| Summary: | ASTERISK-17855: Endpoint call forwarding fails with congestion | ||
| Reporter: | rsw686 (rsw686) | Labels: | |
| Date Opened: | 2011-05-13 08:47:01 | Date Closed: | 2012-01-28 11:50:27.000-0600 |
| Priority: | Minor | Regression? | No |
| Status: | Closed/Complete | Components: | Channels/chan_sip/General |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| 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 | ||