[Asterisk-Users] transfering calls, no ringing sent to caller
Jeremy Koski
jayk at danner.net
Fri Sep 30 16:58:56 MST 2005
I'm still trying to get this to work.
I tried downgrading my Cisco phones to an earlier SIP image from cisco,
thinking that might be the problem. Not it. Currently running SIP 7.5.
I noticed when I use the transfer feature, a ZOMBIE appears on the
channel, and the caller I'm transfering an extension or number to,
doesn't hear any ringing. When using blind transfer, no zombie, and the
ringing works!
Anybody have any ideas how I can resolve this? Here's the console output.
I was runninsg asterisk-1.2.0-beta1, but have upgraded to cvs head twice
in hopes to get this problem resolved. Still no luck.
-- Accepting call from '2532612594' to '2180650' on channel 0/1, span 1
-- Executing Dial("Zap/1-1", "SIP/120|20|t") in new stack
-- Called 120
-- SIP/120-ccab is ringing
-- SIP/120-ccab answered Zap/1-1
-- Started music on hold, class 'default', on channel 'Zap/1-1'
-- Executing NoOp("SIP/120-841d", "") in new stack
-- Executing Goto("SIP/120-841d", "intern-post|500|1") in new stack
-- Goto (intern-post,500,1)
-- Executing Dial("SIP/120-841d", "SIP/500|20|tr") in new stack
-- Called 500
-- SIP/500-bdae is ringing
-- Stopped music on hold on Zap/1-1
== Spawn extension (local, 2180650, 1) exited non-zero on 'SIP/120-841d<ZOMBIE>'
-- Channel 0/1, span 1 got hangup request
== Spawn extension (intern-post, 500, 1) exited non-zero on 'Zap/1-1'
-- Executing Hangup("Zap/1-1", "") in new stack
== Spawn extension (intern-post, h, 1) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
On Wed, 28 Sep 2005, Jeremy Koski wrote:
>
> This worked for me in earlier versions, prior to 1.2.0-beta1. When I
> transfer an incoming caller to another number or extension, the caller
> does not hear any ringing. I have tried to generate the ringing using the
> r feature from the dial command with no luck. If I specify m instead of r,
> the caller hears the onhold music.
>
> I do hear the ringing on my end until I hit the transfer button.
>
> I've tried several combinations, but nothing seems to work.
>
> Any thoughts on how I can get this issue resolved? I'm using Cisco 7960
> phones with asterisk-1.2.0-beta1.
>
>
>
> Thank you in advance!
>
> _______________________________________________
> --Bandwidth and Colocation sponsored by Easynews.com --
>
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
More information about the asterisk-users
mailing list