[asterisk-bugs] [Asterisk 0011727]: Sending 503 after 100 trying and BYE message

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=11727 
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Reported By:                gasparz
Assigned To:                
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Project:                    Asterisk
Issue ID:                   11727
Category:                   Channels/chan_sip/General
Reproducibility:            random
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.4.13 
SVN Branch (only for SVN checkouts, not tarball releases): N/A  
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             01-10-2008 08:01 CST
Last Modified:              01-22-2008 02:04 CST
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Summary:                    Sending 503 after 100 trying and BYE message
Description: 
Asterisk sometimes sends 503 after it receives a 100 trying or a BYE
message. Perhaps after other messages but I can confirm only these 2.

Let's get down to the problems:

sending 503 after 100 trying

2008-01-08 09:00:14 INVITE sip:number at ip SIP/2.0. sending
2008-01-08 09:00:14 SIP/2.0 100 Trying. receiving
2008-01-08 09:00:14 SIP/2.0 503 Server error. sending
2008-01-08 09:00:15 INVITE sip:number at ip SIP/2.0. sending
2008-01-08 09:00:15 SIP/2.0 100 Trying. receiving
2008-01-08 09:00:15 SIP/2.0 503 Server error. sending
2008-01-08 09:00:16 INVITE sip:number at ip SIP/2.0. sending
2008-01-08 09:00:16 SIP/2.0 100 Trying. receiving
2008-01-08 09:00:16 SIP/2.0 503 Server error. sending
2008-01-08 09:00:18 INVITE sip:number at ip SIP/2.0. sending
2008-01-08 09:00:18 SIP/2.0 100 Trying. receiving
2008-01-08 09:00:18 SIP/2.0 183 Session Progress. receiving
2008-01-08 09:00:22 SIP/2.0 200 OK. sending
2008-01-08 09:00:22 ACK sip:number at ip:5060 SIP/2.0. receiving
2008-01-08 09:03:08 BYE sip:number at ip5060 SIP/2.0. sending
2008-01-08 09:03:08 SIP/2.0 200 OK. receiving

sending 503 after BYE

2008-01-09 06:02:55 INVITE sip:+number at ip:port SIP/2.0. receiving
2008-01-09 06:02:55 SIP/2.0 100 Trying. sending
2008-01-09 06:02:55 SIP/2.0 200 OK. sending
2008-01-09 06:02:55 ACK sip:+number at ip SIP/2.0. receiving
2008-01-09 06:05:18 BYE sip:+number at ip SIP/2.0. receiving
2008-01-09 06:05:18 SIP/2.0 503 Server error. sending
2008-01-09 06:08:44 BYE sip:+number at ip:port;transport=udp SIP/2.0. 
2008-01-09 06:08:44 SIP/2.0 481 Call Leg/Transaction Does Not Exist.

I was also able to capture the cli output:
[Jan  9 06:05:18] ERROR[4667]: chan_sip.c:15304 sipsock_read: We could NOT
get the channel lock for SIP/ip-b7d0dce8!
[Jan  9 06:05:18] ERROR[4667]: chan_sip.c:15305 sipsock_read: SIP
transaction failed: call_id

 
the second case is the major issue because in our setup there is an
outbound call after the inbound. The hangup is on the inbound call and
asterisk does not "sense" this causing the outbound call to go until the
call times out (fortunatly we have a max_call duration)

To reproduce the issues is not easy, it doesn't happen for each call :(,
it depends on I don't know what.

The setup is something like (this is a simplified version because it's a
complicated ivr script):

exten _X.,1,answer
exten _X.,2,playback
exten _X.,3 dial <- during this dial we have the 503s but the dial is not
exiting
exten _X.,4,hangup

the 503 after 100 trying happens on the outbound leg, the 503 after bye on
the inbound leg (a bigger problem from our point of view).

It seems like a timeout when trying to aquire a lock to handle the
incomming packet. Because we send the 503 the party thinks we confirmed the
packet and does not resend it. Then for the invite we time out (asterisk is
not taking into account the 100 trying it received) and resend the invite
,this is why it isn't such a big issue.

For accurate packet capture I suggest a separate logging (ngrep, etheral
whatever) on a production type (high traffic) server and the problem will
occure sooner or later :)

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---------------------------------------------------------------------- 
 gasparz - 01-22-08 02:04  
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Hi,

Upgraded to 1.4.17 and the problem stopped.

Thanks 

Issue History 
Date Modified   Username       Field                    Change               
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01-22-08 02:04  gasparz        Note Added: 0080981                          
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