Hi,
This is the first time I experience this problem with Asterisk:
all of a sudden SIP registrations stopped working. Active calls kept working but
new calls could not be established (I did NOT perform a "graceful
restart").
Besides, would a "restart gracefully" actually deny SIP registration?
I did not have a network issue because killing asterisk and starting it again
solved the problem.
How can I diagnose what happened to the SIP service (I checked the log but am
quite lost)?
Also, how can I do a simple command-line "check" to see that SIP
registrations are OK? I would like to use a SIP client (like sipsak) to perform
a simple registration from a custom bash script so I can quickly detect if this
problem occurs again and "auto-kill+restart" the asterisk process. I
know this sounds ugly but on my production server, it's better to bring the
whole system down and back up in as little time as possible.
Any suggestions?
Asterisk is 1.2.31.1
Some log lines:
Dec 23 13:13:16 DEBUG[11482] channel.c: Avoiding initial deadlock for
'SIP/4053-b4520e98'
Dec 23 13:13:16 WARNING[11482] channel.c: Avoided initial deadlock for
'0xb4302278', 9 retries!
Dec 23 13:13:43 VERBOSE[18837] logger.c: -- Executing
Dial("SIP/6174-b456d828", "SIP/4062|20|tTwWM(auto-blkvm)")
in new stack
Dec 23 13:13:43 NOTICE[18837] app_dial.c: Unable to create channel of type
'SIP' (cause 3 - No route to destination)
Dec 23 13:13:43 VERBOSE[18837] logger.c: == Everyone is busy/congested at this
time (1:0/0/1)
Dec 23 13:13:43 DEBUG[18837] app_dial.c: Exiting with DIALSTATUS=CHANUNAVAIL.
Thanks,
Vieri
Danny Nicholas
2009-Dec-23 19:16 UTC
[asterisk-users] how to check Asterisk SIP registration
"Sip show users" or "sip show peers" should do the trick,
but I'm not sure
about 1.2; all of my experience is in the 1.4 branch.
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Vieri
Sent: Wednesday, December 23, 2009 1:09 PM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] how to check Asterisk SIP registration
Hi,
This is the first time I experience this problem with Asterisk:
all of a sudden SIP registrations stopped working. Active calls kept working
but new calls could not be established (I did NOT perform a "graceful
restart").
Besides, would a "restart gracefully" actually deny SIP registration?
I did not have a network issue because killing asterisk and starting it
again solved the problem.
How can I diagnose what happened to the SIP service (I checked the log but
am quite lost)?
Also, how can I do a simple command-line "check" to see that SIP
registrations are OK? I would like to use a SIP client (like sipsak) to
perform a simple registration from a custom bash script so I can quickly
detect if this problem occurs again and "auto-kill+restart" the
asterisk
process. I know this sounds ugly but on my production server, it's better to
bring the whole system down and back up in as little time as possible.
Any suggestions?
Asterisk is 1.2.31.1
Some log lines:
Dec 23 13:13:16 DEBUG[11482] channel.c: Avoiding initial deadlock for
'SIP/4053-b4520e98'
Dec 23 13:13:16 WARNING[11482] channel.c: Avoided initial deadlock for
'0xb4302278', 9 retries!
Dec 23 13:13:43 VERBOSE[18837] logger.c: -- Executing
Dial("SIP/6174-b456d828", "SIP/4062|20|tTwWM(auto-blkvm)")
in new stack
Dec 23 13:13:43 NOTICE[18837] app_dial.c: Unable to create channel of type
'SIP' (cause 3 - No route to destination)
Dec 23 13:13:43 VERBOSE[18837] logger.c: == Everyone is busy/congested at
this time (1:0/0/1)
Dec 23 13:13:43 DEBUG[18837] app_dial.c: Exiting with
DIALSTATUS=CHANUNAVAIL.
Thanks,
Vieri
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I have asterisk 1.6.1.10 and a Rhino CB24 Channel Bank... A few channels seem to have locked up... If I plug an analog phone in the port, I get either dead air or a busy tone... Is there any way to reset this channel without restarting asterisk?