Call Manager SIP Trunk - Hold Resume Fails
Submitted by mpberry on Fri, 08/01/2008 - 9:11am.
I have a Cisco Call Manager 4.1 server setup with a SIP trunk to Trixbox 2.6.1. I am experiencing problems with hold and resume from the Cisco end of things. Everything on Trixbox seems to work fine. What happens is if you put a call made into the trixbox over the SIP trunk on hold when you try to resume the call the audio comes through but the call is dropped after about 20 seconds. We have Cisco phones on our CM server and what this is what the user experiences:
1. Place call on hold
2. Call indicator starts flashing, audio cuts out, and pause sign appears on the display
3. User presses the resume button
4. Audio starts playing again but the display still keeps flashing with the pause sign like the call is still on hold. The call counter on the phone keeps counting.
5. About 20 seconds later the call is dropped.
Looking at the asterisk CLI on TrixBox I don't see anything happening from the time the initial call is connected until the call is dropped. I'm wondering if anyone else has experienced this issue with Call Manager. I'm thinking that maybe the SIP trunking in this version of CM may have some issues with either translating the SCCP commands to SIP or may be a poor implementation and it can't rejoin the call together after being on hold. When putting a call on hold using an xlite softphone registered to TrixBox it works flawlessly.
What I would like is to find a way that I can see what Call Manager is doing during the call. I love the CLI interface for Asterisk and I'm wondering if there is some equivalent on CM or if its just a mystery black box.
Re: Call Manager SIP Trunk - Hold Resume Fails
Fri, 08/01/2008 - 9:49am
These are my comments about this configuration with Call Manager 6.1.
Our previous 4.12 implementation worked fine with hold and resume.
The issue is that the Cisco Phone is held on the call manager, and the remote phone is held on the tribox.
This would not be a problem BUT the re INVITE from the call manager to the trixbox does not have an SDP in the INVITE so the trixbox does not understand where to go to get back into the stream.
BUT the phone IS sending the INVITE with the correct SDP to the call manager.....
This is as far as I have got, I have not yet figured out how to fix this.
Call Manager 6.1.1 mostly configured in SIP endpoints.
When I put the call on hold
Mon, 08/04/2008 - 12:47pm
When I put the call on hold from my CM attached phone I don't see anything at all come up in the asterisk cli debug window. This makes me think that Asterisk doesn't even know that the call was put on hold. I wish there was some way that I could see inside Call Manager to be able to see what it is trying to do during the hold. The other interesting thing is that the normal on hold music that is on our CM system does not play when putting the call from Trix on hold.
Member Since:
2006-10-14