[Freeswitch-users] Intermittent Audio Problem - Incomming Calls

[David Ponzone]

Brian,

twice you say "the IVR doesn't seem to hear anything the caller is saying".
I suppose that means it is an assumption because it does not react to DTMFs ?
Do you have the possibility in this IVR to record the call ?
This way, you would be sure.

Also you say that Wireshark tells you the RTP traffic is fine.
Even when the call is not working ?
Do you notice anything different in this RTP ?
You may also take a RTP trace between FS and IVR of a non-working call, analyse the RTP flows and save the payload as raw to a file.
You should be able to listen to the audio with the right tool.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
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Service Client IPeva
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Le 21/06/2011 à 04:29, Brian Campbell a écrit :

> I have run into a very strange issue with periodically losing Audio on Incomming calls
>  
> I am receiving incomming calls to FreeSwitch via a DID from an ITSP. Once FreeSwitch receives the call, I am adding a header to pass the UUID (which I use later to inject some custom values into CDRs) and then bridge to an IVR application running on Microsoft Speech Server 2007 which is on a separate box, using the following dial plan
>  
> <extension name="Inbound_FromSoTel">
>   <condition field="destination_number" expression="^1(\d+)$">
>     <action application="set" data="sip_h_X-FS_UUID=${uuid}" /> 
>     <action application="bridge" data="sofia/external/$1 at 173.14.17.213:5060;transport=tcp" /> 
>   </condition>
> </extension>
>  
> Some calls are performing properly, but in some of the calls (about 30%) the IVR application doesn't seem to hear anything the caller is saying
>  
> I have eliminated any Firewalls for testing, so I go straight from the ITSP to FreeSwitch and then to Speech Server. Thee is no NAT in play.
>  
> There doesn't seem to be any rhyme or reason as to when a call is going to work, or when it isn't. I have also seen exactly the same behavior in a completely different environment with a different ITSP and a completely different IVR application.
>  
> So I figure that the problem is somewhere between FreeSwitch and Speech Server, but I don't see anything wrong. 
>  
> Using WireShark I can see the SIP signalling creating/tearing down the call and I see RTP traffic flowing both ways between FreeSwitch and Speech Server
>  
> The version of FreeSwitch that I am running is 'FreeSWITCH Version 1.0.head (git-38f06a3 2011-05-19 15-39-43 -0500)'
>  
> I have pasted 2 FreeSwitch traces into pastebin. The first one is of a working call, the second one is of a call where the IVR application doesn't seem to hear anything the caller is saying
>  
> Can anyone advise on what the issue might be. I am completely out of ideas.
>  
> Thanks
>  
> Brian Campbell
> 
>  
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