Thanks for reply. Yes i am getting audio. It hangs-up automaticly after 10 secs, or the line goes down. Softphone has the line still open though.
I dont get this 404 anymore, it was just before the missing canreinvite= -----Original Message----- Cinoss, Are you getting audio during the call? Or are you just seeing the call setup? Is it really 10 seconds? Or just seems like it? I have seen the SIP 404 when the codec matches were incorrect. With Debug and Versobose, -dvvvvvgc, on in Asterisk look for messages about codec matching. Also turn on sip debugging. "*CLI> sip debug" Found description format CN Capabilities: us - 0x10e(GSM|ULAW|ALAW|G729A), peer - audio=0x105(G723|ULAW|G729A)/video=0x0(EMPTY), combined - 0x104(ULAW|G729A) Non-codec capabilities: us - 0x1(G723), peer - 0x3(G723|GSM), combined - 0x1(G723) Urgent handler If "combined" equals empty then you need to adjust the codecs. -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Cinoss Sent: 19 October 2004 13:04 To: asterisk-users at lists.digium.com Subject: [Asterisk-Users] SIP 404 - circuit busy when dialing out Well i have now sorted dialing out. Only needed to add fromdomain= to my [sipprovider]. Still got small problem with it. The call gets automaticly hang-up after 10secs. I tried both canreinvite=no and yes and my sip.conf but it doesn't seem to do any difference, well other than fail code. -- Cinoss cinosss at f-m.fm _______________________________________________ Asterisk-Users mailing list Asterisk-Users at lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Cinoss [EMAIL PROTECTED] _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
