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Call barge In *81 and Teach Mode *82 - don't work


voipguy
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Hi,

 

We are using PBXNSIP version 4.0.1.3433 centos.

 

The Call barge In *81 and Teach Mode *82 are not working. I can do *81 or *82 and the extension but the audio is all static and you can't make out what they are saying. I can do Call Listen In *83 and I can hear the two parties talking perfectly clear.

 

Is this a bug in version 4 or a setting I have wrong?

 

We are using SNOM 370 phones. We are also doing call recording in the Sales Queue where the two parties are talking.

 

Thanks

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What codecs are you using? With G711 we did not see any issues here.

 

We are using G729.

 

Customer calls into the sales queue - agent takes the call - I can do *83 and listen in perfectly but when I try *82 or *81 I can hardley hear the 2 people talking - it's all distorted. I have all the right permissions in place. I even turned off call recording thinking that might be causing a problem but same results.

 

How are you doing your testing? Meaning my phones are remote - not on the local lan - would that make a difference for your testing method? I'm also using SNOM 370's - would the calls be encrypted - would that make a difference? Our PBXNSIP is located in a data center.

 

Thanks

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Mystery solved - why it was working for you and not me = a bug.

 

I changed my codec to G711 and now I can do the *82 and *81 and it works - this means the system has a bug when using codec G729.

 

Can you confirm this?

 

Thanks.

 

For anyone else reading this - PBXNSIP confirmed this is a bug and will be fixed in future release.

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can you post how many active calls on your system when you had the issues? and what was the CPU performance level?

 

 

1 caller in Sales Queue talking to 1 agent and then I was trying to do the *81 or *82 - that's all the people at the time on the system. I never looked at the CPU level at that time but it's never high - like less then 5%.

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  • 2 weeks later...
Is there a fix for this yet? I'm using ver 4.0.1.3446 and this bug still exists. I'm using the G729 codec.

 

Right now we don't do packet ordering for live recording (no jitter buffer). Packets are being played back as they arrive. For G.711 and other simple codecs that is no big problem; but it becomes very obvious with G.729. G.729 was designed for TDM networks and they did not anticipate that packets might ever get lost or arrive in the wrong sequence. Just like FAX.

 

Anyway, we need to fix this later. Right now we just want to get 4.0 out and such a change would bring in a lot of new risks.

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Right now we don't do packet ordering for live recording (no jitter buffer). Packets are being played back as they arrive. For G.711 and other simple codecs that is no big problem; but it becomes very obvious with G.729. G.729 was designed for TDM networks and they did not anticipate that packets might ever get lost or arrive in the wrong sequence. Just like FAX.

 

Anyway, we need to fix this later. Right now we just want to get 4.0 out and such a change would bring in a lot of new risks.

 

Hi pbxnsip,

 

Did this ever work in ver 3.4 or earlier? If not then PBXNSIP needs to update their web site sales info/documents letting potential customers know that *81 and *82 are currently only supported if useing codec G711 before they invest thousands of dollars in this software like I did.

 

This is going to be a major problem for us until version 4 supports *81 and *82. We use the hosted version and there's no way we will sign up customers for a call centre and use codec G711 - too much bandwidth.

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Did this ever work in ver 3.4 or earlier? If not then PBXNSIP needs to update their web site sales info/documents letting potential customers know that *81 and *82 are currently only supported if useing codec G711 before they invest thousands of dollars in this software like I did.

 

At least someone barged into my call yesterday. But I believe this was on G.711. G.729 works if the jitter is not too much. Unfortunately, in hosted mode that assumption is not always true... We need to add a jitter buffer for these features; something we don't want to do in 4 as it would screw a lot of the media subsystem up and that is the last thing we want right now!

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