x.x 255.255.255.255
no ip directed-broadcast
h323-gateway voip interface
h323-gateway voip id haiyun ipaddr x.x.x.x 1719
h323-gateway voip h323-id xxxx
h323-gateway voip tech-prefix 1#
!
interface Ethernet0
ip address 210.53.232.78 255.255.255.192
no ip directed-broadcast
no ip route-cache
no ip mroute-cache
!
interface Serial0
no ip address
no ip directed-broadcast
no ip route-cache
no ip mroute-cache
shutdown
no fair-queue
clockrate 2015232
!
interface Serial1
no ip address
no ip directed-broadcast
no ip route-cache
no ip mroute-cache
shutdown
no fair-queue
clockrate 2015232
!
interface Serial2
no ip address
no ip directed-broadcast
no ip route-cache
no ip mroute-cache
shutdown
no fair-queue
clockrate 2015232
!
interface Serial3
no ip address
no ip directed-broadcast
no ip route-cache
no ip mroute-cache
shutdown
no fair-queue
clockrate 2015232
!
interface FastEthernet0
no ip address
no ip directed-broadcast
no ip route-cache
no ip mroute-cache
shutdown
duplex auto
speed auto
!
ip classless
no ip http server
!
!
!
line con 0
transport input none
line aux 0
line vty 0 4
!
end"
VG语音网关R2信令典型配置 - 中国主流IT|思科华为3COM微软Juniper|认证网络技术专业站-无兄弟不技术(56Cto.Com)
VG语音网关R2信令典型配置 - 中国主流IT|思科华为3COM微软Juniper|认证网络技术专业站-无兄弟不技术(56Cto.Com): "VG语音网关R2信令典型配置
来源：作者： 发布时间：2008-06-04 阅读次数8
一、组网需求:
北京、深圳两地的电话利用语音网关直接经由IP网络进行通话，北京语音网关通过E1语音用户线连接PBX交换机并采用R2信令；深圳语音网关通过E1语音用户线连接PBX交换机，采用R2信令。拨号采用一次拨号方式。
二、组网图:
三、配置步骤：
1．北京侧语音网关的参数配置
# 配置TS组。
[VG] controller e1 0
//进入E1接口配置界面
[VG-E1-0] timeslot-set 0 timeslot-list 1-15,17-31 signal r2
//配置信令方式（R2）
# 建立E1端口上的POTS语音实体（电话号码010-1001）。
[VG-voice-dial-entity1003] entity 1001 pots
[VG-voice-dial-entity1001] match-template 0101001
# 配置E1端口上的POTS语音实体与逻辑端口对应。
[VG-voice-dial-entity1001] line 0:1
[VG-voice-dial-entity1001] send-number all
//缺省情况下是“截号发送”
# 建立E1端口上的POTS语音实体（电话号码010-1002）。
[VG-voice-dial-entity1001] entity 1002 pots
[VG-voice-dial-entity1002] match-template 0101002
# 配置E1端口上的POTS语音实体与逻辑端口对应。
[VG-voice-dial-entity1002] line 0:1
[VG-voice-dial-entity1001] send-number all
//缺省情况下是“截号发送”
# 建立VoIP语音实体。
[VG-voice-dial-entity1002] entity 0755 voip
[VG-voice-dial-entity755] match-template 0755....
[VG-voice-dial-entity755] address ip 2.2.2.2
2．深圳侧语音网关的参数配置与北京侧大体一致
# 配置TS组。
[VG] controller e1 0
//进入E1接口配置界面
[VG-E1-0] timeslot-set 0 timeslot-list 1-15,17-31 signal r2
//配置E1链路信令方式（R2）
[VG-E1-0] cas 0
//进入时隙组配置界面
[VG-E1-0] trunk-direction timeslots 17-31 in
//配置时隙方向，缺省情况下时隙都是“dual”
[VG-E1-0] trunk-direction timeslots 1-15 out
//配置时隙方向，缺省情况下时隙都是“dual”
[VG-E1-0] mode china default
//配置信令类别（中国一号）
[VG-E1-0] ani
//要求对端设备发送主叫号码（缺省不要求）
[VG-E1-0] quit
# 建立E1端口上的POTS语音实体（电话号码0755-2001）。
[VG] voice-setup
[VG-voice] dial-program
[VG-voice-dial] entity 2001 pots
[VG-voice-dial-entity2001] match-template 07552001
# 配置E1端口上的POTS语音实体与逻辑端口对应。
[VG-voice-dial-entity2001] line 1:1
[VG-voice-dial-entity2001] send-number all
//缺省情况下是“截号发送”
# 建立E1端口上的POTS语音实体（电话号码0755-2002）。
[VG-voice-dial-entity2001] entity 2002 pots
[VG-voice-dial-entity2002] match-template 07552002
# 配置E1端口上的POTS语音实体与逻辑端口对应。
[VG-voice-dial-entity2002] line 1:1
[VG-voice-dial-entity2002] send-number all
//缺省情况下是“截号发送”
# 建立VoIP语音实体。
[VG-voice-dial-entity2002] entity 010 voip
[VG-voice-dial-entity10] match-template 010....
