Music on Hold never stops if the call comes over Asterisk

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Dec 16th, 2009
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Hi There,


I use CUCM 7.1(3) (                                               7.1.3.21900-4                     ), i think it should be the most actual version of the Callmanager.


Because the Callmanager is not so surprised in registering to a sip-registrar i set up an Asterisk-PBX to get the calls in.

Dialing in and out works really fine, i set up an SIP Trunk between Cisco and Asterisk.


I have just one Problem and it seems that it comes from the Callmanager.

If someone calls in the call gets from the Sip-Provider to Asterisk and then through the Callmanager to the Phone (Cisco 7960).

When i press the hold Button the caller gets on hold and hears the moh-music from Asterisk. Thats ok.

But if i press resume on the IP Phones, the IP Phone thinks the caller is back, but the caller always hears moh and no matter what i press or do it never stops.


i looked around in some asterisk forums and found other people with the same problems. The development team looked at the .cap files and wrote that cisco makes something wrong with the sdp header and that RFC 3264 says that it should not be that way.

i can post the links here if you want me to do this.


so is it possible to


1. Configure the Callmanager to make it work


or


2. Configure the Callmanager to put the calls on hold on the Callmanager-moh (i have configured moh and for Callmanager internal calls it works).


Thanks in advance


Rafael / Germany

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gogasca Thu, 12/24/2009 - 15:00
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1. can you post that information.

2. CUCM SDI, SDL traces

3. Cap files

4. try with MTP in SIP Trunk and see if it works or not    


Thanks!

rafael_rung Wed, 12/30/2009 - 11:05
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Thanks for your reply.


I updated the cucm to 7.1(3b)SU1 and Asterisk to 1.6.2.0 but it didn't help.


i upload the trace files now, they contain one call in my test-lap.


the caller is put on hold one time, then i pressed the resume button on the 7960 phone. The phone display then changed to a normal connected call display, but the caller heared moh music from asteris until the call ends.


here are the links from the asterisk forum you requested :

0014385: [patch] Unhold fails if first SDP on OK, particularly Cisco CCM 6: https://issues.asterisk.org/view.php?id=14385

0016373: Asterisk ignoring sendonly SDP generated from Cisco UCM after generating inactive SDP when a Cisco phone initiates hold: https://issues.asterisk.org/view.php?id=16373&nbn=2

0016313: Responds sendrecv to recvonly SDP, but RFC 3264 says sendonly and inactive are only possible replie:

https://issues.asterisk.org/view.php?id=16313&nbn=2


When i activate MTP on the SIP Trunk everything works fine, the callers hears the cisco-moh and hold and unhold works.

but then every RTP packet goes to the cucm and back to the phone. if i get it to work we planned to instell the cucm in the datacenter and to install asterisk and the ip-phones in the office. and there is just a VPN connection between. it's not possible to route every rtp packet to the datacenter.

so i have to deactivate MTP on the sip trunk.

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p-vincent Mon, 02/08/2010 - 00:00
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Hello Rafael,


Did you find a solution on this one, I have the same problem over here.


Thanks in advance


Peter

rafael_rung Mon, 02/08/2010 - 06:55
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Hello Peter,


until now i have no solution. i'm still waiting for the answer from gogasca (Cisco Systems), perhaps he knows something after he looked at the trace files.


if both servers are on the same location you can enable MTP (as explained above).

p-vincent Mon, 02/08/2010 - 07:28
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Hello Rafael,


Thx for your answer.


Our SIP Connection is connected to a external server, but when I activate the MTP on the Trunk everything works fine.

rafael_rung Thu, 02/25/2010 - 00:46
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Well, as long as i don't find another solution i solved it this way:


I configured cucm and asterisk with H.323 and for the beginning it looks good.


hold and unhold works fine and the RTP packets don't move over the vpn to the datacenter.

ausjustin Wed, 02/26/2014 - 07:24
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Hi Rafael


how you resolved this issue yet ?

I had same problem , but i dont want enable MTP on trunk as it will play moh music from asteris until the call ends.

r.rung Thu, 02/27/2014 - 11:09
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Hello ausjustin,


well my solution was to kick out asterisk as my home pbx and put in a Cisco UC540 instead.

so i did no further investigation in this issue, sorry.


Rafael

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