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Summary:ASTERISK-01837: SIP Registration with Softswitch header information missing
Reporter:unknownLabels:
Date Opened:2004-06-16 00:56:09Date Closed:2008-01-15 14:58:23.000-0600
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Core/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) Asterisk-Xlite-sip-debug.txt
Description:We are trying to get asterisk to register with a Entice softswitch and receving a "603 Decline" error. Registering with the softswitch using a Sipura SPA-2000 and X-Lite works fine but not with Asterisk.

We have noticed that the SIP Debug that there are some differences between the registration of Asterisk and other IP phones/gateways that seem to be making the entice softswitch return this "601" error.

Asterisk seems to be missing the rport in the via header and also the contact field. (See text file attachment for details). The caller ID field is also different

I have also included part of my sip.conf file.

Any ideas?

****** ADDITIONAL INFORMATION ******

We have the same problem on both our Asterisk boxes. Both are using the latest CVS-Head 06/15/04
Comments:By: Mark Spencer (markster) 2004-06-16 08:53:10

the word "rport" does not occur in RFC3261.  It's part of RFC3581 which says "Send it back to the same port you saw this come from".  We could probably safely include it in all our Via messages, but I think it's extremely unlikely that this is your problem. It's easy enough to add it in, but I'm guessing they might be keeping you tied to certain SIP agents?  You can also change your SIP user agent and see if that makes a difference, but I have no reason to suspect this is a bug.

By: Mark Spencer (markster) 2004-06-16 09:10:06

Okay I added RFC3581 support (both directions -- it's basically like asking for "nat=yes")

By: Mark Spencer (markster) 2004-06-16 09:14:18

Not sure "fixed" is the right word, but implemented the RFC requested.

By: Digium Subversion (svnbot) 2008-01-15 14:58:23.000-0600

Repository: asterisk
Revision: 3224

U   trunk/channels/chan_sip.c

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r3224 | markster | 2008-01-15 14:58:23 -0600 (Tue, 15 Jan 2008) | 2 lines

Enable support for RFC3581 (bug ASTERISK-1837)

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http://svn.digium.com/view/asterisk?view=rev&revision=3224