[VG-voice-dial-entity10] address ip 1.1.1.1
四、配置关键点：
缺省情况下，E1采用R2信令的时候，30个时隙的方向都是dual（双向），即时隙可以用于双向传输信令（但是某一时刻只能传输一路信令）。语音网关E1链路的R2配置必须和PBX侧参数一致，包括时隙方向（in、out、dual）以及信令类别（如中国一号）。如果需要VoIP网络现实来自PSTN的主叫号码，那么必须配置ani，请求PBX侧发送主叫（此前提是PBX侧能够发送主叫，否则会导致呼叫不成功）"
来源：作者： 发布时间：2008-06-04 阅读次数8
一、组网需求:
北京、深圳两地的电话利用语音网关直接经由IP网络进行通话，北京语音网关通过E1语音用户线连接PBX交换机并采用R2信令；深圳语音网关通过E1语音用户线连接PBX交换机，采用R2信令。拨号采用一次拨号方式。
二、组网图:
三、配置步骤：
1．北京侧语音网关的参数配置
# 配置TS组。
[VG] controller e1 0
//进入E1接口配置界面
[VG-E1-0] timeslot-set 0 timeslot-list 1-15,17-31 signal r2
//配置信令方式（R2）
# 建立E1端口上的POTS语音实体（电话号码010-1001）。
[VG-voice-dial-entity1003] entity 1001 pots
[VG-voice-dial-entity1001] match-template 0101001
# 配置E1端口上的POTS语音实体与逻辑端口对应。
[VG-voice-dial-entity1001] line 0:1
[VG-voice-dial-entity1001] send-number all
//缺省情况下是“截号发送”
# 建立E1端口上的POTS语音实体（电话号码010-1002）。
[VG-voice-dial-entity1001] entity 1002 pots
[VG-voice-dial-entity1002] match-template 0101002
# 配置E1端口上的POTS语音实体与逻辑端口对应。
[VG-voice-dial-entity1002] line 0:1
[VG-voice-dial-entity1001] send-number all
//缺省情况下是“截号发送”
# 建立VoIP语音实体。
[VG-voice-dial-entity1002] entity 0755 voip
[VG-voice-dial-entity755] match-template 0755....
[VG-voice-dial-entity755] address ip 2.2.2.2
2．深圳侧语音网关的参数配置与北京侧大体一致
# 配置TS组。
[VG] controller e1 0
//进入E1接口配置界面
[VG-E1-0] timeslot-set 0 timeslot-list 1-15,17-31 signal r2
//配置E1链路信令方式（R2）
[VG-E1-0] cas 0
//进入时隙组配置界面
[VG-E1-0] trunk-direction timeslots 17-31 in
//配置时隙方向，缺省情况下时隙都是“dual”
[VG-E1-0] trunk-direction timeslots 1-15 out
//配置时隙方向，缺省情况下时隙都是“dual”
[VG-E1-0] mode china default
//配置信令类别（中国一号）
[VG-E1-0] ani
//要求对端设备发送主叫号码（缺省不要求）
[VG-E1-0] quit
# 建立E1端口上的POTS语音实体（电话号码0755-2001）。
[VG] voice-setup
[VG-voice] dial-program
[VG-voice-dial] entity 2001 pots
[VG-voice-dial-entity2001] match-template 07552001
# 配置E1端口上的POTS语音实体与逻辑端口对应。
[VG-voice-dial-entity2001] line 1:1
[VG-voice-dial-entity2001] send-number all
//缺省情况下是“截号发送”
# 建立E1端口上的POTS语音实体（电话号码0755-2002）。
[VG-voice-dial-entity2001] entity 2002 pots
[VG-voice-dial-entity2002] match-template 07552002
# 配置E1端口上的POTS语音实体与逻辑端口对应。
[VG-voice-dial-entity2002] line 1:1
[VG-voice-dial-entity2002] send-number all
//缺省情况下是“截号发送”
# 建立VoIP语音实体。
[VG-voice-dial-entity2002] entity 010 voip
[VG-voice-dial-entity10] match-template 010....
[VG-voice-dial-entity10] address ip 1.1.1.1
四、配置关键点：
缺省情况下，E1采用R2信令的时候，30个时隙的方向都是dual（双向），即时隙可以用于双向传输信令（但是某一时刻只能传输一路信令）。语音网关E1链路的R2配置必须和PBX侧参数一致，包括时隙方向（in、out、dual）以及信令类别（如中国一号）。如果需要VoIP网络现实来自PSTN的主叫号码，那么必须配置ani，请求PBX侧发送主叫（此前提是PBX侧能够发送主叫，否则会导致呼叫不成功）"
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