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
Versions:Frequency of
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.


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