Jeremy Kister
2010-Aug-29 07:25 UTC
[asterisk-users] evil disconnect of call with cisco 1760
I have asterisk 1.6.2.11 talking to a cisco 1760 running ipvoicek9-mz.124-25b. whenever a call goes through the 1760's FXO or FXS (in or out) there is a 915 second maximum call time due to asterisk hanging up the call because of a "critical packet" being missed. I read doc/sip-retransmit.txt and I don't see anything there that is helpful to my situation - the asterisk box is on the same subnet as the c1700; there are no nat/firewalls/sbcs in the middle. at ~15:15 the asterisk console reads: WARNING[2492]: chan_sip.c:3778 retrans_pkt: Maximum retries exceeded on transmission CB674A02-B25C11DF-B6D5A08D-652FE73E at 10.9.1.9 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.net./
Jeremy Kister
2010-Sep-05 23:28 UTC
[asterisk-users] evil disconnect of call with cisco 1760
On 9/4/2010 1:31 AM, Jeremy Kister thought: > On 8/29/2010 3:25 AM, Jeremy Kister wrote:>> whenever a call goes through the 1760's FXO or FXS (in or out) there is >> a 915 second maximum call time due to asterisk hanging up the call >> because of a "critical packet" being missed.> > hmm, either no one has any clue/suggestions or they just don't care > about the issue - I better figure it out myself. I wonder if it has > to do do with progress indicators on the 1760. Thanks Jeremy, that was it! I ended up putting: progress_ind progress enable 8 progress_ind connect enable 8 on each dial-peer pots, and then: progress_ind setup enable 3 progress_ind connect enable 8 on each dial-peer voip. Problem seems solved. -- Jeremy Kister http://jeremy.kister.net./