I am using Asterisk CVS from last week and have been using Realtime SIP for a couple weeks now without any problems. Yesterday I decided to turn on Realtime IAX but I am having problems dialing to my long distance providers like Voicepulse, Sixtel or Nufone. I get the following: -- Executing Dial("SIP/2001-3761", "IAX2/password@voicepulse/19566680301") in new stack -- SIP Seeding peer from astdb: '2001' at 2001@192.168.2.23:5060 for 3600 -- Called password@voicepulse/19566680301 Aug 5 10:25:50 NOTICE[29140]: chan_iax2.c:2736 auto_congest: Auto-congesting call due to slow response -- IAX2/voicepulse-11 is circuit-busy -- Hungup 'IAX2/voicepulse-11' == Everyone is busy/congested at this time (1:0/1/0) -- Executing Dial("SIP/2001-3761", "IAX2/user@NuFone/19566680301") in new stack -- Called user@NuFone/19566680301 Aug 5 10:25:54 NOTICE[29140]: chan_iax2.c:2736 auto_congest: Auto-congesting call due to slow response -- IAX2/NuFone-2 is circuit-busy -- Hungup 'IAX2/NuFone-2' == Everyone is busy/congested at this time (1:0/1/0) -- Executing Dial("SIP/2001-3761", "IAX2/user@sixTel/19566680301") in new stack -- Called user@sixTel/19566680301 -- Seeding 'pbxserver' at 66.135.38.93:4569 for 60 Aug 5 10:25:58 NOTICE[29140]: chan_iax2.c:2736 auto_congest: Auto-congesting call due to slow response -- IAX2/sixTel-13 is circuit-busy -- Hungup 'IAX2/sixTel-13' == Everyone is busy/congested at this time (1:0/1/0) As you can see none of them go through. I have another Asterisk server connected with IAX2 that does work. To that server I can dial any extension without problems. I used http://www.voip-info.org/tiki-index.php?page=Asterisk%20RealTime%20IAX to configure my * server. Any ideas? All three providers were working before I changed to Realtime IAX and I made sure to put all the necessary information into the Database. -- Carlos Chavez Director de Tecnolog?a Telecomunicaciones Abiertas de M?xico S.A. de C.V. Tel: +52-55-91169161 Ext 2001
On 8/5/05, Carlos Chavez <cursor@telecomabmex.com> wrote:> > I am using Asterisk CVS from last week and have been using Realtime SIP > for a couple weeks now without any problems. Yesterday I decided to turn > on > Realtime IAX but I am having problems dialing to my long distance > providers > like Voicepulse, Sixtel or Nufone. I get the following: > > -- Executing Dial("SIP/2001-3761", "IAX2/password@voicepulse/19566680301") > in new stack > -- SIP Seeding peer from astdb: '2001' at 2001@192.168.2.23:5060 for 3600 > -- Called password@voicepulse/19566680301 > Aug 5 10:25:50 NOTICE[29140]: chan_iax2.c:2736 auto_congest: > Auto-congesting > call due to slow response > -- IAX2/voicepulse-11 is circuit-busy > -- Hungup 'IAX2/voicepulse-11' > == Everyone is busy/congested at this time (1:0/1/0) > -- Executing Dial("SIP/2001-3761", "IAX2/user@NuFone/19566680301") in new > stack > -- Called user@NuFone/19566680301 > Aug 5 10:25:54 NOTICE[29140]: chan_iax2.c:2736 auto_congest: > Auto-congesting > call due to slow response > -- IAX2/NuFone-2 is circuit-busy > -- Hungup 'IAX2/NuFone-2' > == Everyone is busy/congested at this time (1:0/1/0) > -- Executing Dial("SIP/2001-3761", "IAX2/user@sixTel/19566680301") in new > stack > -- Called user@sixTel/19566680301 > -- Seeding 'pbxserver' at 66.135.38.93:4569 <http://66.135.38.93:4569> for > 60 > Aug 5 10:25:58 NOTICE[29140]: chan_iax2.c:2736 auto_congest: > Auto-congesting > call due to slow response > -- IAX2/sixTel-13 is circuit-busy > -- Hungup 'IAX2/sixTel-13' > == Everyone is busy/congested at this time (1:0/1/0) > > As you can see none of them go through. I have another Asterisk server > connected with