[Asterisk-Users] realtime, iax, trunk

2006-02-20 Thread Ben Dinnerville

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

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Re: [Asterisk-Users] realtime, iax, trunk

2006-02-20 Thread yusuf

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
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Re: [Asterisk-Users] Realtime IAX

2005-11-19 Thread bbench
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
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Re: [Asterisk-Users] Realtime IAX

2005-11-19 Thread Are
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
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[Asterisk-Users] Realtime IAX

2005-11-18 Thread Carlos Chavez
 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

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Re: [Asterisk-Users] Realtime IAX

2005-09-14 Thread Dana Olson
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
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Re: [Asterisk-Users] Realtime IAX

2005-09-02 Thread Chris A. Icide

 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

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Re: [Asterisk-Users] Realtime IAX

2005-09-02 Thread Dana Olson
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
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Re: [Asterisk-Users] Realtime IAX

2005-09-02 Thread Chris A. Icide




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


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Re: [Asterisk-Users] Realtime IAX

2005-09-01 Thread Dana Olson
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
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[Asterisk-Users] Realtime IAX

2005-08-05 Thread Carlos Chavez
 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

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[Asterisk-Users] Realtime IAX/SIP with 2 asterisk servers but 1 central iax/sipfriends Database

2005-03-03 Thread niels
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?



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