Found a syntax err in my dialplan on the far side Asterisk config. Thanks, Dado
On Thu, Jun 27, 2013 at 10:41 AM, DadoMaker <[email protected]> wrote: > The cogerence works but doesnt go over my trunk. Its bypassing and the > codec is PCM of phone. But in phone to phone call, the rtp traverses the > trunk and I capture gsm packets to verify. > > The sip debug for conf call setup and leave: > *CLI> == Using SIP RTP CoS mark 5 > -- Executing [5777@public:1] Goto("SIP/127.0.0.1-00000012", > "conf-confDemo,join,1") in new stack > -- Goto (conf-confDemo,join,1) > -- Executing [join@conf-confDemo:1] > ConfBridge("SIP/127.0.0.1-00000012", "1") in new stack > > 0x7f006c015150 -- Probation passed - setting RTP source address > to 192.168.100.100:4002 > -- <SIP/127.0.0.1-00000012> Playing 'conf-onlyperson.ulaw' (language > 'en') > -- <SIP/127.0.0.1-00000012> Playing 'confbridge-join.ulaw' (language > 'en') > -- <Bridge/0x7f0058001af8-input> Playing 'confbridge-join.slin' > (language 'en') > == Using SIP RTP CoS mark 5 > -- Executing [5777@default:1] Goto("SIP/5700-00000013", > "conf-confDemo,join,1") in new stack > -- Goto (conf-confDemo,join,1) > -- Executing [join@conf-confDemo:1] ConfBridge("SIP/5700-00000013", > "1") in new stack > > 0x7f006c031d90 -- Probation passed - setting RTP source address > to 127.0.0.1:4004 > -- <SIP/5700-00000013> Playing 'confbridge-join.ulaw' (language 'en') > > 0x7f006c031d90 -- Switching RTP source address to > 192.168.1.10:4004 > -- <Bridge/0x7f0058001af8-input> Playing 'confbridge-join.slin' > (language 'en') > -- <Bridge/0x7f0058001af8-input> Playing 'confbridge-leave.slin' > (language 'en') > -- <Bridge/0x7f0058001af8-input> Playing 'confbridge-leave.slin' > (language 'en') > > Thanks, > Dado > > > On Thu, Jun 27, 2013 at 10:36 AM, Steve Totaro < > [email protected]> wrote: > >> >> >> >> On Thu, Jun 27, 2013 at 9:53 AM, DadoMaker <[email protected]> wrote: >> >>> My conference call wont go thru my SIP trunk. I may be missing a >>> dialplan configuration setting as my PCM phone to phone calls go over the >>> (GSM) tunk. >>> >>> >>> The server with the conference: >>> exten => 5777,1,GoTo(conf-confDemo,join,1) >>> [conf-confDemo] >>> exten => join,1,ConfBridge(confDemo/S/1) >>> >>> The server from which some users dial in from: >>> exten => 5777,1,Dial(SIP/$EXTEN}@200_PBX) >>> >>> Any insight appreciated. >>> >>> Thanks, >>> >>> Dado >>> >>> >> Dado, subject sounds like a personal problem. Sorry couldn't resist. >> >> How about some CLI debug info while trying a call? >> >> Thanks, >> Steve T >> >> -- >> _____________________________________________________________________ >> -- 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 >> > >
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