asterisk 1.4.21-rc1?
Yes, i'm trying to test for
Wed, 06/04/2008 - 1:05pm
Yes, i'm trying to test for two SIP bugs, which were fixed between 1.4.20 and 1.4.21-rc1.
We are having problems with SIP communication, dropped calls with errors like: chan_sip.c: Maximum retries exceeded on transmission 2cc62c79934cc54c@192.168.1.102 for seqno 35083 (Critical Response). We believed at first that the port times out, but further debugging proved that wrong, the call dies while data is being transmitted both ways and while the UDP port is open (no timeout).
Second problem is with SIP message 481, 487, 484 and 415. The call is made and the remote phone rings but they can't hear each other. For some unknown reason the call fails to transmit even though UDP communication works fine. SIP message 415 "unsupported media type" may show whats wrong but we have no idea why it occurs since all phones and the trixbox server support all the codecs (alaw/ulaw/etc) and are all enabled in both sides.
Digium seems to have fixed part of the second problem with proxy and authentication, while the first is not really proven to be fixed but they suggest that some of their changes are related.
A beta version would give us the opportunity to test and report our findings to Digiums bug tracker, if its not too late for the next release.
Thank you.
Member Since:
2007-12-19