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@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. 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@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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