Displaying 3 results from an estimated 3 matches for "sip_debug".
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sim_debug
2010 Aug 29
1
evil disconnect of call with cisco 1760
...for seqno
102 (Critical Request) -- See doc/sip-retransmit.txt.
WARNING[2492]: chan_sip.c:3805 retrans_pkt: Hanging up call
CB674A02-B25C11DF-B6D5A08D-652FE73E at 10.9.1.9 - no reply to our critical
packet (see doc/sip-retransmit.txt).
I have a full sip debug at: http://jeremy.kister.net/tmp/sip_debug/ast.txt
A running config of the c1760 is at
http://jeremy.kister.net/tmp/sip_debug/c1760.txt
Important parts of sip.conf are at
http://jeremy.kister.net/tmp/sip_debug/most_of_sip.conf
I have verified the same behavior with asterisk 1.6.1.12.
Ideas?
--
Jeremy Kister
http://jeremy.kister....
2003 Nov 06
2
this is the code that breaks outgoing calls on grandstream
Here is the diff from chan_sip.c 15 days ago and 16 days ago. 15 days ago is the point outgoing calls made via grandstream budgetone stopped working.
Any help on why it breaks? Any possible fix?
/tmp# diff asterisk/channels/chan_sip.c asterisk.works/channels/chan_sip.c
289d288
< int capability;
3921,3922d3919
< p->capability = user->capability;
2005 Mar 19
2
More HEAD wierdness (chan_sip, jitterbuffer/PLC problems)
...aller hears no music. Same with before, if I cycle hold/resume enough,
the phone locks up. Only the Cisco just drops the call.
The SIP debug of the conversation between * and the IP 600 (firmware
1.4.1) can be found below. I can provide the Cisco as well (7.2).
http://www.krisk.org/asterisk/sip_debug.log
Now for more on MOH. I am using native MOH with my files in ulaw
format. I setup an extension for running MusicOnHold, and have been
listening to my hold music for 8 minutes with my Polycom IP600, so I
know that * sees the files and can play them. However, if someone calls
in over IAX...