CME with SIP Trunk and Cisco Unity Express

Unanswered Question
May 2nd, 2007
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I have a Cisco 3745 running 12.4(11)XJ2 / CME 4.1 / CUE with our PSTN access through a SIP Trunk. Everything is working ok except access to Cisco Unity Express, when it gets transferred to VM the caller hears fast busy tone.

Any Ideas? Thanks in advance.


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Overall Rating: 5 (1 ratings)
paolo bevilacqua Thu, 05/03/2007 - 04:02
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Are the calls coming in with G.711? Nothing else would work with CME.

Does it fails only when transferred, or also when calling main VM number directly ? If the first, you may need to configure:

voice service voip

allow-connections sip to sip

no supplementary-service sip moved-temporarily

To confirm collect output of "debug ccsip message" with "term mon" to know a little more.

Hope this helps, please rate post if it does!

spazziani Fri, 11/18/2011 - 05:24
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Hi Damm,

Did you solve your problem, because I'm having the same problem. Could you help me please ?



Arun Thomas Tue, 11/22/2011 - 04:13
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Please check the dial-peer configs. Sample configs below.

dial-peer voice 200 voip

description CUE-Voicemail

destination-pattern 8000


session protocol sipv2

session target ipv4:

dtmf-relay sip-notify

codec g711ulaw

no vad

spazziani Wed, 11/23/2011 - 14:19
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Arun Thomas,

I verified the dial-peer and it doesn't have two commands "dtfm-relay sip-notify" and "no vad". After I put both commands solved my problem.

Thank you very much.


Arun Thomas Thu, 11/24/2011 - 00:18
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Happy to see that your issue is resolved. but i dont think that its the dtmf command but may be the "no vad"

Please rate post if it does!


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