| Summary: | ASTERISK-07685: transfering call originating from agent channel fails | ||
| Reporter: | vortex_0_o (vortex_0_o) | Labels: | |
| Date Opened: | 2006-09-06 16:54:00 | Date Closed: | 2007-02-19 11:12:38.000-0600 |
| Priority: | Minor | Regression? | No |
| Status: | Closed/Complete | Components: | Channels/chan_sip/Transfers |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ( 0) debug.txt | |
| Description: | In the attached file 1. Agent/2000 on SIP/1012 originates a call going to 07966000000 this goes out via peer "mags" 2. SIP/1012 tranfers the active call to SIP/1015 and the call fails/is dropped immediately. - agent callbacklogin on sip hard phones - transfer using hardphones - not asterisk features | ||
| Comments: | By: vortex_0_o (vortex_0_o) 2006-09-06 17:52:18 same result when canreinvite=no By: Olle Johansson (oej) 2006-10-25 14:48:13 Can you please try with 1.4 beta and see if the behaviour has changed? Thanks. By: Steve Murphy (murf) 2006-11-16 17:44:07.000-0600 Any progress here? Have you been able to try this, vortex_O_o? By: vortex_0_o (vortex_0_o) 2006-11-21 03:42:07.000-0600 waiting for a free weekend to retest ;) By: Steve Murphy (murf) 2007-01-04 10:53:19.000-0600 vortex_O_o -- How goes the battle? Now, when you get a propitious weekend, you can test against 1.4.0 (not beta!)... Might you get a chance soon? By: Joshua C. Colp (jcolp) 2007-02-16 19:29:38.000-0600 Fixed in 1.2 as of revision 55073, 1.4 as of revision 55086, and trunk as of revision 55087. Peace! | ||