[asterisk-dev] ast_sched_add() question!!!

[Rizwan Hisham]

Hi all,
Im trying to understand the behaviour of asterisk when a sip call is
initiated (sip invite is sent to the callee) but the callee is behind a
bad-nat and becoz of that either our packets dont reach the callee or the
callee does not responde to us properly, my asterisk does not get any reply
of the invite packet at all even after 5 re-invites have been sent.

The bottom line is, as the call could not be setup the caller's channel
hangsup without any problem, but the callee channel does not hangup, which
means its inuse limit for peer never goes back to 0, which means if the
caller tries again and again the callee's inuse limit to recieve calls keeps
on increasing utill it reaches the max limit (call-limit=2 for peer in
sip.conf) and never decremented, which means the callee can no longer
recieve any calls irrespective of the fact that maybe its nat conditions
have improved and its now able to recieve calls, or not.

I see ast_sched_add() is called in sip_call() function for auto-congestion,
which mean auto-congestion is declared after timeout. I need to ask you guys
that in the above described scenario, shoulden't asterisk declare
auto-congestion if it does not get any reply? or does it?

I have noticed a strange thing when i saw a debug on asterisk cli under the
above scenario:

[Oct 24 06:50:34] DEBUG[10748]: chan_sip.c:3312 sip_hangup: Hangup call
SIP/gusgus-009a7510, SIP callid
6b25d4d65b65986436195f224b67c365 at 69.90.174.98)
>>[Oct 24 06:50:34] DEBUG[10748]: chan_sip.c:3321 sip_hangup:
update_call_counter(gusgus)
- decrement call limit counter on hangup
[Oct 24 06:50:34] DEBUG[10748]: chan_sip.c:3003 update_call_counter:
Updating call counter for outgoing call
>>[Oct 24 06:50:34] DEBUG[10748]: chan_sip.c:3051 update_call_counter: Call
to peer 'gusgus' removed from call limit 2
[Oct 24 06:50:34] DEBUG[10340]: chan_sip.c:15247 sip_devicestate: Checking
device state for peer gusgus
>>[Oct 24 06:50:34] DEBUG[10748]: chan_sip.c:2071 __sip_ack: Acked pending
invite 102
[Oct 24 06:50:34] DEBUG[10748]: chan_sip.c:2089 __sip_ack: Stopping
retransmission on '6b25d4d65b65986436195f224b67c365 at 69.90.174.98' of Request
102: Match Not Found
>>[Oct 24 06:50:34] DEBUG[10748]: chan_sip.c:3003 update_call_counter: Updating
call counter for outgoing call
>>[Oct 24 06:50:34] DEBUG[10748]: chan_sip.c:3077 update_call_counter: Call
to peer 'gusgus' is 1 out of 2
[Oct 24 06:50:34] DEBUG[10340]: chan_sip.c:15247 sip_devicestate: Checking
device state for peer gusgus
[Oct 24 06:50:34] DEBUG[10340]: chan_sip.c:15247 sip_devicestate: Checking
device state for peer gusgus
[Oct 24 06:50:34] NOTICE[10748]: cdr_addon_mysql.c:1444 mysql_log:
cdr_mysql: Inserting a CDR record.

You can see in the debug output that after decrementing the call limit it
increments it again. Maybe because there is a pending invite, but y, Its
retransmission should be stopped but no.

So can anybody help me in preventing the chanel from stucking?

-- 
Best Regards
Rizwan Hisham
Software Engineer
Axvoice Inc.
www.axvoice.com
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