Hi all, I've setup two * servers which are SIP interconnected ala osaka/toronto from the * book (before anyone sugggests using IAX instead, no, I NEED to have them SIP interconnected for verification/test purposes). Then I have a Zoiper connected to one of them via IAX (so that * will not reinvite (?)). As soon as I try to call (via Zoiper) an extension on the other * I get a "Failed to authenticate on INVITE" on the * to which the Zoiper is registered: -- Accepting AUTHENTICATED call from 192.168.10.113: << Zoiper IP > requested format = gsm, > requested prefs = (), > actual format = ulaw, > host prefs = (ulaw|alaw|gsm), > priority = mine -- Executing [010001 at users:1] Dial("IAX2/2200-12940", "SIP/010001 at 192.168.10.11") in new stack == Using SIP RTP CoS mark 5 -- Called 010001 at 192.168.10.11 << Other * [Oct 23 11:08:25] NOTICE[13576]: chan_sip.c:15031 handle_response_invite: Failed to authenticate on INVITE to '"2200" <sip:2200 at 192.168.10.77>;tag=as3e4fedb8' << 192.168.10.77 == * for Zoiper -- SIP/192.168.10.11-0a1716f8 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Auto fallthrough, channel 'IAX2/2200-12940' status is 'CONGESTION' -- Hungup 'IAX2/2200-12940' Why does * try to authenticate on sip:2200 at 192.168.10.77, it is IAX for crying out loud :) ? I've set canreinvite=no on the IAX phone (not sure this has any meaning in IAX at all) Not sure that this is root of the interconnection problem, since I then get SIP/192.168.10.11.. is circuit-busy... ? TIA /R
you need to post you SIP.conf and your Extensions.conf so someone can have a look at them and see if there is anything missing what are the contexts you are using with your peers? what is the dial plan triggered when calling your destination number? -- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308> Date: Sun, 25 Oct 2009 15:19:28 +0100 > From: robert.bielik at xponaut.se > To: asterisk-users at lists.digium.com > Subject: [asterisk-users] SIP interconnection problem > > Hi all, > > I've setup two * servers which are SIP interconnected ala osaka/toronto from the * book (before anyone sugggests using > IAX instead, no, I NEED to have them SIP interconnected for verification/test purposes). Then I have a > Zoiper connected to one of them via IAX (so that * will not reinvite (?)). As soon as I try to call (via Zoiper) an extension > on the other * I get a "Failed to authenticate on INVITE" on the * to which the Zoiper is registered: > > -- Accepting AUTHENTICATED call from 192.168.10.113: << Zoiper IP > > requested format = gsm, > > requested prefs = (), > > actual format = ulaw, > > host prefs = (ulaw|alaw|gsm), > > priority = mine > -- Executing [010001 at users:1] Dial("IAX2/2200-12940", "SIP/010001 at 192.168.10.11") in new stack > == Using SIP RTP CoS mark 5 > -- Called 010001 at 192.168.10.11 << Other * > [Oct 23 11:08:25] NOTICE[13576]: chan_sip.c:15031 handle_response_invite: Failed to authenticate on INVITE to '"2200" <sip:2200 at 192.168.10.77>;tag=as3e4fedb8' << 192.168.10.77 == * for Zoiper > -- SIP/192.168.10.11-0a1716f8 is circuit-busy > == Everyone is busy/congested at this time (1:0/1/0) > -- Auto fallthrough, channel 'IAX2/2200-12940' status is 'CONGESTION' > -- Hungup 'IAX2/2200-12940' > > Why does * try to authenticate on sip:2200 at 192.168.10.77, it is IAX for crying out loud :) ? I've set canreinvite=no on > the IAX phone (not sure this has any meaning in IAX at all) > > Not sure that this is root of the interconnection problem, since I then get SIP/192.168.10.11.. is circuit-busy... ? > > TIA > /R > > _______________________________________________ > -- Bandwidth and Colocation Provided by api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > lists.digium.com/mailman/listinfo/asterisk-users_________________________________________________________________ Windows 7: I wanted more reliable, now it's more reliable. Wow! microsoft.com/windows/windows-7/default-ga.aspx?h=myidea?ocid=PID24727::T:WLMTAGL:ON:WL:en-US:WWL_WIN_myidea:102009 -------------- next part -------------- An HTML attachment was scrubbed... URL: lists.digium.com/pipermail/asterisk-users/attachments/20091025/527b289f/attachment.htm
Since you are doing peer-to-peer, this should be harmless. -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Robert Bielik Sent: Tuesday, October 27, 2009 11:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP interconnection problem Lacking any response I tried to set "insecure=invite" on both sides. And lo and behold, the call gets through. Now, is this good or bad? /R _______________________________________________ -- Bandwidth and Colocation Provided by api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: lists.digium.com/mailman/listinfo/asterisk-users