Summary:ASTERISK-17855: Endpoint call forwarding fails with congestion
Reporter:rsw686 (rsw686)Labels:
Date Opened:2011-05-13 08:47:01Date Closed:2012-01-28 11:50:27.000-0600
Versions:Frequency of
Environment:Attachments:( 0) debug_call_forward.txt
Description:I have tested with both Polycom and Cisco phones. When configuring the phone to forward all calls, calling that phone results in a congestion dial status. I also receive "Failed to authenticate on INVITE" on the console. It looks like asterisk is creating the INVITE to itself since the call is forward to number@asteriskfqdn by the phone. Shouldn't Asterisk go ahead and execute the dialplan for the forwarded number as if it was called from a phone then bridge the channels?

I have attached debug output. I call from SIP/1000, which was picked up by SIP/1050. SIP/1050 is a Polycom phone with call forward enable to 2002. I have both a SIP endpoint at 2002 and dialplan extension for 2002.
Comments:By: Leif Madsen (lmadsen) 2011-05-17 07:37:01

You're definitely going to need to provide your sip.conf file and dialplan to allow someone to reproduce this effectively.

By: Matt Jordan (mjordan) 2011-12-15 09:06:51.333-0600

As Leif mentioned, we will need your sip.conf and extensions.conf to move this issue forward.

By: Paul Belanger (pabelanger) 2012-01-28 11:50:10.675-0600

Suspended due to lack of activity. Please request a bug marshal in #asterisk-bugs on the IRC network irc.freenode.net to reopen the issue should you have the additional information requested.  Further information can be found at http://www.asterisk.org/developers/bug-guidelines