IAX2 that does work. To that server I can dial any > extension > without problems. > > I used > http://www.voip-info.org/tiki-index.php?page=Asterisk%20RealTime%20IAX to > configure my * server. Any ideas? All three providers were working before > I > changed to Realtime IAX and I made sure to put all the necessary > information > into the Database. > > -- > Carlos Chavez > Director de Tecnolog?a > Telecomunicaciones Abiertas de M?xico S.A. de C.V. > Tel: +52-55-91169161 Ext 2001 >I am having the exact same issues. I even tried to madk my IAX peer account in both the database, and in the iax.conf file (with different names, but same info) and the static one works, but not the database one. I am using 1.2.0-beta1. If I specify the user:password@host on the dialplan, it works, but this is bypassing the peer in the iaxpeers table in the database. I contacted my IAX provider, and he was not seeing the dial request come across or anything, so where that circuit-busy is coming from, I don't know... Did you ever get a resolution? Is this maybe a bug that should be opened on the Digium tracker? -- Dana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050901/30633066/attachment.htm
I've been having a problem dialing IAX extensions since I implemented Realtime for IAX Extensions. The problem is that I cannot seem to dial in a simplified manner an extension like: _9001.,1,Dial(IAX2/uid@voipjet/${EXTEN:3}) I get the following on the console: -- Executing Dial("SIP/2001-ca0d", "IAX2/uid@voipjet/19566680301") in new stack -- Called uid@voipjet/19566680301 Nov 18 16:58:47 NOTICE[2982]: chan_iax2.c:2818 auto_congest: Auto-congesting call due to slow response -- IAX2/voipjet-1 is circuit-busy -- Hungup 'IAX2/voipjet-1' I have the following information in my iax_buddies table: name: voipjet username: voipjet type: friend secret: md5secret: (md5password) dbsecret: NULL notransfer: NULL inkeys: auth: md5 accountcode: VoipJet amaflags: default callerid: context: casa defaultip: NULL host: 64.34.45.100 language: NULL mailbox: NULL deny: NULL permit: NULL qualify: no disallow: all allow: gsm;alaw;ulaw ipaddr: NULL port: 0 regseconds: 0 The only way I can use that service is to change my diaplan to the following: _9001.,Dial(IAX2/uid:password@64.34.45.100/${EXTEN:3}) This obviously does not use the information in iax_buddies to make the call. Any ideas? -- Carlos Chavez Director de Tecnolog?a Telecomunicaciones Abiertas de M?xico S.A. de C.V. Tel: +52-55-91169161 Ext 2001
I am using the following on my server: name: voipjet username: voipjet type: friend secret: NULL md5secret: (md5password) auth: md5 accountcode: VoipJet context: casa defaultip: NULL host: 64.34.45.100 <http://64.34.45.100> qualify: no disallow: all allow: gsm;alaw;ulaw ipaddr: NULL port: NULL regseconds: NULL I have to use the following Dial command Dial(IAX2/uid@voipjet/${EXTEN:3}) This takes the password and host from REALTIME but you still have to put the voipjet uid in extensions.conf That is the best we have been able to do. If anybody can do better I like to know. Actually in AstBill we are getting the complete Dial String from the MySQL database by returning a Variable from the AGI script. This make our dialing 100% flexible and make it simple to implement efficient LCR. exten => _XXXXXX.,5,Dial(${DIALSTRING}) -- Are Casilla http://astartelecom.com - Independent VOIP Telecoms Broker. Asterisk Consultants http://astbill.com - Open Source Billing, Routing and Management software for Asterisk and VOIP AstBill DEMO: http://demo.astbill.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051119/62872c11/attachment.htm