| Summary: | ASTERISK-03853: Supervised transfer problems | ||
| Reporter: | Daniele Gallina (gallysoft) | Labels: | |
| Date Opened: | 2005-04-04 05:56:16 | Date Closed: | 2005-04-06 11:25:19 |
| Priority: | Minor | Regression? | No |
| Status: | Closed/Complete | Components: | Core/General |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ( 0) 2005_04_05_sipdebug.txt ( 1) sipdebug.txt | |
| Description: | when I try to transfer a call asterisk say me: -- Executing SetCallerID("SIP/20012-cb87", ""Gallina Daniele" <20012>") in new stack -- Executing Dial("SIP/20012-cb87", "SIP/20013") in new stack -- Called 20013 -- SIP/20013-034d is ringing -- SIP/20013-034d answered SIP/20012-cb87 -- Attempting native bridge of SIP/20012-cb87 and SIP/20013-034d -- Started music on hold, class '3psystem', on SIP/20013-034d Apr 4 12:12:39 NOTICE[4710]: chan_sip.c:5161 get_refer_info: Supervised transfer requested, but unable to find callid '48b5cce3d8e0d84f@192.168.0.53' Apr 4 12:12:40 NOTICE[4710]: chan_sip.c:5161 get_refer_info: Supervised transfer requested, but unable to find callid '48b5cce3d8e0d84f@192.168.0.53' Apr 4 12:12:42 NOTICE[4710]: chan_sip.c:5161 get_refer_info: Supervised transfer requested, but unable to find callid '48b5cce3d8e0d84f@192.168.0.53' Why? Any ideas? | ||
| Comments: | By: Mark Spencer (markster) 2005-04-04 09:27:36 Please include your sip debug as per the bug guidelines. By: Daniele Gallina (gallysoft) 2005-04-04 09:55:58 I have attached the sip debug but I don't know what there is in. My problem is this: with my Grandstream Budgetone 102 I call 4000 (echo test, or another telephone) and I press Flash and I compose another number, when I press Transfer Asterisk say me the error in the Description. By: Kevin P. Fleming (kpfleming) 2005-04-04 11:34:35 Your 'sip debug' is very hard to follow. If you can: - turn off 'qualify' - use 'sip debug x.x.x.x', where the x's are the IP address of the phone you are using - try not to have any other SIP traffic in/out of your server while you are generating the trace log - start and stop the log just before/after the call you are having trouble with, or indicate to us the specific call you had trouble with so we can ignore the remaining entries in the log edited on: 04-04-05 11:35 By: Daniele Gallina (gallysoft) 2005-04-05 01:52:56 I know that it is hard to follow ;-) I have attacched one more clear. - Called 4000 - Flash - Called 20014 - Press Transfer By: Kevin P. Fleming (kpfleming) 2005-04-05 18:59:28 What device is at 192.168.0.53? That address does not appear in the log until the REFER is sent, so there is no way Asterisk could possibly know about a call involving it. By: Brian West (bkw918) 2005-04-05 19:07:43 http://bugs.digium.com/bug_view_page.php?bug_id=0003710 This seems to be related. /b By: Daniele Gallina (gallysoft) 2005-04-06 01:34:44 192.168.0.53 is a Grandstream Budgetone 102 behind a nat. It connect to sip1.telezero.net with the user 20012. By: Kevin P. Fleming (kpfleming) 2005-04-06 11:24:25 Yes, this is the same problem (or very similar) to the one in bug ASTERISK-3683710. Please participate in testing the patches in that bug as a resolution to this problem. | ||