Sounds like you are having a codec issue with 2 of your providers. Make sure you find out what codecs are supported and that your config is set up accordingly.
On Sun, 06 Mar 2005 00:14:05 +0000, w fm3 <[EMAIL PROTECTED]> wrote: > Hi > > Hope someone can help :) > > I am testing 4 PSTN termination providers. 3 SIP and 1 IAX > > IAX and 1 of the SIP providers work fine. > > Now the wierdness: > > 2 SIP providers I can only get oubound calls to ring at the destination and > then nothing more. 1 gets as far as SIP code 183 (and ringing on the src > handset ...yay) the other doesn't get past 100. > > Added to this inbound calls (PSTN->provider->asterisk->handset) work fine > 100% of the time. > > I have tried alot of config options from the wiki and lists but can't seem > to get any further. AFAIK from sip debug and the console it looks like > that the call is placed and then no further communication. Looks like they > might be using SER / CISCO GW at the VOIP Provider end. > Don't think it a open UDP port type thing. > > Cheers > > Walt > > PS Newbie > > _________________________________________________________________ > Express yourself instantly with MSN Messenger! Download today it's FREE! > http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@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 > _______________________________________________ Asterisk-Users mailing list Asterisk-Users@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