[Asterisk-Users] realtime, iax, trunk
Hi All, Is there some way to verify that a channel is using iax trunking? The reason i ask is that i have a scenario where 1 Asterisk system is communicating with another over IAX. System A is using static configuration from the standard files, System B is using realtime with MySQL config for IAX. The table for iax as per voip-info (iax_buddies) does not contain a field for trunk, so i decided to add the field on the basis that you can add / remove fields to the table as per the static config file parameters, but i am not sure if this is valid or not - can anyone confirm that using trunk from iax realtime is valid? I would like to find some way to verify that trunking is happening on a channel so that i can confirm the above does work, unless someone can inform me otherwise? I have tried IAX debug, but could not find anything obvious in the logs. Cheers, Ben ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] realtime, iax, trunk
Ben Dinnerville wrote: Hi All, Is there some way to verify that a channel is using iax trunking? The reason i ask is that i have a scenario where 1 Asterisk system is communicating with another over IAX. System A is using static configuration from the standard files, System B is using realtime with MySQL config for IAX. The table for iax as per voip-info (iax_buddies) does not contain a field for trunk, so i decided to add the field on the basis that you can add / remove fields to the table as per the static config file parameters, but i am not sure if this is valid or not - can anyone confirm that using trunk from iax realtime is valid? I would like to find some way to verify that trunking is happening on a channel so that i can confirm the above does work, unless someone can inform me otherwise? I have tried IAX debug, but could not find anything obvious in the logs. on cli, try iax2 trunk debug, it will tell you how many peers it is trunking with ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime IAX
On Saturday 19 November 2005 01:05, Carlos Chavez wrote: 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/[EMAIL PROTECTED]/${EXTEN:3}) snip port: 0 Try port:4569 or whatever port you're using because by default it try port 0 and than you have congestion as a result. benchev ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime IAX
I am using the following on my server: name: voipjet username: voipjet type: friend secret: NULL md5secret: (md5password) auth: md5 accountcode: VoipJet context: casa defaultip: NULL host: 64.34.45.100 qualify: no disallow: all allow: gsm;alaw;ulaw ipaddr: NULL port: NULL regseconds: NULL I have to use the following Dial command Dial(IAX2/[EMAIL PROTECTED]/${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 = _XX.,5,Dial(${DIALSTRING}) -- Are Casillahttp://astartelecom.com - Independent VOIP Telecoms Broker. Asterisk Consultantshttp://astbill.com - Open Source Billing, Routing and Management software for Asterisk and VOIP AstBill DEMO: http://demo.astbill.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Realtime IAX
I've been having a problem dialing IAX extensions since I implemented Realtime for IAX Extensions. The problem is that I cannot seem to dial in a simplified manner an extension like: _9001.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:3}) I get the following on the console: -- Executing Dial(SIP/2001-ca0d, IAX2/[EMAIL PROTECTED]/19566680301) in new stack -- Called [EMAIL PROTECTED]/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:[EMAIL PROTECTED]/${EXTEN:3}) This obviously does not use the information in iax_buddies to make the call. Any ideas? -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 ___ --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] Realtime IAX
On 9/2/05, Chris A. Icide [EMAIL PROTECTED] wrote: Dana Olson wrote: Chris, Thanks for the reply. I checked those settings, and they were commented out, so I uncommented them. I assumed you meant rtnoupdate=yes, so that's what I put, but that didn't work. I tried rtnoupdate=no, and that didn't work either. I do have a register statement in my iax.conf, and that works - I can get my inbound calls no problem. Dana Actually, the current CVS Head usage is rtupdate=yes|no, it was changed from rtnoupdate=yes|no not too long ago. If you are using 1.2 I'm not sure which is correct. I went through this battle of getting this to work the beginning of this week, and the four settings I listed in my last post made all the difference. -Chris Just to follow up with this thread, kpflemming provided the solution that I overlooked - the port column in the iax table was set to 0 instead of 4569. I didn't think to change it because the wiki said that the port, ipaddr, etc were all optional. For IAX peers, the port is not optional. I added a note to the wiki stating so as well. -- Dana ___ --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] Realtime IAX
