[Freeswitch-users] Incoming call audio problem

Josip Djuricic josip.djuricic at primatel.hr
Thu Mar 13 13:47:04 PDT 2008


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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