Just noticed a problem, and I'm trying to narrow it down. Caller A calls extension B. It can be an extension-to-extension call or A can call in from outside. B does an attended transfer to C. But, B and C don't hear each other (no-way audio). If B hangs up the transfer is completed and A can talk to C. If B calls C directly they can talk.
Everybody is using g.711. No reinvites anywhere.
Any idea what could be causing this?
Seems now to be all phones.
Seems now to be all phones. I've verified it with Polycom 330 and 550, Aastra 57i, and Bria softphone. The phones are behind a NAT, but they are registering with an Edgemarc 4500 router; which does sip proxy. I've got a call in to my Edgemarc vendor.
Is it one tennant or all
Is it one tennant or all phones on the system?
I can give you the number of a really good Edgewater engineer if you want it....just drop me a note off list or give me a call.
-Matt
Blind transfer seems to work
Blind transfer seems to work fine. Attended transfer works, too, except for the fact that person doing the transfer and the person to whom the call is being transfered can't talk to each other. So, attended is in effect a blind transfer.
what version of
what version of asterisk/zaptel are you running? it sounds similar to a problem with the sip jitter buffer. do you have the jb settings turned on in sip.conf?
Yes, I do have jb turned on.
Yes, I do have jb turned on. I will turn it off and give it a try. Also, Matt and Kevin, thanks for the info from Edgewater. I will call Monday.
try turning off jb
try turning off jb everywhere in sip.conf. If the problem still doesnt go away we will move on to the next possibility. I just want to make sure that the jitter-buffer shrink bug isn't haunting you somehwere.
:%s/jbenable=yes/jbenable=no/
:%s/jbenable=yes/jbenable=no/g all the way around, and the problem appears to be solved!
Thanks for all of the good advice, guys!
Hmmm... Well, it's partially
Hmmm... Well, it's partially fixed. It works if I transfer with #* (as defined in features.conf), but I can't transfer with the transfer soft key on my Polycoms. Also, if I put an incoming call on hold and call another extension I can't talk.
well if you want to rule out
well if you want to rule out your sip proxy take 2 phones' mac-registration.cfg file and remove sip proxy settings to see if you get the exact same behavior. If you get identical behavior you can cross the sip proxy off the list. By chance is the sip proxy trying to re-route your call internally without going through asterisk? Sometimes they do that if they know the address your calling is registered with the sip proxy.
No, it is built in to the
No, it is built in to the Edgemarc 4500 router. I'm not sure what it is. I'm going to try capturing some packets this morning to see if I can see what's going on from there, and I will call Edgewater Networks later.
I spoke with support and
I spoke with support and they haven't heard from you, yet.
While they probably won't be able to help you with specific PBX issues, they are very helpful sorting signaling issues. We do have a few customers that use Thirdlane successfully, so it would be good to see what is different.
-Kevin
kevinfvc, would you mind
kevinfvc, would you mind contacting me at dan@env-sol.net? I have a couple of Edgemarc product questions that I would like to ask off of this forum.
is this problem specific to a certain model of phone or does it exist for all models including softphones?
Are you behind nat or is the PBX on the same network?