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:[EMAIL PROTECTED] 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 Make sure you have the following setting in your iax.conf file. rtcachefriends=yes rtupdate=yes rtautoclear=no rtignoreexpire=yes Also, you will still need your register = statement if you needed it before you started using realtime -Chris ___ --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] Realtime IAX
On 9/2/05, Chris A. Icide [EMAIL PROTECTED] wrote: 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:[EMAIL PROTECTED] 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? -- DanaMake sure you have the following setting in your iax.conf file.rtcachefriends=yesrtupdate=yesrtautoclear=nortignoreexpire=yesAlso, you will still need your register = statement if you needed it before you started using realtime-Chris Chris, Thanks for the reply. I checked those settings, and they were commented out, so I uncommented them. I assumed you meant rtnoupdate=yes, so that's what I put, but that didn't work. I tried rtnoupdate=no, and that didn't work either. I do have a register statement in my iax.conf, and that works - I can get my inbound calls no problem. Dana ___ --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] Realtime IAX
Dana Olson wrote: Chris, Thanks for the reply. I checked those settings, and they were commented out, so I uncommented them. I assumed you meant rtnoupdate=yes, so that's what I put, but that didn't work. I tried rtnoupdate=no, and that didn't work either. I do have a register statement in my iax.conf, and that works - I can get my inbound calls no problem. Dana Actually, the current CVS Head usage is rtupdate=yes|no, it was changed from rtnoupdate=yes|no not too long ago. If you are using 1.2 I'm not sure which is correct. I went through this battle of getting this to work the beginning of this week, and the four settings I listed in my last post made all the difference. -Chris ___ --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] Realtime IAX
On 8/5/05, Carlos Chavez [EMAIL PROTECTED] wrote: I am using Asterisk CVS from last week and have been using Realtime SIPfor a couple weeks now without any problems.Yesterday I decided to turn onRealtime 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/[EMAIL PROTECTED]/19566680301)in new stack-- SIP Seeding peer from astdb: '2001' at [EMAIL PROTECTED]:5060 for 3600-- Called [EMAIL PROTECTED]/19566680301Aug5 10:25:50 NOTICE[29140]: chan_iax2.c:2736 auto_congest: Auto-congestingcall 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/[EMAIL PROTECTED]/19566680301) in newstack-- Called [EMAIL PROTECTED]/19566680301Aug5 10:25:54 NOTICE[29140]: chan_iax2.c:2736 auto_congest: Auto-congestingcall 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/[EMAIL PROTECTED]/19566680301) in newstack-- Called [EMAIL PROTECTED]/19566680301-- Seeding 'pbxserver' at 66.135.38.93:4569 for 60Aug5 10:25:58 NOTICE[29140]: chan_iax2.c:2736 auto_congest: Auto-congestingcall 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 serverconnected with IAX2 that does work.To that server I can dial any extensionwithout problems. I used http://www.voip-info.org/tiki-index.php?page=Asterisk%20RealTime%20IAX toconfigure my * server.Any ideas?All three providers were working before Ichanged to Realtime IAX and I made sure to put all the necessary information into the Database.--Carlos ChavezDirector de TecnologíaTelecomunicaciones Abiertas de México 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:[EMAIL PROTECTED] 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 ___ --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
[Asterisk-Users] Realtime IAX
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/[EMAIL PROTECTED]/19566680301) in new stack -- SIP Seeding peer from astdb: '2001' at [EMAIL PROTECTED]:5060 for 3600 -- Called [EMAIL PROTECTED]/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/[EMAIL PROTECTED]/19566680301) in new stack -- Called [EMAIL PROTECTED]/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/[EMAIL PROTECTED]/19566680301) in new stack -- Called [EMAIL PROTECTED]/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 Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 ___ 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
[Asterisk-Users] Realtime IAX/SIP with 2 asterisk servers but 1 central iax/sipfriends Database
Hello I was wandering If I let 2 asterisk boxes (let's name them ast01 and ast02) connect to one SQL realtime iaxfriends/sipfriends database What happens if I register my client to ast01, The ast01 box will update the client's record in the iaxfriends database (ipaddr/port/regseconds) Let's say there is an incoming call then for this client but this call arrives on ast02 (the box where the client is NOT registerd to at the moment) .. ast02 will 'know' then (with a DB lookup in the same table) where (which ipaddr/port) to route the call to am I right? will that work? And... Will that work too if the client is behind a NAT? ___ 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