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Summary:ASTERISK-01738: SIP handset Transfer fails and Multi-line fails
Reporter:davet (davet)Labels:
Date Opened:2004-06-01 10:07:57Date Closed:2011-06-07 14:10:46
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Core/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) sip-multi-line-prob.txt
Description:Using ACT Telecom P104SLD SIP handset it is not possible to use the multi-line feature. Transferring calls using the transfer button also fail since this seems to use the same technique.


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

SIP handset works fine. "Hold" uses the same technique as Transfer and Multi-line but as long as you don't start another session, you can retrieve the call from Asterisk by pressing line key M1. I've attached an annotated SIP DEBUG of a multi-line session. After the call gets dropped at the end, there is an unattached channel left open that has to be killed with "soft hangup".
Comments:By: Mark Spencer (markster) 2004-06-01 15:59:22

You're going to have to find someone on IRC.  I don't immediately see any way that an "INVITE" can come in and never get any sort of response at all.

By: Mark Spencer (markster) 2004-06-03 00:40:32

If you're still interested in this, please find me on IRC (kram, irc.freenode.net, #asterisk).  thanks.

By: davet (davet) 2004-06-03 05:42:13

Thanks for the offer. Just had a look on #asterisk and you were away. What TZ are you in and I'll try at a more appropriate time. Cheers.

By: Mark Spencer (markster) 2004-06-03 08:33:21

Central time zone.

By: Mark Spencer (markster) 2004-06-03 11:35:05

The phone is not sending the INVITE with the proper sequence number -- it's using a single sequence number space for all calls which is not valid in the SIP protocol (but would be fine with MGCP).

See RFC3261 Section 12.2.1.1