 | [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? |
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 TrimlinePremium join:2004-10-24 Windermere, FL Reviews:
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| 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. |
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 | 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. |
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 AndrewZPremium join:2003-07-17 somewhere | reply 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) |
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 | 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. |
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 XCOMdigitalnUllPremium join:2002-06-10 Spring, TX Reviews:
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| 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 -- [nUll@dcypher ~]$ |
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 | 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. |
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 XCOMdigitalnUllPremium join:2002-06-10 Spring, TX Reviews:
·flowroute
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| 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. -- [nUll@dcypher ~]$ |
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