[Freeswitch-users] Incoming call audio problem

[Brian West]

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 89.18.32.19:5080 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 
>>>>> >;tag=2D49486C-1855 To: <sip: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 
>>>>>  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>;party=calling;screen=yes;privacy=off  
>>>>> Timestamp: 1205419179 Contact: <sip: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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>>>>>
>>>>
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>>
>
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