[asterisk-users] SIP_CODEC variable

[Mike]

Hi,

 

I am using Aserisk 1.4.23.1 and trying to use SIP_CODEC to define the codec
being used. I have exclusively Polycom phones for this test, and basically I
want all communications to use g729 (preferred codec), except for pagine 20
phones (which busts my g729 license count). In that case I want to use gsm.

 

I have therefore specified Set(SIP_CODEC=gsm) I my dialplan before the
appropriate Page command call. But I get this in th CLI:

 

NOTICE[4764]: chan_sip.c:3706 try_suggested_sip_codec: Ignoring ${SIP_CODEC}
variable because it is not shared by both ends.

 

All my registered phones are using g729 and gsm in the sip definitions. 

 

What could it be?

 

Mike

 

 

 

 

 

 

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