Cisco Router-MD110: Auto Callback not working over VOIP

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Nov 9th, 2007
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I connected cisco 3800 router with ericsson MD110 exchange using QSIG over E1 Pri. Calls are forwarding properly over ip network but few advance features are not being enabled like auto callback. In existing scenario,

User A-----MD110 PBX------------------MD110PBX-----User B

If a user A calls user B and user B is busy with someone else. User A will get busy tone, then user A will press digit 6 at that busy tone and put handset back to phone. When user B ends his call, a call is initiated automatically that rings bell of both phones A and B. Now when I implement voip and place routers in it, this feature no more working. My New scenario is below:

User A-----MD110 PBX ----RouterA---------------RouterB----MD110 PBX-----User B

I think, In voip topology when user A dials number, router A receives all digits and match dialpeer against it. When dialpeer matches, it forwards it to relevant router B and router B forward to pbx. And after that, router will not forward upcoming digits like 6 in this case and that's why this feature is not working. I think, I will become success to forward upcoming digits to other router after dialpeer match, this feature will start working over voip.

Waiting for immediate positive response.



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paolo bevilacqua Sat, 11/10/2007 - 13:17
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Yes it's possible that something like you say happens, as the audio path may be closed after busy singal, I'm not sure if there is a feature configuration to avoid that, and some debug may be needed to understa who is not passing the digit.

Anyway one thing you can do to have all and any feature working between PBX's, is configure a "trunking" with transparent CCS, so both the signalling channel and the audio channels would be permanently tied basically emulating a physical link. That has the disavantage that unless you configure VAD, you would be generating more traffic than 2 mbos bandwidht you are carring, maybe a 25% more traffic.

Hope this helps, please rate post if it does!

vinay_verma80 Tue, 01/01/2008 - 21:48
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What is the signaling standard u are using in voip dialer peer , use h245-signaling

also configure the pbx to give an IVR announcement in case of busy tone if the other phone is busy so that ur no ^ will be fwded

I hope this will solved ur problem



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