[asterisk-users] Conference calls wont traverse my trunk

[DadoMaker]

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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