Closed GoogleCodeExporter closed 9 years ago
Issue 349 has been merged into this issue.
Original comment by r3gis...@gmail.com
on 7 Nov 2010 at 12:10
It seems this is due to a double registration made by CSipSimple on the SIP /
VoIP backend
Some logs to be provided
Original comment by eha...@gmail.com
on 14 Nov 2010 at 6:59
If so probably cause the registrar doesn't support one shot un-registration -
re-registration.
Pjsip use a technique to detect automatically NAT. But it need to do a first
register and then a re-register.
The problem was present with SipSorcery and they fix that on their server.
However I've just found an option in pjsip that may solve the problem using a
'legacy' method. It will be available in next release in expect account mode.
But in this case I'd deeply advise to report the issue to the sip provider
cause it mean they don't fully support RFCs.
Original comment by r3gis...@gmail.com
on 14 Nov 2010 at 8:15
The option to fall back to old legacy re-register method is now available in
expert wizard.
Original comment by r3gis...@gmail.com
on 26 Nov 2010 at 11:47
Trying the workaround, but in the meantime, it's even worse than reported
here--when the second call is marked "missed", the first call (in progress)
loses audio.
Original comment by jad...@gmail.com
on 14 Feb 2011 at 12:52
Original issue reported on code.google.com by
r3gis...@gmail.com
on 17 Aug 2010 at 12:13