[Freeswitch-users] Incoming call audio problem

Thu Mar 13 14:52:11 PDT 2008

i've sent you an email to brian.west at mac.com with trace

Brian West wrote:
> pastebin the logs 
>
> /b
>
> On Mar 13, 2008, at 3:47 PM, Josip Djuricic wrote:
>
>> Ok, seems like I can't include any file larger then 100kb, so I can 
>> send the trace to people that want to check it out.
>>
>> As i said I still get no audio, but I'm rid of sip 415. From the 
>> trace you can see that there is a RTP from softphone trough FS, all 
>> the way to Cisco AS5300, but it doesn't get out on PSTN. Also FS 
>> get's no RTP from AS5300. This is all happening when I call from PSTN 
>> side into Voip network. When the call is coming from voip phone 
>> everything works well.
>>
>> Thank you all
>>
>> Djuricic wrote:
>>> Hi there,
>>>
>>> I am including the new trace, this time with sip and rtp.
>>>
>>> I hope that somebody can see something I don't, cause there is still 
>>> no audio.
>>>
>>> This time I am doing the trace behind a NAT with a x-lite softphone 
>>> because I am at home.
>>>
>>> Josip Djuricic wrote:
>>>> I meant GTD ;)
>>>>
>>>> Anyway, the traces now looks alot better, no SIP 415 anymore, but 
>>>> there is still no voice, and rtp seems to be going only from the 
>>>> phone to the FS
>>>>
>>>> Josip Djuricic wrote:
>>>>> I can disable GDP but still no audio, I'm checkin the traces now
>>>>>
>>>>> Brian West wrote:
>>>>>> Oh this bug has already been sent to MikeJ and Anthm. This is one 
>>>>>> strange invite. anyway its being worked on. You might want to see 
>>>>>> what in your config causes the addition of the application/gtd 
>>>>>> type. I suspect 'supplementary-service pass-through'
>>>>>>
>>>>>> /b
>>>>>>
>>>>>>
>>>>>> INVITE sip:38514001148 at 89.18.32.19 
>>>>>> <mailto:38514001148 at 89.18.32.19>:5080 SIP/2.0 Via: SIP/2.0/UDP 
>>>>>> 10.0.0.19:5060;x-route-tag="tgrp:Iskra" From: 
>>>>>> <sip:38513391900 at 10.0.0.19 
>>>>>> <mailto:38513391900 at 10.0.0.19>>;tag=2D49486C-1855 To: 
>>>>>> <sip:38514001148 at 89.18.32.19 <mailto:38514001148 at 89.18.32.19>> 
>>>>>> Date: Thu, 13 Mar 2008 14:39:39 GMT Call-ID: 
>>>>>> 253738A2-F04211DC-81719B71-D67790FB at 10.0.0.19 
>>>>>> <mailto:253738A2-F04211DC-81719B71-D67790FB at 10.0.0.19> Supported: 
>>>>>> timer,100rel Min-SE: 1800 Cisco-Guid: 
>>>>>> 624255889-4030861788-2171509617-3598160123 User-Agent: 
>>>>>> Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, 
>>>>>> ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO CSeq: 101 
>>>>>> INVITE Max-Forwards: 6 Remote-Party-ID: 
>>>>>> <sip:38513391900 at 10.0.0.19 
>>>>>> <mailto:38513391900 at 10.0.0.19>>;party=calling;screen=yes;privacy=off 
>>>>>> Timestamp: 1205419179 Contact: <sip:38513391900 at 10.0.0.19 
>>>>>> <mailto:38513391900 at 10.0.0.19>:5060> Expires: 180 Allow-Events: 
>>>>>> telephone-event MIME-Version: 1.0 Content-Type: 
>>>>>> multipart/mixed;boundary=uniqueBoundary Content-Length: 523 
>>>>>> --uniqueBoundary Content-Type: application/sdp v=0 
>>>>>> o=CiscoSystemsSIP-GW-UserAgent 5947 1401 IN IP4 10.0.0.19 s=SIP 
>>>>>> Call c=IN IP4 10.0.0.19 t=0 0 m=audio 16850 RTP/AVP 8 c=IN IP4 
>>>>>> 10.0.0.19 a=rtpmap:8 PCMA/8000 a=ptime:20 --uniqueBoundary 
>>>>>> Content-Type: application/gtd Content-Disposition: 
>>>>>> signal;handling=optional IAM, PRN,isdn*,,NET5*, USI,rate,c,3,c,1 
>>>>>> USI,lay1,alaw TMR,02 CPN,04,,1,14001148 CGN,04,,1,y,4,13391900 
>>>>>> CPC,09 FCI,,,,,,,y, GCI,25356391f04211dc816e9b71d67790fb 
>>>>>> --uniqueBoundary--
>>>>>> On Mar 13, 2008, at 9:51 AM, Josip Djuricic wrote:
>>>>>>
>>>>>>> supplementary-service pass-through
>>>>>>
>>>>>> ------------------------------------------------------------------------
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