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/Username Host Mask Port Status wayne 165.165.164.87 (D) 255.255.255.255 4569 Unmonitored and on the other side iax2 show users shows Username Secret Authen Def.Context A/C Codec Pref wayne password 000000000000001 default No Host When trying to call from this side to that side I get the following -- Executing Dial("SIP/301-2d50", "IAX2/wayne:password@homebase.hidden.com/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 : waynetg@telkomsa.net
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: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] 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/Username Host Mask Port Status wayne 165.165.164.87 (D) 255.255.255.255 4569 Unmonitored and on the other side iax2 show users shows Username Secret Authen Def.Context A/C Codec Pref wayne password 000000000000001 default No Host When trying to call from this side to that side I get the following -- Executing Dial("SIP/301-2d50", "IAX2/wayne:password@homebase.hidden.com/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 : waynetg@telkomsa.net _______________________________________________ --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
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:password@homebase.hidden.com/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-5there'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." dinesh@alphaque.com (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 | +=========================================================================+