provided by Star2Billing S.L.

Hi,
I have this strange problem, my provider supports only G729, therefore I need to force only that codec, but in that way the file gsm are not working any idea?

you need to install G729 codec in asterisk so that you can have codec translation between G729<=>GSM files. you need to by a license for this codec through digium. below is their link

http://www.digium.com/en/products/voice/g729codec.php

thanks

Mohamed

I am also in the same situation, I have installed the codec and registered the license with digium. However, I keep getting a warning, i.e.

WARNING[6412]: channel.c:1913 ast_request: No channel type registered for 'G729'
NOTICE[6412]: app_dial_exec: Unable to create channel of type 'G729'
==Everyone is busy/congested at this time

At CLI>show g729, it says I have a license and the license file is in /var/lib/asterisk/license directory.
Under Trunk -> List Trunk, I have edit'd the Provider Tech = G729

Do I have to let Asterisk2Billing now this in another way?, because my calls are dropping.
Roger

I dont think you need to change anything in asterisk2billing; g729 is already included as one of the codec that is allowed when you create the customers/cards.

verify your g729 codec installation in asterisk and try to configure two friends without using asterisk2billing.

Hello,

There is very clearly written on instruction below PROVIDERTECH:-

Under Trunk -> List Trunk, PROVIDERTECH should be either [SIP, IAX2, ZAP or H323], you have to choose ONE based on your setup and termination provider.


About g729:-

give command on CLI> show g729

and you should get output:-

"0/0 encoders/decoders of 150 licensed channels are currently in use"

[150 = how many licensed you purchased]

You have to set the codec in sip.conf before sending out to termination provider.

There is very small time interval [only 9 milliseconds] between codec conversion of gsm --> g729.

Hope this helps.

Thanks Guys, I problem solved. I was placing g729 in Trunk -> List Trunk, PROVIDERTECH which could only recognise SIP, IAX2, ZAP or H323.
I changed it to SIP and it was cool!!

Hi, if i have a DID number that support G729 and terminating company that supports G729 and the only thing my asterisk does is answers the call and passes it on, do i need G729 licence?
In this case i'm not doing any kind of codec translation??

Hello,

If you are running asterisk2billing, then you need Codec Translator. Because a2b IVR is based on gsm codec.

So, it works as; calls originated with g729 then reach to IVR of gsm and then again to g729 [means g729 -> gsm -> g729]. As you see - need codec translator.

Hi again, is there any way i can use g729 sound files without the codec?

g729 call -> asterisk IVR g729 answers -> destination G729 provider

Hello Folks:

I am curious, which version of Asterisk2Billing you all are using? We are having terrible problems with the latest version. I've posted some messages on this regards.

Thanks
Razer.

Hi,

Please have a look from below link, I 've posted there [Post subject: Extending Topic - G729]

http://forum.asterisk2billing.org/viewt ... =1366#1366

Thank you for all responses. I bought 10 licences , installed them and now everything is ok. I can terminated my calls using g729 codec.

I'm using Centos 4.3 with asterisk 2.7 + latest a2billing

when i type show g729 in my CLI i take :

0/0 encoders/decoders of 1 licensed channels are currently in use

is this ok?

anyone can help me?????i need help pls

I am not sure that I understand the problem. I will re-read the messages later and try to find and answer.