[asterisk-bugs] [Asterisk 0012091]: Asterisk sends 491 Pending for a new INVITE

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=12091 
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Reported By:                atrash
Assigned To:                
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Project:                    Asterisk
Issue ID:                   12091
Category:                   Channels/chan_sip/General
Reproducibility:            sometimes
Severity:                   major
Priority:                   normal
Status:                     ready for testing
Asterisk Version:           1.4.17 
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:             02-27-2008 15:12 CST
Last Modified:              07-03-2008 11:46 CDT
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Summary:                    Asterisk sends 491 Pending for a new INVITE
Description: 
Please help!!!

AsteriskNOW v 1.0.1 is setup with multiple phones running SIP.  

The server itself is communicating with our provider in SIP as well, and
incoming/outgoing works great....

Except that outgoing calls get dropped in random time intervals ranging
from between less than a minute to over 15 minutes.

Each time they get dropped, the SIP Debugging shows the same thing...
Here's a quick overview, with the detailed sip debugging attached.


So here's what happened for the last call:
1. The provider sends an INVITE:
INVITE sip:6787721321 at 10.212.7.226 SIP/2.0
CSeq: 778024721 INVITE

2. Asterisk responds with a Trying:
SIP/2.0 100 Trying 
CSeq: 778024721 INVITE

3. Asterisk completes and sends the OK:
SIP/2.0 200 OK
CSeq: 778024721 INVITE

4. Provider sends another INVITE with a different CSeq:
INVITE sip:6787721321 at 10.212.7.226 SIP/2.0
CSeq: 778024722 INVITE

5. Asterisk thinks that this new invite is pending, and responds with:
SIP/2.0 491 Request Pending
CSeq: 778024722 INVITE

6. The provider ACKS this pending:
ACK sip:6787721321 at 10.212.7.226 SIP/2.0
CSeq: 778024722 ACK

7. The provider doesn't receive a completion, and responds with a BYE (
terminating the call :( ):
BYE sip:6787721321 at 10.212.7.226 SIP/2.0


Now, I'm left wondering why it replies with a 491 when the invite is not
still in processing....


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---------------------------------------------------------------------- 
 oej - 07-03-08 11:46  
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No, the issue here is that we do not accept a re-invite until the first
invite transaction is done and over. Before 4 in your scenario above, we
must have an ACK from the other party. If that doesn't arrive, we simply
deny any new INVITEs since the session is not UP yet. This is perfectly
good according to the RFC.

There's no bug in Asterisk in this bug report. 

If the second invite is ment to be a retransmission, it should have the
same CSEQ. In this case, it's a re-invite in an existing session. 

Issue History 
Date Modified   Username       Field                    Change               
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07-03-08 11:46  oej            Note Added: 0089692                          
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