Music on Hold never stops if the call comes over Asterisk
I use CUCM 7.1(3) ( 184.108.40.20600-4 ), 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
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).
Re: Music on Hold never stops if the call comes over Asterisk
Thanks for your reply.
I updated the cucm to 7.1(3b)SU1 and Asterisk to 220.127.116.11 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 :
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.
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