Re: [Asterisk-Users] Can't create iax channel
On 11/10/05 15:02 Wayne Gemmell said the following: When trying to call from this side to that side I get the following -- Executing Dial(SIP/301-2d50, IAX2/wayne:[EMAIL PROTECTED]/204) in new stack Nov 10 08:37:21 WARNING[30785]: channel.c:455 ast_best_codec: Don't know any of 0xf800 formats Nov 10 08:37:21 WARNING[30785]: channel.c:455 ast_best_codec: Don't know any of 0xf800 formats Nov 10 08:37:21 WARNING[30785]: chan_iax2.c:7745 iax2_request: Unable to create translator path for unknown to ulaw on IAX2/wayne-5 there's your problem right there. what codecs are the SIP peer set to use ? apparently, asterisk cant translate between ulaw and the unknown codec. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
Re: [Asterisk-Users] Can't create iax channel
On 11/10/05 17:36 Wayne Gemmell said the following: On Thursday 10 November 2005 10:55, Jason Walker wrote: The statement of zaptel being required is strange...I use IX trunking exclusively for my servers. Two of them have no zaptel/Digium hardware and the trunk calls are fine. I don't know where I read it, apparently it is needed for timing or something, could be in the old handbook or hitchikers guide to asterisk as I havn't got far enough into the new handbook to comment. IAX trunking works even without digium cards as long as the ztdummy pseudo timer module is loaded. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
Re: [Asterisk-Users] Can't create iax channel
Jason Walker wrote: The statement of zaptel being required is strange...I use IX trunking exclusively for my servers. Two of them have no zaptel/Digium hardware and the trunk calls are fine. IAX trunks require a zaptel timing source, be it hardware or ztdummy. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
RE: [Asterisk-Users] Can't create iax channel
The statement of zaptel being required is strange...I use IX trunking exclusively for my servers. Two of them have no zaptel/Digium hardware and the trunk calls are fine. Based on your post, seems that you have an issue with codecs more than creating an IAX trunk. What version of Asterisk are you using? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wayne Gemmell Sent: Thursday, November 10, 2005 12:02 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Can't create iax channel Hi all Could somebody please give me an idea as to whats wrong here. I'm trying to connect 2 servers using IAX, I'm not trunking them because I read that you need zaptel hardware installed at both sides to do the trunking. Theregistration seems to have worked as the output of iax show peers on the side I'm working from is as follows Name/UsernameHost Mask Port Status wayne165.165.164.87 (D) 255.255.255.255 4569 Unmonitored and on the other side iax2 show users shows Username SecretAuthen Def.Context A/C Codec Pref waynepassword 001 default No Host When trying to call from this side to that side I get the following -- Executing Dial(SIP/301-2d50, IAX2/wayne:[EMAIL PROTECTED]/204) in new stack Nov 10 08:37:21 WARNING[30785]: channel.c:455 ast_best_codec: Don't know any of 0xf800 formats Nov 10 08:37:21 WARNING[30785]: channel.c:455 ast_best_codec: Don't know any of 0xf800 formats Nov 10 08:37:21 WARNING[30785]: chan_iax2.c:7745 iax2_request: Unable to create translator path for unknown to ulaw on IAX2/wayne-5 -- Hungup 'IAX2/wayne-5' Nov 10 08:37:21 NOTICE[30785]: app_dial.c:1091 dial_exec_full: Unable to create channel of type 'IAX2' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Executing Congestion(SIP/301-2d50, ) in new stack == Spawn extension (from-internal, 204, 2) exited non-zero on 'SIP/301-2d50' Any ideas? -- Regards Wayne Gemmell Tel Fax: (011) 894-4081 Cell : 072 836 4325 Email : [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
Re: [Asterisk-Users] Can't create iax channel
On Thursday 10 November 2005 10:55, Jason Walker wrote: The statement of zaptel being required is strange...I use IX trunking exclusively for my servers. Two of them have no zaptel/Digium hardware and the trunk calls are fine. I don't know where I read it, apparently it is needed for timing or something, could be in the old handbook or hitchikers guide to asterisk as I havn't got far enough into the new handbook to comment. Based on your post, seems that you have an issue with codecs more than creating an IAX trunk. Thanks, yes I was disallowing all codecs, :( -- Cheers Wayne ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
