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/
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 <wfm3@hotmail.com> 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 >
>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.Thanks :) I don't think that is it though as I have tried with other codecs initially and inbound calls work fine regardless. My current setup - using G729 exclusively for everything - inbound calls work fine and calls to a test extension on 1 of the providers work. I have confirmed all providers are G729 capable. to the uneducated eye it is like I get initial SIP call progress notifications back from the provider and then nothing more is received. I know it could probably be 100 things on the way but has anyone experienced something like this? Especially if connecting to SER or cisco PSTN GW at the provider end. Cheers Walt _________________________________________________________________ Don't just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/