[asterisk-users] Asterisk 1.8.0-beta1 is Now Available!

[Ira]

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 
10842037066464ef58d4f88d16535b4c at 192.168.233.235:5060 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 
10842037066464ef58d4f88d16535b4c at 192.168.233.235:5060 - 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