Re: [asterisk-users] IAX Trunk issue. (Dale Noll

[Mitchell Johnson]

Dale,

Sorry for taking so long to answer, I've been traveling.

Thanks so much for the suggestion, your solution worked perfectly.  I'm not 
sure why I didn't notice that the IAX trunk was working in the other direction.

Once again, thanks for your help.

Mitch
Date: Mon, 25 Jun 2012 05:44:37 -0500
From: Dale Noll 
Subject: Re: [asterisk-users] IAX Trunk issue.
To: Asterisk Users Mailing List - Non-Commercial Discussion

Message-ID: <4fe84115.60...@wi.rr.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

On 06/24/2012 07:53 PM, Mitchell Johnson wrote:
> I'm testing a few IAX trunk scenarios in a controlled lab.  From server2 
> extension 5000 (Server IP Address 172.16.200.212) I dial 6001 which goes 
> across the IAX trunk to server 1 (IP address 172.16.200.210).  Instead of 
> ringing the 6001 phone, it plays tt-weasels (the s extension).  When I dial 
> 6099 it also plays tt-weasels as it's supposed to, but it's not the 
> tt-weasels under its extension.  It also dials the s extension.
> 
> I only placed the s extension in the dial plan to verify that the traffic was 
> going across the IAX trunk and hitting the correct context.
> 
> Any help would be greatly appreciated.
> 
> Thanks Mitch
> 
> 
> 
> [phones]
> exten =>  _60XX,1,Dial(IAX2/trunk-1)
> exten =>  _X.,1,Dial(IAX2/trunk-1)
> exten =>  5000,1,Dial(SIP/${EXTEN})
> exten =>  5000,n,Hangup
> same =>  n,Hangup()
> exten =>  5099,1,Playback(tt-monkeys)
> exten =>  5099,n,HangUp
You are not telling asterisk-1 where you want the call to go, so it is going to 
's'.

Try adding the extension to the Dial() command on asterisk-2.  Change

Dial(IAX2/trunk-1)

to

Dial(IAX2/trunk-1/${EXTEN})


Note:  It appears that you are doing it correctly from asterisk-1 
towards asterisk-2

exten =>  _5XXX,1,Dial(${IAXTrunk}/${EXTEN})

Assuming, of course, that the variable IAXTrunk is properly set.


Dale

-- 
"The truth speaks for itself. I'm just the messenger."
 Lyta Alexander - Babylon 5






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Re: [asterisk-users] IAX Trunk issue.

[Dale Noll]

On 06/24/2012 07:53 PM, Mitchell Johnson wrote:

I'm testing a few IAX trunk scenarios in a controlled lab.  From server2 
extension 5000 (Server IP Address 172.16.200.212) I dial 6001 which goes across 
the IAX trunk to server 1 (IP address 172.16.200.210).  Instead of ringing the 
6001 phone, it plays tt-weasels (the s extension).  When I dial 6099 it also 
plays tt-weasels as it's supposed to, but it's not the tt-weasels under its 
extension.  It also dials the s extension.

I only placed the s extension in the dial plan to verify that the traffic was 
going across the IAX trunk and hitting the correct context.

Any help would be greatly appreciated.

Thanks Mitch



[phones]
exten =>  _60XX,1,Dial(IAX2/trunk-1)
exten =>  _X.,1,Dial(IAX2/trunk-1)
exten =>  5000,1,Dial(SIP/${EXTEN})
exten =>  5000,n,Hangup
same =>  n,Hangup()
exten =>  5099,1,Playback(tt-monkeys)
exten =>  5099,n,HangUp

You are not telling asterisk-1 where you want the call to go, so it is going to 
's'.

Try adding the extension to the Dial() command on asterisk-2.  Change

Dial(IAX2/trunk-1)

to

Dial(IAX2/trunk-1/${EXTEN})


Note:  It appears that you are doing it correctly from asterisk-1 
towards asterisk-2


exten =>  _5XXX,1,Dial(${IAXTrunk}/${EXTEN})

Assuming, of course, that the variable IAXTrunk is properly set.


Dale

--
"The truth speaks for itself. I'm just the messenger."
 Lyta Alexander - Babylon 5


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[asterisk-users] IAX Trunk issue.

[Mitchell Johnson]

I'm testing a few IAX trunk scenarios in a controlled lab.  From server2 
extension 5000 (Server IP Address 172.16.200.212) I dial 6001 which goes across 
the IAX trunk to server 1 (IP address 172.16.200.210).  Instead of ringing the 
6001 phone, it plays tt-weasels (the s extension).  When I dial 6099 it also 
plays tt-weasels as it's supposed to, but it's not the tt-weasels under its 
extension.  It also dials the s extension.

I only placed the s extension in the dial plan to verify that the traffic was 
going across the IAX trunk and hitting the correct context.

Any help would be greatly appreciated.

Thanks Mitch



Asterisk-1

IP Address 172.16.200.210

SIP.CONF

[6001]
type=friend
host=dynamic
context=internal_users
secret=xxx
nat=yes

[6002]
type=friend
host=dynamic
context=internal_users
secret=xxx
nat=yes

extensions.conf

[internal_users]
exten => 6000,1,Answer()
exten => 6000,2,Playback(hello-world)
exten => 6000,3,Hangup()
exten => 6001,1,Dial(SIP/6001)
exten => 6002,1,Dial(SIP/6002)
exten => 6099,1,Playback(tt-weasels)
exten => 6099,n,HangUp
exten => _5XXX,1,Dial(${IAXTrunk}/${EXTEN})
same => n,Hangup()
exten => s,1,Answer()
exten => s,n,Playback(tt-weasels)
exten => s,n,Hangup()

IAX.conf

[trunk-1]
type=friend
username=trunk-1
trunk=yes
requiretoken=no
secret=password
host=172.16.200.212
context=internal_users
auth=plaintext
disallow=all
;allow=ulaw
;allow=alaw
allow=gsm

Asterisk-2

IP Address 172.16.200.212

sip.conf

[5000]
type=friend
context=phones
host=dynamic
disallow=all
allow=ulaw
secret=xxx

extensions.conf

[phones]
exten => _60XX,1,Dial(IAX2/trunk-1)
exten => _X.,1,Dial(IAX2/trunk-1)
exten => 5000,1,Dial(SIP/${EXTEN})
exten => 5000,n,Hangup
same => n,Hangup()
exten => 5099,1,Playback(tt-monkeys)
exten => 5099,n,HangUp

iax.conf

[trunk-1]
type=friend
username=trunk-1
trunk=yes
requiretoken=no
secret=password
host=172.16.200.210
context=phones
auth=plaintext
disallow=all
;allow=ulaw
;allow=alaw
allow=gsm


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[asterisk-users] IAX trunk two Asterisk

[bakko]

Hello,

maybe is a stupid question but I'd like know if I'm doing some mistake on 
IAX trunk configuration.

I have two Asterisk (1.6.2.14) with this configuration:

A (iax.conf)

register => serverb:passw...@192.168.142.246

[servera]
type=friend
host=dynamic
trunk=yes
secret=password
context=phones
deny=0.0.0.0/0.0.0.0
permit=192.168.142.246/255.255.255.255
qualify=yes

B (iax.conf)

register => servera:passw...@192.168.159.4

[serverb]
type=friend
host=dynamic
trunk=yes
secret=password
context=phones
deny=0.0.0.0/0.0.0.0
permit=192.168.159.4/255.255.255.255
qualify=yes

The password it's the same on two server.

If i use two differents passwords (one for servera and one for serverb), the 
trunk don't work (Call rejected No authority found)

On the iax.conf general I have:

calltokenoptional=0.0.0.0/0.0.0.0

Am I doing wrong something?

Thank you for support

Best Regards.

- Bakko 


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[asterisk-users] Iax trunk quality

[Thalassoline - Service technique]

Hi,


I try to find a solution for this problem : 

[Jul 3 09:30:38] WARNING[3756]: chan_iax2.c:7312 socket_process:
Received trunked frame before first full voice frame 
[Jul 3 09:30:38] WARNING[3757]: chan_iax2.c:7312 socket_process:
Received trunked frame before first full voice frame 
[Jul 3 09:30:38] WARNING[3751]: chan_iax2.c:7312 socket_process:
Received trunked frame before first full voice frame 
[Jul 3 09:30:38] WARNING[3759]: chan_iax2.c:7312 socket_process:
Received trunked frame before first full voice frame 
[Jul 3 09:30:38] WARNING[3752]: chan_iax2.c:7312 socket_process:
Received trunked frame before first full voice frame 
...etc ... 

In the same time the user complain of the telephony quality. 

I have 14 ms of ping in my trunk. And i use Cisco with QOS on a
dedicaced SDSL. 

*CLI> iax2 show netstats 
 LOCAL -  REMOTE
 
Channel RTT Jit Del Lost % Drop OOO Kpkts Jit Del Lost % Drop OOO Kpkts

IAX2/blabla-16385 15 -1 0 -1 -1 0 -1 0 0 40 0 0 0 0 0 
IAX2/blabla-16387 12 -1 0 -1 -1 0 -1 0 0 40 0 0 0 0 0 
2 active IAX channels 

## My configuration ( I use a sangoma key for the timing source on my
slave.) : 

server_master:/etc/asterisk# asterisk -r 
Asterisk 1.4.24.1, Copyright (C) 1999 - 2008 Digium, Inc. and others. 
server_slave:/etc/asterisk# asterisk -r 
Asterisk 1.4.24.1, Copyright (C) 1999 - 2008 Digium, Inc. and others. 

server_master:/var/lib/asterisk/agi-bin/inc# dahdi_test 
Opened pseudo dahdi interface, measuring accuracy... 
99.998535% 99.995308% 99.996284% 99.997269% 99.998924% 99.996872%
99.998634% 
99.993553% 99.998245% 99.995216% 99.998039% 99.997856% 99.992966%
99.999321% 
--- Results after 14 passes --- 
Best: 99.999 -- Worst: 99.993 -- Average: 99.996930, Difference:
99.997334 

; iax.conf server_master 
; 
[general] 
bindport= 
bindaddr=222.222.222.222 
delayreject=yes 
language=fr 
allow=alaw 
maxjitterbuffer=800 
trunktimestamps=yes 
tos=ef 

[jourdain] 
type=friend 
host=111.111.111.111 
port= 
context=iax-jourdain 
trunk=yes 
disallow=all 
allow=alaw 
jitterbuffer=no 
forcejitterbuffer=no 
transfer=no 

server_salve:~# dahdi_test 
Opened pseudo dahdi interface, measuring accuracy... 
99.989647% 99.985260% 99.989449% 99.989357% 99.989555% 99.989456%
99.989853% 
99.989548% 99.986908% 99.989258% 99.989250% 99.989250% 99.990234%
99.989647% ^C 
--- Results after 14 passes --- 
Best: 99.990 -- Worst: 99.985 -- Average: 99.989048, Difference:
99.989048 

server_slave*CLI> iax2 show peers 
Name/Username Host Mask Port Status 
toulouse 222.222.222.222 (S) 255.255.255.255  (T) OK (18 ms) 
; 
; iax.conf server_salve 
; 
[general] 
bindport= 
bindaddr=111.111.111.111 
language=fr 
maxjitterbuffer=250 
trunktimestamps=yes 
tos=ef 

[toulouse] 
type=friend 
host=222.222.222.222 
port= 
context=iax-toulouse 
trunk=yes 
allow=alaw 
jitterbuffer=no 
forcejitterbuffer=no 
transfer=no 

Best regards 

Hugues 





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Re: [asterisk-users] IAX trunk mixing

[Steve Totaro]

Just use SIP so you don't have to back later to change all your IAX2
entries to SIP.


