Hello, Yes, I also believe that this is some sort of codec issue. Here is my sip.conf file: [201]<?xml:namespace prefix = o ns = "urn:schemas-microsoft-com:office:office" />
type=friend ;secret=201 record_out=Adhoc record_in=Adhoc qualify=no port=5060 nat=no host=dynamic dtmfmode=rfc2833 dial=SIP/201 context=from-internal canreinvite=no callerid=device <201> [202] type=friend ;secret=202 record_out=Adhoc record_in=Adhoc qualify=no port=5060 nat=no host=dynamic dtmfmode=rfc2833 dial=SIP/202 context=from-internal canreinvite=no callerid=device <202> Note: The "secret" is commented out so that there is no authentication when registering with the Jain-Sip phones. Thanks, -----Original Message----- From: Gerald A [mailto:[EMAIL PROTECTED] Sent: Tuesday, September 11, 2007 5:12 PM To: Kutman [EMAIL PROTECTED](Mat) DAEPM(R&CS)@Ottawa-Hull Subject: Re: [asterisk-users] Chan_sip Entry Hi again, On 9/11/07, [EMAIL PROTECTED] < [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]> > wrote: I am trying to get to Jain Sip softphones to call one another via an Asterisk server. When I call from phone 1 to phone 2 there is audio transmission both ways, but when I call from phone 2 to phone 1 I don't get audio transmission and reception both ways. When I look at the asterisk log file it has an entry which says: "Oooh, format changed to 2". Usually this is a codec selection problem. Are both Jain's the same version? Maybe posting your sip.conf for the phones might help. Thanks, Gerald.
_______________________________________________ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
