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VoIP / Asterisk / rtcomm
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R-R
2007-11-29 , 16:49
Posts: 739 | Thanked: 242 times | Joined on Sep 2007 @ Montreal
#
17
Originally Posted by
dblank
Shouldn't that be quali
f
y?
Indeed, thanks! Also, i saw that dtmf= should be dtmfmode= ...
But even with that fixed i still have the same symptoms.
in:
tcpdump -i eth1
n800-IP
-A
, from the gateway with asterisk, i can see:
11:31:08.732719 IP 192.168.1.1.11981 > 192.168.1.178.7079: UDP, length 44
11:31:08.738876 IP 192.168.1.178 > 192.168.1.1: ICMP 192.168.1.178 udp port 7079 unreachable, length 80
I got lucky and got a call to go through and had this in
asterisk -rvvvvvv
just around where it got disconnected after 12 minutes:
[Nov 29 11:34:43] NOTICE[28729]: chan_sip.c:15460 sip_poke_noanswer: Peer 'n800' is now UNREACHABLE! Last qualify: 6
was still working a bit after the previous warning...
[Nov 29 11:35:50] NOTICE[28729]: chan_sip.c:12336 handle_response_peerpoke: Peer 'n800' is now Reachable. (1428ms / 2000ms)
then it died.... "Connection to one or more account lost" in rtcomm.
== Spawn extension (home, 1234, 1) exited non-zero on 'SIP/n800-081edfb8'
i probably hung up my home ATA/Phone here...
[Nov 29 11:36:46] WARNING[28729]: chan_sip.c:12038 handle_response_invite: Re-invite to non-existing call leg on other UA. SIP dialog '347a5eeb-1910-122b-3595-00194fd4ead7'. Giving up.
And after a while i see this.
Same goes with qualify=no, nat=no ... :|
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