On Fri, Dec 5, 2008 at 4:23 AM, Tóth Csaba <[EMAIL PROTECTED]> wrote:
> hi,
>
> i have a problem, and i am completely stuck with it, i hope someone can
> point out where is my config wrong.
>
> I have three server, connect together with IAX trunking. The server are
> at romania (10.0.4.23, V1.4.22), hungary (10.0.1.23, V1.4.20) and serbia
> (10.0.3.4, V1.4.22). I have a hardphone (6251) connected to the romanian
> server, i dial a hungarian telephone number, the call goes to the
> hungarian server well, but that server recognise the call come from
> serbia.. and everything is mixed inside..
>

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Re: [asterisk-users] IAX trunk mixing

[Jim Dickenson]

I am not sure this is the best way but this is the way I have three servers
connected to each other.

On server 1 I have this:

   iax.conf:
  register => f2t1:[EMAIL PROTECTED]
  register => f3t1:[EMAIL PROTECTED]

  [f1t2]
  type=friend
  host=dynamic
  trunk=yes
  secret=password
  context=from_outside
  deny=0.0.0.0/0.0.0.0
  permit=2.2.2.2/255.255.255.255
  timezone=America/Los_Angeles

  [f1t3]
  type=friend
  host=dynamic
  trunk=yes
  secret=password
  context=from_outside
  deny=0.0.0.0/0.0.0.0
  permit=3.3.3.3/255.255.255.255
  timezone=America/Los_Angeles

   extensions.conf:
  exten => _2XX,1,NoOp(Calling to server 2)
  exten => _2XX,n,Dial(IAX2/f1t2/${EXTEN})
  exten => _2XX,n,Hangup()

  exten => _3XX,1,NoOp(Calling to server 3)
  exten => _3XX,n,Dial(IAX2/f1t3/${EXTEN})
  exten => _3XX,n,Hangup()

On server 2 I have this:

   iax.conf:
  register => f1t2:[EMAIL PROTECTED]
  register => f3t2:[EMAIL PROTECTED]

  [f2t1]
  type=friend
  host=dynamic
  trunk=yes
  secret=password
  context=from_outside
  deny=0.0.0.0/0.0.0.0
  permit=1.1.1.1/255.255.255.255
  timezone=America/Los_Angeles

  [f2t3]
  type=friend
  host=dynamic
  trunk=yes
  secret=password
  context=from_outside
  deny=0.0.0.0/0.0.0.0
  permit=3.3.3.3/255.255.255.255
  timezone=America/Los_Angeles

   extensions.conf:
  exten => _1XX,1,NoOp(Calling to server 1)
  exten => _1XX,n,Dial(IAX2/f2t1/${EXTEN})
  exten => _1XX,n,Hangup()

  exten => _3XX,1,NoOp(Calling to server 3)
  exten => _3XX,n,Dial(IAX2/f2t3/${EXTEN})
  exten => _3XX,n,Hangup()

On server 3 I have this:

   iax.conf:
  register => f1t3:[EMAIL PROTECTED]
  register => f2t3:[EMAIL PROTECTED]

  [f3t1]
  type=friend
  host=dynamic
  trunk=yes
  secret=password
  context=from_outside
  deny=0.0.0.0/0.0.0.0
  permit=1.1.1.1/255.255.255.255
  timezone=America/Los_Angeles

  [f3t2]
  type=friend
  host=dynamic
  trunk=yes
  secret=password
  context=from_outside
  deny=0.0.0.0/0.0.0.0
  permit=2.2.2.2/255.255.255.255
  timezone=America/Los_Angeles

   extensions.conf:
  exten => _1XX,1,NoOp(Calling to server 1)
  exten => _1XX,n,Dial(IAX2/f3t1/${EXTEN})
  exten => _1XX,n,Hangup()

  exten => _2XX,1,NoOp(Calling to server 2)
  exten => _2XX,n,Dial(IAX2/f3t2/${EXTEN})
  exten => _2XX,n,Hangup()

-- 
Jim Dickenson
mailto:[EMAIL PROTECTED]

CfMC
http://www.cfmc.com/



> From: Tóth Csaba <[EMAIL PROTECTED]>
> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Date: Sat, 06 Dec 2008 22:14:00 +0200
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Subject: Re: [asterisk-users] IAX trunk mixing
> 
> Hi List,

Help me pls, or you think this can be an asterisk bug and should i
> make
a bug report?

thanks,
Csaba



Tóth Csaba írta:
> hi,
> 
> i have a
> problem, and i am completely stuck with it, i hope someone can
> point out
> where is my config wrong.
> 
> I have three server, connect together with IAX
> trunking. The server are
> at romania (10.0.4.23, V1.4.22), hungary
> (10.0.1.23, V1.4.20) and serbia
> (10.0.3.4, V1.4.22). I have a hardphone
> (6251) connected to the romanian
> server, i dial a hungarian telephone
> number, the call goes to the
> hungarian server well, but that server
> recognise the call come from
> serbia.. and everything is mixed inside..
> 
>
> the phone starts at context do-phoning on the romanian server.
> i called
> 003620XXX from the phone, and as you see, the romanian
> server starts the
> call in good IAX trunk, but the hungarian server
> identifies it badly..
> 
>
> Here is the message on the HUNGARIAN asterisk console about it:
> 
> --
> Accepting AUTHENTICATED call from 10.0.4.23:
>> requested format =
> speex,
>> requested prefs = (gsm),
>> actual format = gsm,
>
> > host prefs = (),
>> priority = caller
> -- Executing
> [EMAIL PROTECTED]:1]
> MixMonitor("IAX2/telsrv-husrb-1541",
> "om_1228466966.19588_6251.wav") in
> new stack
>   == Begin MixMonitor
> Recording IAX2/telsrv-husrb-1541
> -- Executing
> [EMAIL PROTECTED]:2]
> Macro("IAX2/telsrv-husrb-1541",
> "kitelco|0620XXX") in new stack
> -- Executing [EMAIL PROTECTED]:1]
> Set("IAX2/telsrv-husrb-1541",
> "telszam=0620XXX") in new stack
> --
> Executing [EMAIL PROTECTED]:2] Dial("IAX2/telsrv-husrb-1541",
>
> "ZAP/g2/0620

Re: [asterisk-users] IAX trunk mixing

[Tim Panton]

If you set IAX2 debug on the HUNGARIAN machine and send the console  
output
(or a wireshark output) I'll take a look.
At a guess it is a problem with your iax.conf file.

I generally find it clearer to have separate user and peer definitions  
for
each system rather than relying on 'friend' which can be confusing.

Tim.

On 6 Dec 2008, at 20:14, Tóth Csaba wrote:

> Hi List,
>
> Help me pls, or you think this can be an asterisk bug and should i  
> make
> a bug report?
>
> thanks,
> Csaba
>
>
>
> Tóth Csaba írta:
>> hi,
>>
>> i have a problem, and i am completely stuck with it, i hope someone  
>> can
>> point out where is my config wrong.
>>
>> I have three server, connect together with IAX trunking. The server  
>> are
>> at romania (10.0.4.23, V1.4.22), hungary (10.0.1.23, V1.4.20) and  
>> serbia
>> (10.0.3.4, V1.4.22). I have a hardphone (6251) connected to the  
>> romanian
>> server, i dial a hungarian telephone number, the call goes to the
>> hungarian server well, but that server recognise the call come from
>> serbia.. and everything is mixed inside..
>>
>> the phone starts at context do-phoning on the romanian server.
>> i called 003620XXX from the phone, and as you see, the romanian
>> server starts the call in good IAX trunk, but the hungarian server
>> identifies it badly..
>>
>> Here is the message on the HUNGARIAN asterisk console about it:
>>
>>-- Accepting AUTHENTICATED call from 10.0.4.23:
>>> requested format = speex,
>>> requested prefs = (gsm),
>>> actual format = gsm,
>>> host prefs = (),
>>> priority = caller
>>-- Executing [EMAIL PROTECTED]:1]
>> MixMonitor("IAX2/telsrv-husrb-1541",  
>> "om_1228466966.19588_6251.wav") in
>> new stack
>>  == Begin MixMonitor Recording IAX2/telsrv-husrb-1541
>>-- Executing [EMAIL PROTECTED]:2]
>> Macro("IAX2/telsrv-husrb-1541", "kitelco|0620XXX") in new stack
>>-- Executing [EMAIL PROTECTED]:1] Set("IAX2/telsrv-husrb-1541",
>> "telszam=0620XXX") in new stack
>>-- Executing [EMAIL PROTECTED]:2] Dial("IAX2/telsrv-husrb-1541",
>> "ZAP/g2/0620XXX") in new stack
>>-- Requested transfer capability: 0x00 - SPEECH
>>-- Called g2/0620XXX
>>-- Zap/37-1 is proceeding passing it to IAX2/telsrv-husrb-1541
>>
>>
>>
>>
>> here is ROMANIAN console:
>>
>> [Dec  5 08:51:34] -- Executing [EMAIL PROTECTED]:1]
>> Set("SIP/6251-00c888c0", "telszam=0620XXX") in new stack
>> [Dec  5 08:51:34] -- Executing [EMAIL PROTECTED]:2]
>> Set("SIP/6251-00c888c0", "~~EXTEN~~=s") in new stack
>> [Dec  5 08:51:34] -- Executing [EMAIL PROTECTED]:3]
>> Dial("SIP/6251-00c888c0", "IAX2/telsrv-huro/0620XXX") in new  
>> stack
>> [Dec  5 08:51:34] -- Called telsrv-huro/0620XXX
>> [Dec  5 08:51:34] -- Call accepted by 10.0.1.23 (format gsm)
>> [Dec  5 08:51:34] -- Format for call is gsm
>> [Dec  5 08:51:35] -- IAX2/telsrv-huro-16384 is proceeding  
>> passing it
>> to SIP/6251-00c888c0
>> [Dec  5 08:51:35] -- Hungup 'IAX2/telsrv-huro-16384'
>> [Dec  5 08:51:35]   == Spawn extension (macro-kitelsrvhu, s, 3)  
>> exited
>> non-zero on 'SIP/6251-00c888c0' in macro 'kitelsrvhu'
>> [Dec  5 08:51:35]   == Spawn extension (macro-kitelsrvhu, s, 3)  
>> exited
>> non-zero on 'SIP/6251-00c888c0'
>>
>>
>>
>> here are the snippets of the config files:
>>
>>
>> ROMANIAN server
>>
>> iax.conf:
>>
>> 
>> [telsrv-huro]
>> type=friend
>> host = 10.0.1.23
>> user = telsrv-huro
>> secret = xxx
>> bandwidth=low
>> qualify=yes
>> trunk=yes
>> timezone=Europe/Budapest
>> context=incoming-hu
>>
>> [telsrv-rosrb]
>> type=friend
>> host = 10.0.3.4
>> user = telsrv-rosrb
>> secret = xxx
>> bandwidth=low
>> qualify=yes
>> trunk=yes
>> timezone=Europe/Bucharest
>> context=incoming-srb
>> 
>>
>> extensions.ael:
>>
>> 
>> context do-phoning {
>> includes {
>> do-nationalcall;
>> }
>> }
>>
>> abstract context do-nationalcall {
>> _0036. => &kitelsrvhu(06${EXTEN:4});
>> _6[2-8]XX => &kitelsrvhu(${EXTEN});
>> _7[2-8]XX => &kitelsrvhu(${EXTEN});
>>
>> _00381. => &kitelsrvsrb(${EXTEN:4});
>> _51[567]X => &kitelsrvsrb(${EXTEN});
>> }
>>
>> context incoming-hu {
>> includes {
>>template-companynumbers;
>>template-spec;
>>template-helyi;
>>template-mobil;
>>template-orszagos;
>> }
>> }
>>
>> context incoming-srb {
>> includes {
>>template-companynumbers;
>>template-spec;
>>template-helyi;
>>template-mobil;
>>template-orszagos;
>> }
>> }
>>
>> macro kitelsrvhu(telszam) {
>>Dial(IAX2/telsrv-huro/${telszam});
>>
>>switch(${DIALSTATUS}) {
>>case CHANUNAVAIL:
>>Playback(/var/lib/asterisk/sounds/beeperr);
>>case CONGESTION:
>>Playback(/var/lib/asterisk/sounds/beeperr);
>>case BUSY:
>>Busy();
>>Wait(5);
>>};
>> 

