[Freeswitch-users] Incoming call audio problem

[Josip Djuricic]

No problem people, here are all the traces. If you want I can include 
tcpdump with rtp.

To clarify few things, setup is like this.

PSTN <-> CISCO AS5300 (10.0.0.19) <-> FS (89.18.32.19) <-> VoipPhone 
(89.18.42.151)

There is no NAT between any of them.

I call from PSTN (38513391900) to Voip Phone behind FS (38514001148)

Brian West wrote:
> I can almost 100% say that sofia is sending an IP6 in the SDP to the  
> Cisco.  Can you provide a sip trace for this?
>
> /b
>
> On Mar 13, 2008, at 9:12 AM, Josip Djuricic wrote:
>
>   
>> Hi there,
>>
>> I was already writing about this problem, and I still didn't managed  
>> to
>> fix it. I thought it was a NAT problem, but now without NAT the  
>> problem
>> still remains.
>>
>> The error is like this, when I call from pstn, the call goes trough
>> Cisco AS5300, then goes to FS, then to it's final destination - Ip  
>> Phone.
>>
>> Now you can answer the phone, but you can't hear anything on any side.
>> And FS returns lots of SIP 415 (unsupported media type) to Cisco  
>> AS5300.
>>
>> When you call from Voip phone to the same pstn number it all works  
>> well,
>> so I'm starting to be confused more and more. (and in this case  
>> there is
>> no 415 messages).
>>
>> I believe cisco is sending something that FS can't understand, in the
>> INVITE's but that's just a wild guess cause I ain't sure what can FS
>> exactly understand very well.
>>
>> Whoever want's I can send him a tcpdump, it's about 7mb tarred, so I
>> can't send it to the list.
>>
>> Thank you all,
>>
>> Josip
>>
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>
>
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