SIP trunk cannot be transferred

lonetree

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Feb 7, 2009
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#1
Hi guys,

I have recently installed elastix 1.5.

I have installed a openvox A400P card with 2 FXO and 1 FXS module. Card and channel get detected successfully, thats one thing great for the beginning.

Then, I have 1 SIP account from my local VSP. It gets registered successfully too, and call in and out works fine too.

My outbound is as such. Caller from internal will first call out with SIP trunk and subsequently call out with ZAP when SIP is busy.

For ZAP trunk, I can do call transfer when caller call in and also transfer call to internal on ZAP channel. When it comes to SIP trunk, I can't transfer the call to anyone internally.

Have I missed out anything on the configuration?

I hope someone can help me.

Regards,

lonetree
 

dicko

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Oct 24, 2008
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#2
If g729 is involved I suggest you try g711 first to ensure it is not a licensing issue, then check the dtmfmode= line in any trunks and extensions and ATA's.
 

lonetree

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#3
Hi dicko,

can you explain further?

regards,
 

dicko

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#4
I don't know how to explain further, I thought I asked it already,

Do you use g729?
If you do
Do you have a license for g729?
if you do use g729 and don't have a license for g729 don't be surprised if non pass-through signalling doesn't work !. (read the license)

g711 requires no license and should transcode DTMF easily, so if if it's an option with your VSP try it and report back.
 

rafael

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#5
g729 is a codec that does a great job compressing the audio and it is very popular among the vsp because of this. The problem g729 has is that it is no free and you have to pay a license to use it. In the other hand g711 does not compress, but the quality is really good and you don't have to pay license for it. So what dicko is suggesting is to try g711 to discard the issue with g729.

You may want to read this:
http://www.voip-info.org/wiki-Asterisk+G.729+Licensing
 

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