Re: [asterisk-users] IAX trunk mixing

[Tóth Csaba]

Hi List,

Help me pls, or you think this can be an asterisk bug and should i make
a bug report?

thanks,
Csaba



Tóth Csaba írta:
> hi,
> 
> i have a problem, and i am completely stuck with it, i hope someone can
> point out where is my config wrong.
> 
> I have three server, connect together with IAX trunking. The server are
> at romania (10.0.4.23, V1.4.22), hungary (10.0.1.23, V1.4.20) and serbia
> (10.0.3.4, V1.4.22). I have a hardphone (6251) connected to the romanian
> server, i dial a hungarian telephone number, the call goes to the
> hungarian server well, but that server recognise the call come from
> serbia.. and everything is mixed inside..
> 
> the phone starts at context do-phoning on the romanian server.
> i called 003620XXX from the phone, and as you see, the romanian
> server starts the call in good IAX trunk, but the hungarian server
> identifies it badly..
> 
> Here is the message on the HUNGARIAN asterisk console about it:
> 
> -- Accepting AUTHENTICATED call from 10.0.4.23:
>> requested format = speex,
>> requested prefs = (gsm),
>> actual format = gsm,
>> host prefs = (),
>> priority = caller
> -- Executing [EMAIL PROTECTED]:1]
> MixMonitor("IAX2/telsrv-husrb-1541", "om_1228466966.19588_6251.wav") in
> new stack
>   == Begin MixMonitor Recording IAX2/telsrv-husrb-1541
> -- Executing [EMAIL PROTECTED]:2]
> Macro("IAX2/telsrv-husrb-1541", "kitelco|0620XXX") in new stack
> -- Executing [EMAIL PROTECTED]:1] Set("IAX2/telsrv-husrb-1541",
> "telszam=0620XXX") in new stack
> -- Executing [EMAIL PROTECTED]:2] Dial("IAX2/telsrv-husrb-1541",
> "ZAP/g2/0620XXX") in new stack
> -- Requested transfer capability: 0x00 - SPEECH
> -- Called g2/0620XXX
> -- Zap/37-1 is proceeding passing it to IAX2/telsrv-husrb-1541
> 
> 
> 
> 
> here is ROMANIAN console:
> 
> [Dec  5 08:51:34] -- Executing [EMAIL PROTECTED]:1]
> Set("SIP/6251-00c888c0", "telszam=0620XXX") in new stack
> [Dec  5 08:51:34] -- Executing [EMAIL PROTECTED]:2]
> Set("SIP/6251-00c888c0", "~~EXTEN~~=s") in new stack
> [Dec  5 08:51:34] -- Executing [EMAIL PROTECTED]:3]
> Dial("SIP/6251-00c888c0", "IAX2/telsrv-huro/0620XXX") in new stack
> [Dec  5 08:51:34] -- Called telsrv-huro/0620XXX
> [Dec  5 08:51:34] -- Call accepted by 10.0.1.23 (format gsm)
> [Dec  5 08:51:34] -- Format for call is gsm
> [Dec  5 08:51:35] -- IAX2/telsrv-huro-16384 is proceeding passing it
> to SIP/6251-00c888c0
> [Dec  5 08:51:35] -- Hungup 'IAX2/telsrv-huro-16384'
> [Dec  5 08:51:35]   == Spawn extension (macro-kitelsrvhu, s, 3) exited
> non-zero on 'SIP/6251-00c888c0' in macro 'kitelsrvhu'
> [Dec  5 08:51:35]   == Spawn extension (macro-kitelsrvhu, s, 3) exited
> non-zero on 'SIP/6251-00c888c0'
> 
> 
> 
> here are the snippets of the config files:
> 
> 
> ROMANIAN server
> 
> iax.conf:
> 
> 
> [telsrv-huro]
> type=friend
> host = 10.0.1.23
> user = telsrv-huro
> secret = xxx
> bandwidth=low
> qualify=yes
> trunk=yes
> timezone=Europe/Budapest
> context=incoming-hu
> 
> [telsrv-rosrb]
> type=friend
> host = 10.0.3.4
> user = telsrv-rosrb
> secret = xxx
> bandwidth=low
> qualify=yes
> trunk=yes
> timezone=Europe/Bucharest
> context=incoming-srb
> 
> 
> extensions.ael:
> 
> 
> context do-phoning {
>  includes {
>  do-nationalcall;
>  }
> }
> 
> abstract context do-nationalcall {
>  _0036. => &kitelsrvhu(06${EXTEN:4});
>  _6[2-8]XX => &kitelsrvhu(${EXTEN});
>  _7[2-8]XX => &kitelsrvhu(${EXTEN});
> 
>  _00381. => &kitelsrvsrb(${EXTEN:4});
>  _51[567]X => &kitelsrvsrb(${EXTEN});
> }
> 
> context incoming-hu {
>  includes {
> template-companynumbers;
> template-spec;
> template-helyi;
> template-mobil;
> template-orszagos;
>  }
> }
> 
> context incoming-srb {
>  includes {
> template-companynumbers;
> template-spec;
> template-helyi;
> template-mobil;
> template-orszagos;
>  }
> }
> 
> macro kitelsrvhu(telszam) {
> Dial(IAX2/telsrv-huro/${telszam});
> 
> switch(${DIALSTATUS}) {
> case CHANUNAVAIL:
> Playback(/var/lib/asterisk/sounds/beeperr);
> case CONGESTION:
> Playback(/var/lib/asterisk/sounds/beeperr);
> case BUSY:
> Busy();
> Wait(5);
> };
> Hangup();
> 
> }
> 
> macro kitelsrvsrb(telszam) {
> Dial(IAX2/telsrv-srbro/${telszam});
> 
> switch(${DIALSTATUS}) {
> case CHANUNAVAIL:
> Playback(/var/lib/asterisk/sounds/beeperr);
> case CONGESTION:
> Playback(/var/lib/asterisk/sounds/beeperr);
> case BUSY:
> Busy();
> Wait(5);
> };
> Hangup();
> 
> }
> ===

[asterisk-users] IAX trunk mixing

[Tóth Csaba]

hi,

i have a problem, and i am completely stuck with it, i hope someone can
point out where is my config wrong.

I have three server, connect together with IAX trunking. The server are
at romania (10.0.4.23, V1.4.22), hungary (10.0.1.23, V1.4.20) and serbia
(10.0.3.4, V1.4.22). I have a hardphone (6251) connected to the romanian
server, i dial a hungarian telephone number, the call goes to the
hungarian server well, but that server recognise the call come from
serbia.. and everything is mixed inside..

the phone starts at context do-phoning on the romanian server.
i called 003620XXX from the phone, and as you see, the romanian
server starts the call in good IAX trunk, but the hungarian server
identifies it badly..

