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david3
join:2000-03-21

david3

Member

[Asterisk] How to enable G.722 / Asterisk

Right now I've got a small home asterisk setup with a Polycom 321 IP phone and a pair of DECT phones hooked up an Obi110 and Obi100. The only codecs I have enabled are G.711u and G.711A (ulaw/alaw).

I'm going to add a wideband USB headset and softphone and I'm wondering if there's a way to enable G.722 in asterisk in such a way that normal G.711 calls won't get transcoded while still allowing G.722 for the softphone?

Trimline
Premium Member
join:2004-10-24
Windermere, FL

Trimline

Premium Member

said by david3:

I'm going to add a wideband USB headset and softphone and I'm wondering if there's a way to enable G.722 in asterisk in such a way that normal G.711 calls won't get transcoded while still allowing G.722 for the softphone?

Yes, there is a way to accomplish this.

First, make sure you have the G722 codec enabled via Settings, Sip Settings. Should be just a check mark. You can then prioritize the codec by moving up/down. I have mine set to Ulaw then G722. Save these settings.

For your asterisk extension for the softphone, got to the settings and under the settings disallow, set this to "all". Then directly underneath that, allow, place "g722". Save this extension and reload.

You should be good to go.
david3
join:2000-03-21

david3

Member

Thanks. I'm using the CLI and editing the config files manually.

I guess I'll try some different things and see if I can get it to work.

AndrewZ
Premium Member
join:2003-07-17
somewhere

AndrewZ to david3

Premium Member

to david3
said by david3:

I'm wondering if there's a way to enable G.722 in asterisk

Add allow=g722 to the global sip configuration (sip.conf) , extension and trunk (if any) settings (sip.conf and/or users.conf)
david3
join:2000-03-21

david3

Member

said by AndrewZ:

Add allow=g722 to the global sip configuration (sip.conf) , extension and trunk (if any) settings (sip.conf and/or users.conf)

I'm wondering, if I do that, if regular g711 calls to/from the headset will end up getting transcoded to g722 all the time.

XCOM
digitalnUll
Premium Member
join:2002-06-10
Spring, TX
(Software) pfSense
MikroTik CRS125-24G-1S-RM

XCOM

Premium Member

said by david3:

said by AndrewZ:

Add allow=g722 to the global sip configuration (sip.conf) , extension and trunk (if any) settings (sip.conf and/or users.conf)

I'm wondering, if I do that, if regular g711 calls to/from the headset will end up getting transcoded to g722 all the time.

from your peer to your asterisk server it will allways be g722. From your asterisk server to your ITSP/PSTN it will be g711 transcoded.

Is a simple setting on your general and peer in your sip.conf.

disallow=all
allow=g722
david3
join:2000-03-21

david3

Member

said by XCOM:

from your peer to your asterisk server it will allways be g722. From your asterisk server to your ITSP/PSTN it will be g711 transcoded.

That's what I'm trying to avoid, since 99.99% of the calls will be in g711, anyway.

XCOM
digitalnUll
Premium Member
join:2002-06-10
Spring, TX
(Software) pfSense
MikroTik CRS125-24G-1S-RM

XCOM

Premium Member

said by david3:

said by XCOM:

from your peer to your asterisk server it will allways be g722. From your asterisk server to your ITSP/PSTN it will be g711 transcoded.

That's what I'm trying to avoid, since 99.99% of the calls will be in g711, anyway.

g722 is low over head. I have it enable on all my peers and I have no issues with cpu. BUT if you have nothing in your house that uses g722 or anything that you are going to call than is useless.