[asterisk-users] Conference calls wont traverse my trunk

[DadoMaker]

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 <dadomaker at gmail.com> 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 at public:1] Goto("SIP/127.0.0.1-00000012",
> "conf-confDemo,join,1") in new stack
>     -- Goto (conf-confDemo,join,1)
>     -- Executing [join at 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 at default:1] Goto("SIP/5700-00000013",
> "conf-confDemo,join,1") in new stack
>     -- Goto (conf-confDemo,join,1)
>     -- Executing [join at 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 <
> stotaro at totarotechnologies.com> wrote:
>
>>
>>
>>
>> On Thu, Jun 27, 2013 at 9:53 AM, DadoMaker <dadomaker at gmail.com> 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
>>
>> --
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