Here is the message on the HUNGARIAN asterisk console about it:

-- Accepting AUTHENTICATED call from 10.0.4.23:
   > requested format = speex,
   > requested prefs = (gsm),
   > actual format = gsm,
   > host prefs = (),
   > priority = caller
-- Executing [EMAIL PROTECTED]:1]
MixMonitor("IAX2/telsrv-husrb-1541", "om_1228466966.19588_6251.wav") in
new stack
  == Begin MixMonitor Recording IAX2/telsrv-husrb-1541
-- Executing [EMAIL PROTECTED]:2]
Macro("IAX2/telsrv-husrb-1541", "kitelco|0620XXX") in new stack
-- Executing [EMAIL PROTECTED]:1] Set("IAX2/telsrv-husrb-1541",
"telszam=0620XXX") in new stack
-- Executing [EMAIL PROTECTED]:2] Dial("IAX2/telsrv-husrb-1541",
"ZAP/g2/0620XXX") in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g2/0620XXX
-- Zap/37-1 is proceeding passing it to IAX2/telsrv-husrb-1541




here is ROMANIAN console:

[Dec  5 08:51:34] -- Executing [EMAIL PROTECTED]:1]
Set("SIP/6251-00c888c0", "telszam=0620XXX") in new stack
[Dec  5 08:51:34] -- Executing [EMAIL PROTECTED]:2]
Set("SIP/6251-00c888c0", "~~EXTEN~~=s") in new stack
[Dec  5 08:51:34] -- Executing [EMAIL PROTECTED]:3]
Dial("SIP/6251-00c888c0", "IAX2/telsrv-huro/0620XXX") in new stack
[Dec  5 08:51:34] -- Called telsrv-huro/0620XXX
[Dec  5 08:51:34] -- Call accepted by 10.0.1.23 (format gsm)
[Dec  5 08:51:34] -- Format for call is gsm
[Dec  5 08:51:35] -- IAX2/telsrv-huro-16384 is proceeding passing it
to SIP/6251-00c888c0
[Dec  5 08:51:35] -- Hungup 'IAX2/telsrv-huro-16384'
[Dec  5 08:51:35]   == Spawn extension (macro-kitelsrvhu, s, 3) exited
non-zero on 'SIP/6251-00c888c0' in macro 'kitelsrvhu'
[Dec  5 08:51:35]   == Spawn extension (macro-kitelsrvhu, s, 3) exited
non-zero on 'SIP/6251-00c888c0'



here are the snippets of the config files:


ROMANIAN server

iax.conf:


[telsrv-huro]
type=friend
host = 10.0.1.23
user = telsrv-huro
secret = xxx
bandwidth=low
qualify=yes
trunk=yes
timezone=Europe/Budapest
context=incoming-hu

[telsrv-rosrb]
type=friend
host = 10.0.3.4
user = telsrv-rosrb
secret = xxx
bandwidth=low
qualify=yes
trunk=yes
timezone=Europe/Bucharest
context=incoming-srb


extensions.ael:


context do-phoning {
 includes {
 do-nationalcall;
 }
}

abstract context do-nationalcall {
 _0036. => &kitelsrvhu(06${EXTEN:4});
 _6[2-8]XX => &kitelsrvhu(${EXTEN});
 _7[2-8]XX => &kitelsrvhu(${EXTEN});

 _00381. => &kitelsrvsrb(${EXTEN:4});
 _51[567]X => &kitelsrvsrb(${EXTEN});
}

context incoming-hu {
 includes {
template-companynumbers;
template-spec;
template-helyi;
template-mobil;
template-orszagos;
 }
}

context incoming-srb {
 includes {
template-companynumbers;
template-spec;
template-helyi;
template-mobil;
template-orszagos;
 }
}

macro kitelsrvhu(telszam) {
Dial(IAX2/telsrv-huro/${telszam});

switch(${DIALSTATUS}) {
case CHANUNAVAIL:
Playback(/var/lib/asterisk/sounds/beeperr);
case CONGESTION:
Playback(/var/lib/asterisk/sounds/beeperr);
case BUSY:
Busy();
Wait(5);
};
Hangup();

}

macro kitelsrvsrb(telszam) {
Dial(IAX2/telsrv-srbro/${telszam});

switch(${DIALSTATUS}) {
case CHANUNAVAIL:
Playback(/var/lib/asterisk/sounds/beeperr);
case CONGESTION:
Playback(/var/lib/asterisk/sounds/beeperr);
case BUSY:
Busy();
Wait(5);
};
Hangup();

}





HUNGARIAN server

iax.conf:


[telsrv-huro]
type=friend
host = 10.0.4.23
user = telsrv-huro
secret = xxx
bandwidth=low
qualify=yes
trunk=yes
timezone=Europe/Bucharest
context=incoming-ro

[telsrv-husrb]
type=friend
host = 10.0.3.4
user = telsrv-husrb
secret = xxx
bandwidth=low
qualify=yes
trunk=yes
timezone=Europe/Beograd
context=incoming-srb


ex

Re: [asterisk-users] IAX Trunk between two Asterisks

[bilal ghayyad]

This is my configuration in the extensions.conf,
iax.conf at Site A and Site B, so anyone can help why
the call refused?

Site A:

[IPLink]
type=friend
context=IPLinkIncoming
host=192.168.2.3
usename=IPLink
secret=password
canreinvite=no
nat=no

[SiteBInternal]

exten => _2XX,1,Dial(IAX2/[EMAIL PROTECTED])
exten => _2XX,2,Playback(vm-nobodyavail)
exten => _2XX,3,Hangup()
exten => _2XX,102,Playback(tt-allbusy)
exten => _2XX,103,Hangup()

[IPLinkIncoming]

include => SiteBInternal
include => SiteBExternal

And at Site B:

[IPLink]
type=friend
context=IPLinkIncoming
host=192.168.2.2
usename=IPLink
secret=password
canreinvite=no
nat=no

[SiteAInternal]

exten => _2XX,1,Dial(IAX2/[EMAIL PROTECTED])
exten => _2XX,2,Playback(vm-nobodyavail)
exten => _2XX,3,Hangup()
exten => _2XX,102,Playback(tt-allbusy)
exten => _2XX,103,Hangup()

[IPLinkIncoming]

include => SiteAInternal
include => SiteAExternal

Regards
Bilal

--

> Hi All;
>
> I did an IP Trunk using IAX between two Asterisk
> boxes, now Asterisk A can send a call for B but B
> refuse it. The IAX type was configured to be
"friend"
> in the iax.con for Asterisk A and B, is there any
> thing else need to be done to let B accept the call
> from A?
>
> Also, I used an static IP address for the host when
I
> configured the iax client in the iax.conf file.
>
> Any help?
> Regards
> Bilal
>

I used to see this problem when I used to use IAX2. 
Sometimes it would
 just
go away.  I seem to remember using insecure=very to
get it working but
 I may
be wrong.

Anyways, post the relevant parts of your IAX2 confs
from both boxes and
someone might be able to spot something right off the
bat.

Thanks,
Steve Totaro



  

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Re: [asterisk-users] IAX Trunk between two Asterisks: Authority, and Call Rejected

[Steve Totaro]

On Jan 16, 2008 8:46 AM, bilal ghayyad <[EMAIL PROTECTED]> wrote:

> Hi All;
>
> I did an IP Trunk using IAX between two Asterisk
> boxes, now Asterisk A can send a call for B but B
> refuse it. The IAX type was configured to be "friend"
> in the iax.con for Asterisk A and B, is there any
> thing else need to be done to let B accept the call
> from A?
>
> Also, I used an static IP address for the host when I
> configured the iax client in the iax.conf file.
>
> Any help?
> Regards
> Bilal
>

I used to see this problem when I used to use IAX2.  Sometimes it would just
go away.  I seem to remember using insecure=very to get it working but I may
be wrong.

Anyways, post the relevant parts of your IAX2 confs from both boxes and
someone might be able to spot something right off the bat.

Thanks,
Steve Totaro
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[asterisk-users] IAX Trunk between two Asterisks: Authority, and Call Rejected

[bilal ghayyad]

Hi All;

I did an IP Trunk using IAX between two Asterisk
boxes, now Asterisk A can send a call for B but B
refuse it. The IAX type was configured to be "friend"
in the iax.con for Asterisk A and B, is there any
thing else need to be done to let B accept the call
from A?

Also, I used an static IP address for the host when I
configured the iax client in the iax.conf file.

Any help?
Regards
Bilal


  

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[asterisk-users] IAX Trunk

[Jeremy Mann]

Is there a way to limit IAX trunks to a certain number of calls?  For instance, 
if I'm linking two systems in different regions, can I limit the number of 
calls that go across IAX between the systems?

I've got some dialplan logic, but if there's some iax.conf directive to limit 
the number of calls it'd be so much simpler.


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Re: [asterisk-users] IAX trunk with dynamic IPs

[Jaswinder Singh]

Hello
You should use qualify=310 ( any value in millisec ) .. qualify=yes
is not proper .

I am not sure about how asterisk's dnsmgr manages dns refreshing but
maybe someone else can answer that question .

On 10/06/07, Ronaldo Z. Afonso <[EMAIL PROTECTED]> wrote:

Hi Jaswinder,

That is what I did. The thing now is, when I set "enable=yes" in
/etc/asterisk/dnsmgr, Asterisk stops sending IAX POKE messages to the
remote peer (to keep the trunk up).
I've searched on the Internet but I couldn't find any documentation
about how "DNS update manager" works for Asterisk. Do you have any?

Ronaldo.

Jaswinder Singh wrote:
> In your no-ip client set it to update ip every 2 minutes or so . and
> /etc/asterisk/dnsmgr.conf  set refresh interval as 30-40 by defualt
> its 300 ( 5 minutes)
>
> On 10/06/07, Ronaldo Z. Afonso <[EMAIL PROTECTED]> wrote:
>> Hi Matt,
>>
>> Every time I do that, IAX stop sending the POKE messages (necessary for
>> trunk management).
>> Do you know what could be happening?
>>
>> Thanks.
>> Ronaldo.
>>
>> Matt wrote:
>> >
>> > *set "enable=yes" in the "[general]" section of
>> > /etc/asterisk/dnsmgr.conf*
>> >
>> >
>> >
>> 
>> >
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Re: [asterisk-users] IAX trunk with dynamic IPs

[Noah Miller]

Hi Ronaldo -


I have a IAX trunk between two asterisk servers, both with dynamic IP
and both have a DNS name associated with it.
In the iax.conf file I configure the "host" parameter with the DNS name
of the servers. Everything works fine until one of these servers get a
new IP, so the other can't find its peer (the one that has just gotten a
new IP). If I manually issue a "iax2 reload" in the CLI, asterisk tries
to find the IP of the peer (based on its DNS name) and everything starts
working again.
This is the section for my trunk in one of my servers:

[sometrunk]
type=friend
username=someusername
secret=somesecret
auth=plaintext
host=host.no-ip.org
context=incoming
peercontext=incoming
qualify=yes
trunk=yes


Is there any way to tell asterisk to try to find the peer's IP address
if that peer is "unreachable" or each 10 minutes?


I don't know if your DDNS provider would support this, but if you set
the TTL value of your DNS hostnames to something very low, like 10
seconds, it would force your OS to keep finding the latest IP.


- Noah
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Re: [asterisk-users] IAX trunk with dynamic IPs

[Ronaldo Z. Afonso]

Hi Jaswinder,

That is what I did. The thing now is, when I set "enable=yes" in 
/etc/asterisk/dnsmgr, Asterisk stops sending IAX POKE messages to the 
remote peer (to keep the trunk up).
I've searched on the Internet but I couldn't find any documentation 
about how "DNS update manager" works for Asterisk. Do you have any?


Ronaldo.

Jaswinder Singh wrote:

In your no-ip client set it to update ip every 2 minutes or so . and
/etc/asterisk/dnsmgr.conf  set refresh interval as 30-40 by defualt
its 300 ( 5 minutes)

On 10/06/07, Ronaldo Z. Afonso <[EMAIL PROTECTED]> wrote:

Hi Matt,

Every time I do that, IAX stop sending the POKE messages (necessary for
trunk management).
Do you know what could be happening?

Thanks.
Ronaldo.

