[Asterisk-Users] Does congestion exit on a different priority?
Paul Rodan
asterisk at glitch.cc
Tue Jan 4 10:58:41 MST 2005
Customer is having problems with his internet connection, I have in my
context:
[jimballboutiques]
.
exten => 1235690251,1,SetGroup(customer)
exten => 1235690251,2,CheckGroup(3)
exten => 1235690251,3,Dial(SIP/jimball,20,r)
exten => 1235690251,4,VoiceMail(u1235690251 at jimballboutiques)
exten => 1235690251,103,VoiceMail(u1235690251 at jimballboutiques)
.
Now I've had it before where phones are unplugged and such and it goes
straight to voicemail, so this customers calls should go to voicemail,
right? Nope. It's doing something odd, when I call the number (1235690251) I
get this in the logs, and just dead air, no ringing or anything:
main-nuvoip*CLI>
Jan 4 12:46:43 NOTICE[1089849920]: chan_sip.c:7889 sip_poke_noanswer: Peer
'jimball' is now UNREACHABLE!
Jan 4 12:46:46 WARNING[1159722432]: channel.c:472 ast_channel_walk_locked:
Avoided initial deadlock for 'SIP/123.123.94.4-44c269e0', 10 retries!
main-nuvoip*CLI>
main-nuvoip*CLI>
-- Executing Goto("SIP/123.123.94.4-44cc0ca0",
"jimballboutiques|1235690251|1") in new stack
-- Goto (jimballboutiques,1235690251,1)
-- Executing SetGroup("SIP/123.123.94.4-44cc0ca0", "customer") in new
stack
-- Executing CheckGroup("SIP/123.123.94.4-44cc0ca0", "3") in new stack
Jan 4 12:46:50 WARNING[1116214592]: channel.c:472 ast_channel_walk_locked:
Avoided initial deadlock for 'SIP/123.123.94.4-44cc0ca0', 10 retries!
-- Executing Dial("SIP/123.123.94.4-44cc0ca0", "SIP/jimball|20|r") in
new stack
Jan 4 12:46:50 NOTICE[1116214592]: app_dial.c:743 dial_exec: Unable to
create channel of type 'SIP'
== Everyone is busy/congested at this time
-- Executing Congestion("SIP/123.123.94.4-44cc0ca0", "") in new stack
== Spawn extension (jimballboutiques, 1235690251, 104) exited non-zero on
'SIP/123.123.94.4-44cc0ca0'
-- Executing Hangup("SIP/123.123.94.4-44cc0ca0", "") in new stack
== Spawn extension (jimballboutiques, h, 1) exited non-zero on
'SIP/123.123.94.4-44cc0ca0'
The "123.123.94.4" IP is the IP (changed) of my Cisco 3640 Router, which my
PRI's plug into. It then sends it via SIP to my Asterisk server. I have no
idea about the deadlock junk, just started showing up in the logs one day
and it never really affected the system so I ignored it. Tried
stopping/starting asterisk and rebooting to no avail, I don't think it's the
problem here. See where it says "Executing Congestion" I don't get it, I
don't have the word "Congestion" in there anywhere. I also see where it
tried to go to priority 104, which does not exist. I'm thinking that
SIP/jimball is not returning unavailable, just that it's really lagged, so
instead of exiting w/ priority +1 it's exiting with priority +101 instead.
Is this right? I know that if the phone is not available at all, like it's
been smashed by a sledgehammer, jumped on and thrown into trash compactor,
then it will exit with +1 when Asterisk is unable to dial it.
Thanks for your time.
Regards,
Paul
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