Summary: | ASTERISK-01738: SIP handset Transfer fails and Multi-line fails | ||
Reporter: | davet (davet) | Labels: | |
Date Opened: | 2004-06-01 10:07:57 | Date Closed: | 2011-06-07 14:10:46 |
Priority: | Minor | Regression? | No |
Status: | Closed/Complete | Components: | 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 |