Matt wrote:
>
> *set "enable=yes" in the "[general]" section of
> /etc/asterisk/dnsmgr.conf*
>
>
> 


>
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Re: [asterisk-users] IAX trunk with dynamic IPs

[Jaswinder Singh]

In your no-ip client set it to update ip every 2 minutes or so . and
/etc/asterisk/dnsmgr.conf  set refresh interval as 30-40 by defualt
its 300 ( 5 minutes)

On 10/06/07, Ronaldo Z. Afonso <[EMAIL PROTECTED]> wrote:

Hi Matt,

Every time I do that, IAX stop sending the POKE messages (necessary for
trunk management).
Do you know what could be happening?

Thanks.
Ronaldo.

Matt wrote:
>
> *set "enable=yes" in the "[general]" section of
> /etc/asterisk/dnsmgr.conf*
>
>
> 
>
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Re: [asterisk-users] IAX trunk with dynamic IPs

[Ronaldo Z. Afonso]

Hi Matt,

Every time I do that, IAX stop sending the POKE messages (necessary for 
trunk management).

Do you know what could be happening?

Thanks.
Ronaldo.

Matt wrote:


*set "enable=yes" in the "[general]" section of 
/etc/asterisk/dnsmgr.conf*





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Re: [asterisk-users] IAX trunk with dynamic IPs

[Matt]

*set "enable=yes" in the "[general]" section of /etc/asterisk/dnsmgr.conf*
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Re: [asterisk-users] IAX trunk with dynamic IPs

[Justin Moore]

On 6/9/07, Ronaldo Z. Afonso <[EMAIL PROTECTED]> wrote:

First of all, thanks for your help. I just want to check if I understood.
If a set the TTL for 10 seconds for host.no-ip.org and configure the
parameter host as "host=host.no-ip.org", Asterisk will try to find the
IP address of host.no-ip.org each 10 seconds? That is it?


I believe this sort of thing came up a few weeks ago on the list. What
I remember the outcome being was that Asterisk currently will only do
the DNS lookup when you initially start or do a reload from the
Asterisk CLI. So, modifying the TTL on your DNS records will have no
effect unless Asterisk is patched to query the DNS each time instead
of relying on it's DNS cache.

--
Justin Moore
aka wantmoore
---
www.wantmoore.com
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Re: [asterisk-users] IAX trunk with dynamic IPs

[Ronaldo Z. Afonso]

Hi Noah,

First of all, thanks for your help. I just want to check if I understood.
If a set the TTL for 10 seconds for host.no-ip.org and configure the 
parameter host as "host=host.no-ip.org", Asterisk will try to find the 
IP address of host.no-ip.org each 10 seconds? That is it?


Thanks again.

Ronaldo.

Noah Miller wrote:

Hi Ronaldo -


I have a IAX trunk between two asterisk servers, both with dynamic IP
and both have a DNS name associated with it.
In the iax.conf file I configure the "host" parameter with the DNS name
of the servers. Everything works fine until one of these servers get a
new IP, so the other can't find its peer (the one that has just gotten a
new IP). If I manually issue a "iax2 reload" in the CLI, asterisk tries
to find the IP of the peer (based on its DNS name) and everything starts
working again.
This is the section for my trunk in one of my servers:

[sometrunk]
type=friend
username=someusername
secret=somesecret
auth=plaintext
host=host.no-ip.org
context=incoming
peercontext=incoming
qualify=yes
trunk=yes


Is there any way to tell asterisk to try to find the peer's IP address
if that peer is "unreachable" or each 10 minutes?


I don't know if your DDNS provider would support this, but if you set
the TTL value of your DNS hostnames to something very low, like 10
seconds, it would force your OS to keep finding the latest IP.


- Noah
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[asterisk-users] IAX trunk with dynamic IPs

[Ronaldo Z. Afonso]

Hi all,

I have a IAX trunk between two asterisk servers, both with dynamic IP 
and both have a DNS name associated with it.
In the iax.conf file I configure the "host" parameter with the DNS name 
of the servers. Everything works fine until one of these servers get a 
new IP, so the other can't find its peer (the one that has just gotten a 
new IP). If I manually issue a "iax2 reload" in the CLI, asterisk tries 
to find the IP of the peer (based on its DNS name) and everything starts 
working again.

This is the section for my trunk in one of my servers:

[sometrunk]
type=friend
username=someusername
secret=somesecret
auth=plaintext
host=host.no-ip.org
context=incoming
peercontext=incoming
qualify=yes
trunk=yes


Is there any way to tell asterisk to try to find the peer's IP address 
if that peer is "unreachable" or each 10 minutes?


Thanks in advance.
Ronaldo.
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Re: [asterisk-users] IAX Trunk

[Ronaldo]

Hi All,

I'd like to thank everyone that answer my question about IAX Trunk. Now 
I have a working IAX trunking, I just need to tune it.


Thank you.
Ronaldo.

Salvatore Giudice wrote:

Yes of course. If you want to limit it, I think you have to set
'incominglimit' and/or 'outgoinglimit'.

--
Salvatore Giudice
[EMAIL PROTECTED]

VoIP Security Training, LLC
http://VoIPSecurityTraining.com

848 N. Rainbow Blvd. #1676
Las Vegas, NV 89107
Phone: (617) 959-7625
Fax: (214) 279-2906

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ronaldo
Sent: Thursday, May 03, 2007 2:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] IAX Trunk

OK Steve,

Just one more question. Using this configuration can I make more than 
one call at the same time?


Thanks.

Steve Kennedy wrote:
  

On Thu, May 03, 2007 at 01:22:07PM -0300, Ronaldo wrote:

  


Can you suggest me any documentation about using IAX trunking?
Thank you.

  

There are examples in the iax.conf files I think, but basically just put
something like

[iax-toremote]
type=friend
username=whatever
secret=somesecret
auth=plaintext
host=somewhere.com
peercontext=some-context
qualify=yes
trunk=yes

then you dial with Dial(iax2/iax-toremote/number)


Steve

  



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Re: [asterisk-users] IAX Trunk

[Steve Kennedy]

On Thu, May 03, 2007 at 03:23:16PM -0300, Ronaldo wrote:

> OK Steve,
> Just one more question. Using this configuration can I make more than 
> one call at the same time?

The whole point of trunking is to support multiple "calls" down the same
IAX trunk (well actually down the same packets).


Steve

-- 
NetTek Ltd  UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED]
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RE: [asterisk-users] IAX Trunk

[Salvatore Giudice]

Yes of course. If you want to limit it, I think you have to set
'incominglimit' and/or 'outgoinglimit'.

--
Salvatore Giudice
[EMAIL PROTECTED]

VoIP Security Training, LLC
http://VoIPSecurityTraining.com

848 N. Rainbow Blvd. #1676
Las Vegas, NV 89107
Phone: (617) 959-7625
Fax: (214) 279-2906

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ronaldo
Sent: Thursday, May 03, 2007 2:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] IAX Trunk

OK Steve,

Just one more question. Using this configuration can I make more than 
one call at the same time?

Thanks.

Steve Kennedy wrote:
> On Thu, May 03, 2007 at 01:22:07PM -0300, Ronaldo wrote:
>
>   
>> Can you suggest me any documentation about using IAX trunking?
>> Thank you.
>> 
>
> There are examples in the iax.conf files I think, but basically just put
> something like
>
> [iax-toremote]
> type=friend
> username=whatever
> secret=somesecret
> auth=plaintext
> host=somewhere.com
> peercontext=some-context
> qualify=yes
> trunk=yes
>
> then you dial with Dial(iax2/iax-toremote/number)
>
>
> Steve
>
>   

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Re: [asterisk-users] IAX Trunk

[Ronaldo]

OK Steve,

Just one more question. Using this configuration can I make more than 
one call at the same time?


Thanks.

Steve Kennedy wrote:

On Thu, May 03, 2007 at 01:22:07PM -0300, Ronaldo wrote:

  

Can you suggest me any documentation about using IAX trunking?
Thank you.



There are examples in the iax.conf files I think, but basically just put
something like

[iax-toremote]
type=friend
username=whatever
secret=somesecret
auth=plaintext
host=somewhere.com
peercontext=some-context
qualify=yes
trunk=yes

then you dial with Dial(iax2/iax-toremote/number)


Steve

  


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RE: [asterisk-users] IAX Trunk

[Salvatore Giudice]

Good luck. Try these.

http://www.voip-info.org/wiki-IAX

http://www.voip-info.org/wiki-IAX+versus+SIP

http://www.voip-info.org/wiki/view/IAX+encryption

http://www.voip-info.org/wiki/index.php?page=Asterisk+config+iax.conf

http://www.voip-info.org/wiki/index.php?page=Asterisk+IAX+channels



--
Salvatore Giudice
[EMAIL PROTECTED]

VoIP Security Training, LLC
http://VoIPSecurityTraining.com

848 N. Rainbow Blvd. #1676
Las Vegas, NV 89107
Phone: (617) 959-7625
Fax: (214) 279-2906


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ronaldo
Sent: Thursday, May 03, 2007 12:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] IAX Trunk

Hi Bruce,

Can you suggest me any documentation about using IAX truking?
Thank you.

Ronaldo.

Bruce Reeves wrote:
> Yes it is.
>
> On 5/3/07, *Ronaldo* <[EMAIL PROTECTED] 
> <mailto:[EMAIL PROTECTED]>> wrote:
>
> Hi all,
>
> Is it possible to have something like this:
>
> SoftPhone -(SIP)-> Asterisk -(IAX trunk)-> Asterisk -(SIP)-> SoftPhone
>
> I want a IAX trunk between two asterisks and on each tip I have SIP
> clients that need to talk to each other.
>
> Thansk.
>
> Ronaldo
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>
>
> -- 
> Bruce Reeves
> Nortex Networks
> 
>
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Re: [asterisk-users] IAX Trunk

[Bruce Reeves]

The wiki has a decent page about it.

http://www.voip-info.org/wiki-IAX

What you are trying to setup sounds simple enoug, you mainly will have an
extension or pattern match that executes a dial command  from box A to box B
and passes the remote extension.

On 5/3/07, Ronaldo <[EMAIL PROTECTED]> wrote:


Hi Bruce,

Can you suggest me any documentation about using IAX truking?
Thank you.

Ronaldo.

Bruce Reeves wrote:
> Yes it is.
>
> On 5/3/07, *Ronaldo* <[EMAIL PROTECTED]
> > wrote:
>
> Hi all,
>
> Is it possible to have something like this:
>
> SoftPhone -(SIP)-> Asterisk -(IAX trunk)-> Asterisk -(SIP)->
SoftPhone
>
> I want a IAX trunk between two asterisks and on each tip I have SIP
> clients that need to talk to each other.
>
> Thansk.
>
> Ronaldo
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> asterisk-users mailing list
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>
>
>
> --
> Bruce Reeves
> Nortex Networks
> 
>
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--
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Nortex Networks
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Re: [asterisk-users] IAX Trunk

[Gordon Henderson]

On Thu, 3 May 2007, Ronaldo wrote:


Hi all,

Is it possible to have something like this:

SoftPhone -(SIP)-> Asterisk -(IAX trunk)-> Asterisk -(SIP)-> SoftPhone

I want a IAX trunk between two asterisks and on each tip I have SIP clients 
that need to talk to each other.


