Audio switches to One way

[Andyd358]

We have a client on our multi tennant PBX whis has an issue but only when calling one number. Ffter around 50 seconds or so the audio drops to incoming only. The called party can hear the client but the client cant hear the called party. This only happens ith this one Number and no other number they call. I raised with with Gamma our sip provider and they came back with this:

Hi Team,

Yes. There is an issue with the invite from the A party i.e., your client. The media attribute is sent as inactive and 102 events rather than 101.

Why is the media inactive on the initial call invite? As this then prompts a re-invite from the destination with a send only attribute. While the B party is the one initiating one-way audio, they only do this because your client initially proposes no audio.

If this is change locally this will resolve the issue.

What I dont understand is why is this only happening with just the won number. Aparantly if the Called party calls in to the client there isnt an issue. Also where would i chnage this setting they mention.

Regards

Andrew

 

[Adrian Fretwell]

Not sure why the media attribute would be inactive, unless this is being copied over from the other leg, i.e from the phones invite. It is quite normal not to see this type of media attribute at all in an initial invite (inactive, sendonly, sendrecv etc.).

Not sure why they are mentioning events, these are for RFC2833 DTMF transmission and should have no bearing on what happens to the media stream. You should see the DTMF settings in your external SIP profile, I normally have dtmf-type set to rfc2833 and enabled, and rfc2833-pt set to 101 and enabled.

Try to look at a couple of packet captures for the call, one taken between the calling phone and your PBX and one taken between your PBX and your provider.

 

[Andyd358]

Not sure why the media attribute would be inactive, unless this is being copied over from the other leg, i.e from the phones invite. It is quite normal not to see this type of media attribute at all in an initial invite (inactive, sendonly, sendrecv etc.).

Not sure why they are mentioning events, these are for RFC2833 DTMF transmission and should have no bearing on what happens to the media stream. You should see the DTMF settings in your external SIP profile, I normally have dtmf-type set to rfc2833 and enabled, and rfc2833-pt set to 101 and enabled.

Try to look at a couple of packet captures for the call, one taken between the calling phone and your PBX and one taken between your PBX and your provider.

Hi Adrian,

Just checked the profile and its set as you have mentioned. What I find strange is that its only effecting this on phone number. I intially thought it may have been a firewall issue at the clients place. PBX is Hosted on a VPS and no one else has ever reported this issue before.