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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090225/a33783fe/attachment.htm
On Wed, 2009-02-25 at 07:54 -0500, Mike wrote:> 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.This is a wild guess (and I don't currently have the time to check it out properly), but if my memory serves me the Polycom phones don't support the GSM codec. You might try ulaw instead. -- Jared Smith Digium, Inc. | Training Manager