[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:                     feedback
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:              03-07-2008 16:34 CST
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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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---------------------------------------------------------------------- 
 atrash - 03-07-08 16:34  
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So after some packet sniffing, it appears that this is caused by a
discrepency in the way that the RFC is interpreted.

We're attached to a MetaSwitch UC9000.

When the OK is sent in step http://bugs.digium.com/view.php?id=3 above, it never
reaches the switch (packet
gets lost), the switch does not reply with an ACK, and then 2.1 seconds
later sends anoter INVITE.

Apparently, the way that MetaSwitch has interpreted the RFC dictates that
this reinvite gets a new CSeq.  Asterisk however, expects re-transmitted
INVITEs to have the SAME CSeq.

Anyone have thoughts on this? 

Issue History 
Date Modified   Username       Field                    Change               
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03-07-08 16:34  atrash         Note Added: 0083632                          
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