Try using canreinvite=yes in all three contexts. Would that screw-up ATA, I do not know, cause I have no Cisco's ?
SW Message: 5 Date: Mon, 05 Jan 2004 02:29:49 -0500 From: SamW <[EMAIL PROTECTED]> To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Codec Negotiation Does not seem to work as expected ?? Help Please !! Reply-To: [EMAIL PROTECTED] Hello, I have been trying to get my coders to work without a conversion. I have read all the available asterisk documentation and support groups without any luck. Here is my issue. (Please feel free to ask questions if you do not understand what I am talking about.) I am using Cisco ATA-186 set to g729 codec. (But it will switch to g711 if sip-server request g711) I have 2 SIP-services to which I have to deliver the call in 2 coder formats. Lets call 2 sip-providers, SIP-A and SIP-B. SIP-A accept g729 and g711, SIP-B only accept g711. I do not have any g729 licence, but I believe the * should negotiate to have the correct passthrough coders as ATA is capable of both coders. (I think even if you have the licenses, * should try avoid codec-conversions when ever it can) Here is my settings in sip.conf. I will only list the required codec related lines, for easy understanding, [general] disallow=all allow=g729 allow=ulaw allow=alaw register => [EMAIL PROTECTED] register => [EMAIL PROTECTED] [sip-a] .... disallow=all allow=ulaw [sip-b] ... disallow=all allow=g729 [ATA] ..... canreinvite=no _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