Absolutely.

Have a look at this: http://astrecipes.net/index.php?n=204

Gordon
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Re: [asterisk-users] IAX Trunk

[Steve Kennedy]

On Thu, May 03, 2007 at 01:22:07PM -0300, Ronaldo wrote:

> Can you suggest me any documentation about using IAX trunking?
> Thank you.

There are examples in the iax.conf files I think, but basically just put
something like

[iax-toremote]
type=friend
username=whatever
secret=somesecret
auth=plaintext
host=somewhere.com
peercontext=some-context
qualify=yes
trunk=yes

then you dial with Dial(iax2/iax-toremote/number)


Steve

-- 
NetTek Ltd  UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED]
Euro Tech News Blog http://eurotechnews.blogspot.com
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Re: [asterisk-users] IAX Trunk

[Ronaldo]

Hi Bruce,

Can you suggest me any documentation about using IAX truking?
Thank you.

Ronaldo.

Bruce Reeves wrote:

Yes it is.

On 5/3/07, *Ronaldo* <[EMAIL PROTECTED] 
> wrote:


Hi all,

Is it possible to have something like this:

SoftPhone -(SIP)-> Asterisk -(IAX trunk)-> Asterisk -(SIP)-> SoftPhone

I want a IAX trunk between two asterisks and on each tip I have SIP
clients that need to talk to each other.

Thansk.

Ronaldo
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--
Bruce Reeves
Nortex Networks


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Re: [asterisk-users] IAX Trunk

[Ronaldo]

Hi Dean,

Can you suggest me any documentation about using IAX trunking?
Thank you.

Ronaldo.

Dean Collins wrote:
Yes it is. 


Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph


  

-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Ronaldo
Sent: Thursday, 3 May 2007 12:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] IAX Trunk

Hi all,

Is it possible to have something like this:

SoftPhone -(SIP)-> Asterisk -(IAX trunk)-> Asterisk -(SIP)-> SoftPhone

I want a IAX trunk between two asterisks and on each tip I have SIP
clients that need to talk to each other.

Thansk.

Ronaldo
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Re: [asterisk-users] IAX Trunk

[William Moore]

On 5/3/07, Ronaldo <[EMAIL PROTECTED]> wrote:

Hi all,

Is it possible to have something like this:

SoftPhone -(SIP)-> Asterisk -(IAX trunk)-> Asterisk -(SIP)-> SoftPhone

I want a IAX trunk between two asterisks and on each tip I have SIP
clients that need to talk to each other.


Yes, Asterisk will do the conversion from SIP to IAX and back again if
necessary.
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Re: [asterisk-users] IAX Trunk

[Bruce Reeves]

Yes it is

On 5/3/07, Ronaldo <[EMAIL PROTECTED]> wrote:


Hi all,

Is it possible to have something like this:

SoftPhone -(SIP)-> Asterisk -(IAX trunk)-> Asterisk -(SIP)-> SoftPhone

I want a IAX trunk between two asterisks and on each tip I have SIP
clients that need to talk to each other.

Thansk.

Ronaldo
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--
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Nortex Networks
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Re: [asterisk-users] IAX Trunk

[Bruce Reeves]

Yes it is.

On 5/3/07, Ronaldo <[EMAIL PROTECTED]> wrote:


Hi all,

Is it possible to have something like this:

SoftPhone -(SIP)-> Asterisk -(IAX trunk)-> Asterisk -(SIP)-> SoftPhone

I want a IAX trunk between two asterisks and on each tip I have SIP
clients that need to talk to each other.

Thansk.

Ronaldo
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--
Bruce Reeves
Nortex Networks
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RE: [asterisk-users] IAX Trunk

[Dean Collins]

Yes it is. 

Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Ronaldo
> Sent: Thursday, 3 May 2007 12:07 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] IAX Trunk
> 
> Hi all,
> 
> Is it possible to have something like this:
> 
> SoftPhone -(SIP)-> Asterisk -(IAX trunk)-> Asterisk -(SIP)-> SoftPhone
> 
> I want a IAX trunk between two asterisks and on each tip I have SIP
> clients that need to talk to each other.
> 
> Thansk.
> 
> Ronaldo
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[asterisk-users] IAX Trunk

[Ronaldo]

Hi all,

Is it possible to have something like this:

SoftPhone -(SIP)-> Asterisk -(IAX trunk)-> Asterisk -(SIP)-> SoftPhone

I want a IAX trunk between two asterisks and on each tip I have SIP 
clients that need to talk to each other.


Thansk.

Ronaldo
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Re: [asterisk-users] IAX Trunk Failover

[Andrew Joakimsen]

On 4/5/07, Mike Lynchfield <[EMAIL PROTECTED]> wrote:

tried x+102 ?




NEVER do that. The call can fail for other reasons besides the
carrier. It can and will create conditions where your carrier properly
connects the call and then the call is re-dialed via another provider
or line.
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Re: [asterisk-users] IAX Trunk Failover

[Justin Hamade]

You have call-${DIALSTATUS} and s-CONGESTION.

It might not be CONGESTION.  Do a Noop(${DIALSTATUS}) you should get
something.

Justin

On 4/5/07, Mike Lynchfield <[EMAIL PROTECTED]> wrote:


tried x+102 ?

On 4/5/07, Brent <[EMAIL PROTECTED]> wrote:

>  I'm trying to get an IAX trunk to failover to a local trunk it the
> trunk is down.
>
>
>
> This is what I've been working on:
>
>
>
> [macro-forward1];
>
> exten => s,1,Dial(IAX2/192.168.1.1/${ARG1},20)
>
> exten => s,2,Goto(call-${DIALSTATUS},1)
>
> exten => s-CONGESTION,1,Dial(LOCAL/${ARG2},20)
>
> exten => s-CHANUNAVAIL,1,Dial(LOCAL/${ARG2},20
>
> ;end macro-forward1
>
>
>
> exten => 6222626,1,Macro(forward1,6222626,6222627)
>
>
>
> ...in the debug, I never see dialstatus...the call just fails.  Doesn't
> ever try to dial the second extension.
>
>
>
> Any ideas?
>
>
>
>
>
>
>
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>


--
Mike
Sales Manager
http://www.voicemeup.com
Making it happen
1.877.807.VOIP (8647)
1.514.312.7030
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--
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Re: [asterisk-users] IAX Trunk Failover

[Mike Lynchfield]

tried x+102 ?

On 4/5/07, Brent <[EMAIL PROTECTED]> wrote:


 I'm trying to get an IAX trunk to failover to a local trunk it the trunk
is down.



This is what I've been working on:



[macro-forward1];

exten => s,1,Dial(IAX2/192.168.1.1/${ARG1},20)

exten => s,2,Goto(call-${DIALSTATUS},1)

exten => s-CONGESTION,1,Dial(LOCAL/${ARG2},20)

exten => s-CHANUNAVAIL,1,Dial(LOCAL/${ARG2},20

;end macro-forward1



exten => 6222626,1,Macro(forward1,6222626,6222627)



...in the debug, I never see dialstatus...the call just fails.  Doesn't
ever try to dial the second extension.



Any ideas?







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--
Mike
Sales Manager
http://www.voicemeup.com
Making it happen
1.877.807.VOIP (8647)
1.514.312.7030
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[asterisk-users] IAX Trunk Failover

[Brent]

I'm trying to get an IAX trunk to failover to a local trunk it the trunk is
down.

 

This is what I've been working on:

 

[macro-forward1];

exten => s,1,Dial(IAX2/192.168.1.1/${ARG1},20)

exten => s,2,Goto(call-${DIALSTATUS},1)

exten => s-CONGESTION,1,Dial(LOCAL/${ARG2},20)

exten => s-CHANUNAVAIL,1,Dial(LOCAL/${ARG2},20

;end macro-forward1

 

exten => 6222626,1,Macro(forward1,6222626,6222627)

 

...in the debug, I never see dialstatus...the call just fails.  Doesn't ever
try to dial the second extension.

 

Any ideas?

 

 

 

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Re: [asterisk-users] IAX Trunk timing

[Michiel van Baak]

On 12:06, Tue 16 Jan 07, Andy Hester wrote:
> > -Original Message-
> > From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > [EMAIL PROTECTED] On Behalf Of Zoa
> > Sent: Tuesday, January 16, 2007 11:08 AM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [asterisk-users] IAX Trunk timing
> > 
> > 
> > You need a timing device on both ends.
> > 
> > Zoa
> > 
> 
> But ztdummy should suffice yes?

yes
-- 

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RE: [asterisk-users] IAX Trunk timing

[Andy Hester]

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Zoa
> Sent: Tuesday, January 16, 2007 11:08 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] IAX Trunk timing
> 
> 
> You need a timing device on both ends.
> 
> Zoa
> 

But ztdummy should suffice yes?

Andy


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Re: [asterisk-users] IAX Trunk timing

[Zoa]

You need a timing device on both ends.

Zoa

Vicky wrote:
If the other server doesnt have any hardware device that can act as 
timer. then just compile zaptel and modprobe ztdummy .. This kernel 
module should act as timing source i think . ( it works with meetme ) .


On 16/01/07, *Andy Hester* <[EMAIL PROTECTED] 
> wrote:


I have read that an IAX trunk requires a timing device.  What wasn't
clear to me was whether it is like TDM ie 1 timing device for the
trunk,
or if each end requires a timing device.  I have a zaptel card in one
server; do I have to have one in the second server in order to do
an IAX
trunk?

I set up a trunk and so far calls can be made one way, but not the
other.  It is probably just not configured correctly, but I just
wanted
to make sure as I can't seem to find any reason at the moment.

Thanks,
Andy

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Re: [asterisk-users] IAX Trunk timing

[Vicky]

If the other server doesnt have any hardware device that can act as timer.
then just compile zaptel and modprobe ztdummy .. This kernel module should
act as timing source i think . ( it works with meetme ) .

On 16/01/07, Andy Hester <[EMAIL PROTECTED]> wrote:


I have read that an IAX trunk requires a timing device.  What wasn't
clear to me was whether it is like TDM ie 1 timing device for the trunk,
or if each end requires a timing device.  I have a zaptel card in one
server; do I have to have one in the second server in order to do an IAX
trunk?

