At 02:58 PM 7/23/2010, you wrote: >The Asterisk Development Team has announced the release of Asterisk >1.8.0-beta1. >This release marks the beginning of the testing process for the >eventual release >of Asterisk 1.8.0.
One more problem. Everything seems to work fine but this morning I decided to test something. Picked up my SIP phone and tried to call myself and it doesn't work. Phone is an Aastra 480i. I can dial out via SIP or POTS via a TDM400. All possible options go straight to voicemail. If I call in from my cell or from 2 cells at once it usually works fine. When it doesn't work, I get 3 pairs of these, I assume one for each of the SIP phones in the house. WARNING[14583]: chan_sip.c:3339 retrans_pkt: Retransmission timeout reached on transmission [email protected]:5060 for seqno 102 (Critical Request) -- See doc/sip-retransmit.txt. Packet timed out after 6400ms with no response WARNING[14583]: chan_sip.c:3368 retrans_pkt: Hanging up call [email protected]:5060 - no reply to our critical packet (see doc/sip-retransmit.txt). It the same dial line in extensions.conf whether it works or not. I did in fact read doc/sip-retransmit.txt, but it didn't seem to contain anything I understood. I assume this should also be in the bug tracker? Ira -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
