I have tried installing many of the beta versions and most of the
release versions of 1.8. I have 3 SIP phones which we use for all our
calls. After installing 1.8 the first thing I try is calling out port
one of my Digium TDM04 back into port 2. I can see that the call
comes in and tries to call all three SIP phones but the phones never
ring. Eventually the call goes to voice mail and these error messages
pop up. I've read doc/sip-retransmit.txt and as far as I can tell,
there's nothing there for me to try.
Is there anything else I might try or do to help troubleshoot this.
Ira
WARNING[24384] chan_sip.c: Retransmission timeout reached on
transmission [email protected]:5060 for
seqno 102 (Critical Request) Packet timed out after 20672ms with no response
WARNING[24384] chan_sip.c: Hanging up call
[email protected]:5060 - no reply to our
critical packet (see doc/sip-retransmit.txt).
WARNING[24384] chan_sip.c: Retransmission timeout reached on
transmission [email protected]:5060 for
seqno 102 (Critical Request) Packet timed out after 20736ms with no response
WARNING[24384] chan_sip.c: Hanging up call
[email protected]:5060 - no reply to our
critical packet (see doc/sip-retransmit.txt).
WARNING[24384] chan_sip.c: Retransmission timeout reached on
transmission [email protected]:5060 for
seqno 102 (Critical Request) Packet timed out after 20801ms with no response
WARNING[24384] chan_sip.c: Hanging up call
[email protected]:5060 - no reply to our
critical packet (see doc/sip-retransmit.txt).
--
_____________________________________________________________________
-- 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