Sergio Serrano
2012-Jul-09 13:24 UTC
[asterisk-users] seems like call is picked and returned to me
Hi all
I hope that someone of you can solve this. Right now I'm stuck!!!!!
I'm using asterisk with some SIP extensions. Basically I want to
establish a call between desktop voip phone (ext 181) and embedded sip
system (ext 182)
All I can see in CLI is:
== Using SIP RTP CoS mark 5
-- Executing [182 at default:1] Dial("SIP/181-0000000a",
"SIP/182")
in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/182
-- SIP/182-0000000b is ringing
-- SIP/182-0000000b is making progress passing it to SIP/181-0000000a
-- SIP/182-0000000b answered SIP/181-0000000a
-- Remotely bridging SIP/181-0000000a and SIP/182-0000000b
== Spawn extension (default, 182, 1) exited non-zero on
'SIP/181-0000000a'
Seems like extension 182 (called ext) is getting call and passing them
another time to me 181 (origin call)
I've try it with siemens pbx and works as expected
cheers!
Sergio
Andres
2012-Jul-09 19:27 UTC
[asterisk-users] seems like call is picked and returned to me
On 7/9/2012 8:24 AM, Sergio Serrano wrote:> Hi all > > I hope that someone of you can solve this. Right now I'm stuck!!!!! > I'm using asterisk with some SIP extensions. Basically I want to > establish a call between desktop voip phone (ext 181) and embedded sip > system (ext 182) > > All I can see in CLI is: > == Using SIP RTP CoS mark 5 > -- Executing [182 at default:1] Dial("SIP/181-0000000a", "SIP/182") > in new stack > == Using SIP RTP CoS mark 5 > -- Called SIP/182 > -- SIP/182-0000000b is ringing > -- SIP/182-0000000b is making progress passing it to SIP/181-0000000a > -- SIP/182-0000000b answered SIP/181-0000000a > -- Remotely bridging SIP/181-0000000a and SIP/182-0000000b > == Spawn extension (default, 182, 1) exited non-zero on 'SIP/181-0000000a' >My guess is you need to add canreinvite=no to both SIP Peers in order to avoid the re-invite which apparently is what is happening. eRepublik - Join Me! http://www.erepublik.com/en/referrer/csredes> Seems like extension 182 (called ext) is getting call and passing them > another time to me 181 (origin call) > I've try it with siemens pbx and works as expected >
Olle E. Johansson
2012-Jul-10 08:28 UTC
[asterisk-users] seems like call is picked and returned to me
9 jul 2012 kl. 15:24 skrev Sergio Serrano:> Hi all > > I hope that someone of you can solve this. Right now I'm stuck!!!!! > I'm using asterisk with some SIP extensions. Basically I want to > establish a call between desktop voip phone (ext 181) and embedded sip > system (ext 182) > > All I can see in CLI is: > == Using SIP RTP CoS mark 5 > -- Executing [182 at default:1] Dial("SIP/181-0000000a", "SIP/182") > in new stack > == Using SIP RTP CoS mark 5 > -- Called SIP/182 > -- SIP/182-0000000b is ringing > -- SIP/182-0000000b is making progress passing it to SIP/181-0000000a > -- SIP/182-0000000b answered SIP/181-0000000a > -- Remotely bridging SIP/181-0000000a and SIP/182-0000000b > == Spawn extension (default, 182, 1) exited non-zero on 'SIP/181-0000000a' > > Seems like extension 182 (called ext) is getting call and passing them > another time to me 181 (origin call) > I've try it with siemens pbx and works as expected >It's very hard to see what's happening without seeing the SIP logs. You just see that something went wrong in the process of setting up the bridge. /O