|Summary:||ASTERISK-01738: SIP handset Transfer fails and Multi-line fails|
|Date Opened:||2004-06-01 10:07:57||Date Closed:||2011-06-07 14:10:46|
|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 220.127.116.11