Hi All,
We would like to install and use G729 on the SIP trunk only, while mantaine 711alaw on the LAN between extensions.
Do we have to install G729 and pay the license for it? or there is another way "Passthrough" where we can use it without having to install.
One more thing, we heard quite good things about G722, how can we use it? do we have to install it first as well?
Thanks for your help.
Asterisk will do g729 passthrough with no additional software. If you want to terminate g729 in Asterisk you have to add the codec which is licensed $10/channel from Digium
How can we configure that.
last time we tried we have done the following:
1- Set G729 as the only allow= statement under the extension configs
2- Set G729 as the first preference on the phone GUI
That has caused all communications between these phones to use G729 even on the LAN, which is fine but the problem was Voicemail. I think all these VM prompts are not G729 compatible so we were not able to hear any VM prompts etc....
What we ideally want is to use standard alwa on LAN and G729 in passthrough mode over SIP trunk only. That also applies for G722
your help is much appreciated.
You can install the G.729 codec per the Asterisk v1.2 instructions at Digium. Once you have the codec installed, it becomes an option in the trixbox Pro VoIP configuration page codec selection.
Thanks,
I understand that. But what we are interested in is configuring G729 or G722 in passthrough mode where we don't need to purchase any licenses, and why we have to since all phones do come with G729 license already.
I guess we will have to test different settings untill we get it to work without installing any CODECs.
What I meant to say is that G.729 support is not available within the GUI without installing licenses. It may be possible to do what you are trying to accomplish, but I don't know how to do it offhand.
Thanks,
Hi yesmat,
If you configure Asterisk for g729 passthrough with no licenses, Asterisk will not be able to process the g729 audio stream. More specifically, Asterisk wouldn't be able to listen for DTMF from the g729 endpoints. Voicemail, IVRs, inband transfers or feature codes, etc. would not work for these calls because Asterisk wouldn't be able to decode and process them, only pass them through to the other call leg.
You might consider purchasing enough g729 channels to cover the highest number of concurrent calls you expect to have to oicemail, if that's your primary application. This is usually a fraction of the total number of endpoints. You could have 100 g729-capable extensions that passthrough Asterisk, but if no more than five call voicemail simultaneously, then you'd only need five g729 channels. Just be sure to configure SIP reinvites so calls release g729 channels they don't need.
Also, if you find that you need more in the future, Asterisk can use multiple g729 keys to add channels to the pool. This would let you purchase only as many as you actually need.
Hope this helps,
rm
--
Rod Montgomery
Director of Services, Digium, Inc.
Rod Montgomery
Product Manager, Digium, Inc.
Hi Rod,
Thanks for your detailed explanation. When you say "Configure SIP reinvites so calls release g729 channels.." is the way to do that by configuring canreinvite=yes for each extension and also for the SIP trunk to provider? or there is more to configure?
Is it good practice to configure canreinvite=yes anyway?
Thanks
Member Since:
2006-06-08