On 11/28/05, Pablo Chacón <[EMAIL PROTECTED]> wrote: > Hi I'm trying to connect Avaya S8700 and Asterisk through H323 trunk > (using channel oh323). > I can make calls from S8700 H323 extension to Asterisk SIP phone using > G711a codec but when I try to make a call from SIP phone to S8700 > extension I listen one ringing tone and the call is dropped. > Can anybody help me??? >
I've had greater success increasing the number of frames in an RTP packet when dealing with the med pro resources on the S8700. Also, make sure you're sending the call to the IP that is bound to the CLAN board that also has the signaling group you're trying to call into bound to it. With the connection refused here it seems like you might be trying to send the call to the IP of the med pro board instead of a CLAN board. BJ -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
