asterisk@gatheringpoint.net
2005-Jul-06 23:02 UTC
[Asterisk-Users] Dropped calls if transferred across servers into MeetMe with mobile source
I have an application where calls come into an *box from a DID provider, and may be transferred to a meetme conference on another *box (the call is released by the first *box after transfer). These are ulaw IAX channel calls, and if the source is from a Verizon or Nextel mobile phone to the DID (other carriers not tested), the call drops about 2-3 minutes after it joined the meetme conference. POTS originated calls do fine - they do not drop. I've reproduced this consistently, and across two different DID termination providers and several different mobile phones. I'm seeing the behavior on 1.0.7 and 1.0.9. Calls don't fully drop. Meetme shows a reduction in the participant count, and the conference exit tone plays, but the mobile phone thinks it is still connected... AND other call participants can still hear the 'dropped' person, but that person can't hear anything. Also, if I change the first *box iax.conf to notransfer=yes, all calls are reliable (but, of course, I'm tying up resources...not a good long term solution). Console output is as follows for problem calls: Jul 5 15:18:44 WARNING[9256]: chan_iax2.c:1477 attempt_transmit: Max retries exceeded to host xx.xx.xx.xx on IAX2/yyy@ xx.xx.xx.xx:4569/3 (type = 2, subclass = 4, ts=65540, seqno=1) Jul 5 15:18:44 WARNING[9256]: app_meetme.c:962 conf_run: Unable to write frame to channel: No child processes == Spawn extension (toll, 1001074, 5) exited non-zero on 'IAX2/ yyy@ xx.xx.xx.xx:4569/3' -- Hungup 'IAX2/yyy@ xx.xx.xx.xx:4569/3' I've experimented with jitterbuffer on and off, different qos settings (including high reliability), and different meetme options. I haven't been able to impact this behavior. There is an agi that executes when the call arrives at the meetme *box (before meetme is joined). It just hits a db, sets some variable values, and exits cleanly - and again, it's not until 2-3 minutes later that I see the problem, and I don't have any problem with POTs sourced calls. The big variable seems to be whether the call originated from a cell phone or not, and that it was transferred to a second server. This is really strange, and I've even pulled in someone else that does Asterisk work just to do a sanity check and make sure I wasn't missing something obvious... no such luck. Any thoughts or insights?