I set up a trunk and so far calls can be made one way, but not the
other.  It is probably just not configured correctly, but I just wanted
to make sure as I can't seem to find any reason at the moment.

Thanks,
Andy

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[asterisk-users] IAX Trunk timing

[Andy Hester]

I have read that an IAX trunk requires a timing device.  What wasn't
clear to me was whether it is like TDM ie 1 timing device for the trunk,
or if each end requires a timing device.  I have a zaptel card in one
server; do I have to have one in the second server in order to do an IAX
trunk?

I set up a trunk and so far calls can be made one way, but not the
other.  It is probably just not configured correctly, but I just wanted
to make sure as I can't seem to find any reason at the moment.

Thanks,
Andy

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[asterisk-users] IAX trunk problem

[Lee Archer]

> I wonder if anyone can help me with this.  I have 4 sites running
> Asterisk and these are linked via IAX trunks and ADSL lines.  Calls
> coming into any of these sites are received locally and forwarded to a
> central operator.  E.g.  Call comes in on site A and is forwarded to
> the operator on site B.  99 out of 100 times the operator will send
> the call back to someone at the site from where it came but site B's
> Asterisk server seems to be staying in the loop.  E.g. A > B > A.
> I've had a look and can't see anything obvious as I had assumed that
> Asterisk would pass the call off.  I've tried notransfer on the trunks
> but site B's Asterisk server doesn't seem to be joining the endpoints
> and staying in the loop and therefore the call is going over the
> trunks twice.
> 
> Thanks
> 
> Lee
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[asterisk-users] IAX trunk problem

[Lee Archer]

I wonder if anyone can help me with this.  I have 4 sites running
asterisk and calls coming into any of these sites are received locally
and forwarded to a central operator.  E.g.  Call comes in on site A and
is forwarded to the operator on site B.  99/100 the operator will send
the call back to the site from where it came but site B's Asterisk
server seems to be staying in the loop.  E.g. A > B > A.  I've had a
look and can't see anything obvious as I had assumed that asterisk would
pass the call off.

Thanks

Lee
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Re: [asterisk-users] IAX trunk behing NAT with dynamic IP

[Michiel van Baak]

On 07:51, Tue 08 Aug 06, DM wrote:
> If that doesn't work, the OP can set up a cron job to do a IAX2 Reload
> every 5 minutes like I did.  I have the exact same problem.
> 
> I hope the OP reports back if that works, because I would rather add
> the host statement than run a cron job every 5 minutes.
> 
> On 8/8/06, Jon Schøpzinsky <[EMAIL PROTECTED]> wrote:
> >Set the host=dynamic on serverA, and let the serverB register with serverA
> >
> >Jon
> >

Another option is to create a shellscript that writes a
tempfile with the current ip. Next time it's run it will
compare the current ip with the recorded ip.
If they are the same, terminate.
If they are not, run an IAX2 reload.
That way you will only reload when needed.

I remember there are tools that allow you to run scripts on
ip address change, but cant find them in my 1 minute search.
-- 
Michiel van Baak
[EMAIL PROTECTED]
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GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD

"Why is it drug addicts and computer afficionados are both called users?"

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V: [asterisk-users] IAX trunk behing NAT with dynamic IP

[Andre Courchesne - Consultant]

Would that show serverA the NAted IP address (192.168.10.xxx) or serverB ?



Message: 14
Date: Tue, 8 Aug 2006 14:16:10 +0200
From: Jon Sch?pzinsky <[EMAIL PROTECTED]>
Subject: SV: [asterisk-users] IAX trunk behing NAT with dynamic IP
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Message-ID:
<[EMAIL PROTECTED]>
Content-Type: text/plain;   charset="iso-8859-1"

Set the host=dynamic on serverA, and let the serverB register with serverA

Jon

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Re: [asterisk-users] IAX trunk behing NAT with dynamic IP

[DM]

If that doesn't work, the OP can set up a cron job to do a IAX2 Reload
every 5 minutes like I did.  I have the exact same problem.

I hope the OP reports back if that works, because I would rather add
the host statement than run a cron job every 5 minutes.

On 8/8/06, Jon Schøpzinsky <[EMAIL PROTECTED]> wrote:

Set the host=dynamic on serverA, and let the serverB register with serverA

Jon


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SV: [asterisk-users] IAX trunk behing NAT with dynamic IP

[Jon Schøpzinsky]

Set the host=dynamic on serverA, and let the serverB register with serverA

Jon

-Oprindelig meddelelse-
Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Andre Courchesne - 
Consultant
Sendt: 8. august 2006 14:09
Til: asterisk-users@lists.digium.com
Emne: [asterisk-users] IAX trunk behing NAT with dynamic IP

Hi,

  Ok, I got a working setup where the * server having the telephony card 
has a fixed internet IP address (serverA). I am using an IAX trunk from 
this server to an other one which has a dynamic IP address and is behind 
a NAT firewall (serverB).

  Everything works fine untill serverB internet IP address changes. My 
host line is set to a dynamic DNS entry (with zoneedit.com)

  How can I resolve this so that serverA "see" the IP address change of 
serverB ?

Andre Courchesne - Consultant
http://www.net-forces.com

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[asterisk-users] IAX trunk behing NAT with dynamic IP

[Andre Courchesne - Consultant]

Hi,

 Ok, I got a working setup where the * server having the telephony card 
has a fixed internet IP address (serverA). I am using an IAX trunk from 
this server to an other one which has a dynamic IP address and is behind 
a NAT firewall (serverB).


 Everything works fine untill serverB internet IP address changes. My 
host line is set to a dynamic DNS entry (with zoneedit.com)


 How can I resolve this so that serverA "see" the IP address change of 
serverB ?


Andre Courchesne - Consultant
http://www.net-forces.com

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Re: [Asterisk-Users] IAX Trunk

[Luki]

Senario: If a call is initiated from Server 1 to Server 2,
a trunk is established.  While that call is progress another
call is established from Server 2 to Server 1.
Is a new trunk created, or is the same one used?


I had exactly the same question and looked into this. If I remember
correctly, a new trunk would be created in this case. Only case in the
same direction are trunked together (i.e. if the second call would be
from server 1 -> 2 it would use the existing trunk). You can verify
yourself by watching the network traffic with tcpdump though. The
packet size should give you the answer.

--Luki
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[Asterisk-Users] IAX Trunk

[Forrest Beck]

I have two servers setup on different locations.  They are
both setup as peers and users to each other.

 

Server 1 iax.conf:

[ms-to-us]

type=user

username=ms-to-us

secret=ms-to-us

context=upperschool

[us-to-ms]

type=peer

username=us-to-ms

secret=us-to-ms

host=10.11.1.112

trunk=yes

 

Server 2 iax.conf:

[ms-to-us]

type=peer

username=ms-to-us

secret=ms-to-us

host=10.11.1.111

trunk=yes

[us-to-ms]

type=user

username=us-to-ms

secret=us-to-ms

context=middleschool

 

All works great.  But I am curious.  

 

Senario: If a call is initiated from Server 1 to Server 2, a
trunk is established.  While that call is progress another call is established from
Server 2 to Server 1.  Is a new trunk created, or is the same one used?






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RE: [Asterisk-Users] IAX trunk monitoring

[Steve Dolloff]

Creating a separate user and peer does allow me to call over the trunk
using this format:
exten => 8475551212,3,Dial(IAX/voip2/${EXTEN},31,r)
(this must be a bug.  Why would the same format not work for a friend?)

It does not solve my original problem of failing the call if the trunk
is down.  Calls now are always sent to the voip2 iax user regardless of
whether that user is connected.

Also, voip2 was created as a user and voip2peer was created as a peer.
If I use:
exten => 8475551212,3,Dial(IAX/voip2peer/${EXTEN},31,r)
or
exten => 8475551212,3,Dial(IAX/[EMAIL PROTECTED]/${EXTEN},31,r)

the call fails as unavailable regardless of whether or not the other
server is running.

The registry information looks wrong too:

voip1*CLI> iax show registry
Host  UsernamePerceived Refresh  State
209.242.15.34:5036voip1peer60  Request
Sent
209.242.15.34:5036voip160
Rejected



> -Original Message-
> From: Philipp von Klitzing [mailto:[EMAIL PROTECTED]
> aachen.de]
> Sent: Friday, November 14, 2003 4:08 AM
> To: Steve Dolloff
> Subject: RE: [Asterisk-Users] IAX trunk monitoring
> 
> You might want to try this:
> 
> split the entries for voip1 and voip2, i.e. instead of type=friend
have
> an entry for type=peer and type=user for each of the two machines.
> 
> Greetings, Philipp
> 
> 
I have modified the configuration for dynamic host and registered each
server with the other.  The iax show users now lists the other iax
device as registered vs unavailable, but I still don't know how to keep
it from calling if the device becomes unavailable.

I changed the extensions file to:

exten => 8475551212,3,Dial(IAX/[EMAIL PROTECTED]/${EXTEN},,r)

also tried:

exten => 8475551212,3,Dial(IAX/voip2/${EXTEN},,r)

since voip2 is now a registered user, but it is not trying to call the
other server.

If I leave it as [EMAIL PROTECTED] it works as long as the trunk is up
but it doesn't check to make sure before sending the call there.

Any suggestions?

------------
---

To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] IAX trunk monitoring

I have an issue where * tries to route a call over IAX to another server
even if the server is down.  I have included the relevant entries from
my iax.conf, extensions.conf, and some debug output.  If someone could
tell me what I have configured incorrectly, I would appreciate it.  

