Re: [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@default:1] Dial(SIP/181-000a, SIP/182) in new stack == Using SIP RTP CoS mark 5 -- Called SIP/182 -- SIP/182-000b is ringing -- SIP/182-000b is making progress passing it to SIP/181-000a -- SIP/182-000b answered SIP/181-000a -- Remotely bridging SIP/181-000a and SIP/182-000b == Spawn extension (default, 182, 1) exited non-zero on 'SIP/181-000a' 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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@default:1] Dial(SIP/181-000a, SIP/182) in new stack == Using SIP RTP CoS mark 5 -- Called SIP/182 -- SIP/182-000b is ringing -- SIP/182-000b is making progress passing it to SIP/181-000a -- SIP/182-000b answered SIP/181-000a -- Remotely bridging SIP/181-000a and SIP/182-000b == Spawn extension (default, 182, 1) exited non-zero on 'SIP/181-000a' 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [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@default:1] Dial(SIP/181-000a, SIP/182) in new stack == Using SIP RTP CoS mark 5 -- Called SIP/182 -- SIP/182-000b is ringing -- SIP/182-000b is making progress passing it to SIP/181-000a -- SIP/182-000b answered SIP/181-000a -- Remotely bridging SIP/181-000a and SIP/182-000b == Spawn extension (default, 182, 1) exited non-zero on 'SIP/181-000a' 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
