Summary:ASTERISK-07685: transfering call originating from agent channel fails
Reporter:vortex_0_o (vortex_0_o)Labels:
Date Opened:2006-09-06 16:54:00Date Closed:2007-02-19 11:12:38.000-0600
Versions:Frequency of
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!