Thanks,

Stephen

---iax.conf on voip2--
[voip1]
type=friend
username=voip1
host=x.x.x.x (ip address of voip1)
port=5036
mask=255.255.255.255
qualify=yes ; Make sure this peer is alive
trunk=no
context=IAX

--iax.conf on voip1---
[voip2]
type=friend
username=voip2
host=x.x.x.x (ip address of voip2)
port=5036
mask=255.255.255.255
qualify=yes ; Make sure this peer is alive
trunk=no
context=IAX

-extensions.conf on voip1
exten => 8475551212,3,Dial(IAX/[EMAIL PROTECTED]/${EXTEN},,r)

-extensions.conf on voip2
exten => 8475551212,3,Dial(IAX/[EMAIL PROTECTED]/${EXTEN},,r)

voip1*CLI> iax show users
Username Secret   Authen   Def.Context  A/C
voip2 md5,plaintextIAX  No

voip1*CLI> iax show peers
Name/UsernameHost Mask Port  Status
voip2/voip2  x.x.x.x   (S)  255.255.255.255  5036  UNREACHABLE

voip2*CLI> iax show users
Username Secret   Authen   Def.Context  A/C
voip1 md5,plaintextIAX  No

voip2*CLI> iax show peers
Name/UsernameHost Mask Port  Status
voip1/voip1  x.x.x.x  (S)  255.255.255.255  5036  UNREACHABLE

voip1*CLI> iax debug
Rx-Frame Retry[N/A] -- Seqno: 00  Type: IAX Subclass: POKE
Tx-Frame Retry[-01] -- Seqno: 00  Type: IAX Subclass: ACK
Tx-Frame Retry[000] -- Seqno: 00  Type: IAX Subclass: PONG
Rx-Frame Retry[N/A] -- Seqno: 01  Type: IAX Subclass: ACK
Tx-Frame Retry[001] -- Seqno: 00  Type: IAX Subclass: PONG
Rx-Frame Retry[N/A] -- Seqno: 00  Type: IAX Subclass: INVAL
Tx-Frame Retry[000] -- Seqno: 00  Type: IAX Subclass: POKE
Tx-Frame Retry[001] -- Seqno: 00  Type: IAX Subclass: POKE
Tx-Frame Retry[000] -- Seqno: 00  Type: IAX Subclass: POKE
Tx-Frame Retry[001] -- Seqno: 00  Type: IAX Subclass: POKE
Tx-Frame Retry[000] -- Seqno: 00  Type: IAX Subclass: POKE
Tx-Frame Retry[001] -- Seqno: 00  Type: IAX Subclass: POKE
Tx-Frame Retry[000] -- Seqno: 00  Type: IAX Subclass: POKE
Tx-Frame Retry[001] -- Seqno: 00  Type: IAX Subclass: POKE
voip1*CLI>


voip2*CLI> iax debug
IAX Debugging Enabled
Tx-Frame Retry[000] -- Seqno: 00  Type: IAX Subclass: POKE
Rx-Frame Retry[N/A] -- Seqno: 00  Type: IAX Subclass: POKE
Tx-Fr

RE: [Asterisk-Users] IAX trunk monitoring

[Steve Dolloff]

I have modified the configuration for dynamic host and registered each
server with the other.  The iax show users now lists the other iax
device as registered vs unavailable, but I still don't know how to keep
it from calling if the device becomes unavailable.

I changed the extensions file to:

exten => 8475551212,3,Dial(IAX/[EMAIL PROTECTED]/${EXTEN},,r)

also tried:

exten => 8475551212,3,Dial(IAX/voip2/${EXTEN},,r)

since voip2 is now a registered user, but it is not trying to call the
other server.

If I leave it as [EMAIL PROTECTED] it works as long as the trunk is up
but it doesn't check to make sure before sending the call there.

Any suggestions?


---

To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] IAX trunk monitoring

I have an issue where * tries to route a call over IAX to another server
even if the server is down.  I have included the relevant entries from
my iax.conf, extensions.conf, and some debug output.  If someone could
tell me what I have configured incorrectly, I would appreciate it.  

Thanks,

Stephen

---iax.conf on voip2--
[voip1]
type=friend
username=voip1
host=x.x.x.x (ip address of voip1)
port=5036
mask=255.255.255.255
qualify=yes ; Make sure this peer is alive
trunk=no
context=IAX

--iax.conf on voip1---
[voip2]
type=friend
username=voip2
host=x.x.x.x (ip address of voip2)
port=5036
mask=255.255.255.255
qualify=yes ; Make sure this peer is alive
trunk=no
context=IAX

-extensions.conf on voip1
exten => 8475551212,3,Dial(IAX/[EMAIL PROTECTED]/${EXTEN},,r)

-extensions.conf on voip2
exten => 8475551212,3,Dial(IAX/[EMAIL PROTECTED]/${EXTEN},,r)

voip1*CLI> iax show users
Username Secret   Authen   Def.Context  A/C
voip2 md5,plaintextIAX  No

voip1*CLI> iax show peers
Name/UsernameHost Mask Port  Status
voip2/voip2  x.x.x.x   (S)  255.255.255.255  5036  UNREACHABLE

voip2*CLI> iax show users
Username Secret   Authen   Def.Context  A/C
voip1 md5,plaintextIAX  No

voip2*CLI> iax show peers
Name/UsernameHost Mask Port  Status
voip1/voip1  x.x.x.x  (S)  255.255.255.255  5036  UNREACHABLE

voip1*CLI> iax debug
Rx-Frame Retry[N/A] -- Seqno: 00  Type: IAX Subclass: POKE
Tx-Frame Retry[-01] -- Seqno: 00  Type: IAX Subclass: ACK
Tx-Frame Retry[000] -- Seqno: 00  Type: IAX Subclass: PONG
Rx-Frame Retry[N/A] -- Seqno: 01  Type: IAX Subclass: ACK
Tx-Frame Retry[001] -- Seqno: 00  Type: IAX Subclass: PONG
Rx-Frame Retry[N/A] -- Seqno: 00  Type: IAX Subclass: INVAL
Tx-Frame Retry[000] -- Seqno: 00  Type: IAX Subclass: POKE
Tx-Frame Retry[001] -- Seqno: 00  Type: IAX Subclass: POKE
Tx-Frame Retry[000] -- Seqno: 00  Type: IAX Subclass: POKE
Tx-Frame Retry[001] -- Seqno: 00  Type: IAX Subclass: POKE
Tx-Frame Retry[000] -- Seqno: 00  Type: IAX Subclass: POKE
Tx-Frame Retry[001] -- Seqno: 00  Type: IAX Subclass: POKE
Tx-Frame Retry[000] -- Seqno: 00  Type: IAX Subclass: POKE
Tx-Frame Retry[001] -- Seqno: 00  Type: IAX Subclass: POKE
voip1*CLI>


voip2*CLI> iax debug
IAX Debugging Enabled
Tx-Frame Retry[000] -- Seqno: 00  Type: IAX Subclass: POKE
Rx-Frame Retry[N/A] -- Seqno: 00  Type: IAX Subclass: POKE
Tx-Frame Retry[-01] -- Seqno: 00  Type: IAX Subclass: ACK
Tx-Frame Retry[000] -- Seqno: 00  Type: IAX Subclass: PONG
Rx-Frame Retry[N/A] -- Seqno: 01  Type: IAX Subclass: ACK
Tx-Frame Retry[001] -- Seqno: 00  Type: IAX Subclass: PONG
Rx-Frame Retry[N/A] -- Seqno: 00  Type: IAX Subclass: INVAL
Tx-Frame Retry[000] -- Seqno: 00  Type: IAX Subclass: POKE
Tx-Frame Retry[001] -- Seqno: 00  Type: IAX Subclass: POKE
Tx-Frame Retry[000] -- Seqno: 00  Type: IAX Subclass: POKE
Tx-Frame Retry[001] -- Seqno: 00  Type: IAX Subclass: POKE
voip2*CLI>



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[Asterisk-Users] IAX trunk monitoring

[Steve Dolloff]

I have an issue where * tries to route a call over IAX to another server
even if the server is down.  I have included the relevant entries from
my iax.conf, extensions.conf, and some debug output.  If someone could
tell me what I have configured incorrectly, I would appreciate it.  

Thanks,

Stephen

---iax.conf on voip2--
[voip1]
type=friend
username=voip1
host=x.x.x.x (ip address of voip1)
port=5036
mask=255.255.255.255
qualify=yes ; Make sure this peer is alive
trunk=no
context=IAX

--iax.conf on voip1---
[voip2]
type=friend
username=voip2
host=x.x.x.x (ip address of voip2)
port=5036
mask=255.255.255.255
qualify=yes ; Make sure this peer is alive
trunk=no
context=IAX

-extensions.conf on voip1
exten => 8475551212,3,Dial(IAX/[EMAIL PROTECTED]/${EXTEN},,r)

-extensions.conf on voip2
exten => 8475551212,3,Dial(IAX/[EMAIL PROTECTED]/${EXTEN},,r)

voip1*CLI> iax show users
Username Secret   Authen   Def.Context  A/C
voip2 md5,plaintextIAX  No

voip1*CLI> iax show peers
Name/UsernameHost Mask Port  Status
voip2/voip2  x.x.x.x   (S)  255.255.255.255  5036  UNREACHABLE

voip2*CLI> iax show users
Username Secret   Authen   Def.Context  A/C
voip1 md5,plaintextIAX  No

voip2*CLI> iax show peers
Name/UsernameHost Mask Port  Status
voip1/voip1  x.x.x.x  (S)  255.255.255.255  5036  UNREACHABLE

voip1*CLI> iax debug
Rx-Frame Retry[N/A] -- Seqno: 00  Type: IAX Subclass: POKE
Tx-Frame Retry[-01] -- Seqno: 00  Type: IAX Subclass: ACK
Tx-Frame Retry[000] -- Seqno: 00  Type: IAX Subclass: PONG
Rx-Frame Retry[N/A] -- Seqno: 01  Type: IAX Subclass: ACK
Tx-Frame Retry[001] -- Seqno: 00  Type: IAX Subclass: PONG
Rx-Frame Retry[N/A] -- Seqno: 00  Type: IAX Subclass: INVAL
Tx-Frame Retry[000] -- Seqno: 00  Type: IAX Subclass: POKE
Tx-Frame Retry[001] -- Seqno: 00  Type: IAX Subclass: POKE
Tx-Frame Retry[000] -- Seqno: 00  Type: IAX Subclass: POKE
Tx-Frame Retry[001] -- Seqno: 00  Type: IAX Subclass: POKE
Tx-Frame Retry[000] -- Seqno: 00  Type: IAX Subclass: POKE
Tx-Frame Retry[001] -- Seqno: 00  Type: IAX Subclass: POKE
Tx-Frame Retry[000] -- Seqno: 00  Type: IAX Subclass: POKE
Tx-Frame Retry[001] -- Seqno: 00  Type: IAX Subclass: POKE
voip1*CLI>


voip2*CLI> iax debug
IAX Debugging Enabled
Tx-Frame Retry[000] -- Seqno: 00  Type: IAX Subclass: POKE
Rx-Frame Retry[N/A] -- Seqno: 00  Type: IAX Subclass: POKE
Tx-Frame Retry[-01] -- Seqno: 00  Type: IAX Subclass: ACK
Tx-Frame Retry[000] -- Seqno: 00  Type: IAX Subclass: PONG
Rx-Frame Retry[N/A] -- Seqno: 01  Type: IAX Subclass: ACK
Tx-Frame Retry[001] -- Seqno: 00  Type: IAX Subclass: PONG
Rx-Frame Retry[N/A] -- Seqno: 00  Type: IAX Subclass: INVAL
Tx-Frame Retry[000] -- Seqno: 00  Type: IAX Subclass: POKE
Tx-Frame Retry[001] -- Seqno: 00  Type: IAX Subclass: POKE
Tx-Frame Retry[000] -- Seqno: 00  Type: IAX Subclass: POKE
Tx-Frame Retry[001] -- Seqno: 00  Type: IAX Subclass: POKE
voip2*CLI>



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