Re: [asterisk-users] IAX Problem
Thank you for your interest in my question and quick response. I am
relatively new to Asterisk, so I have a few specific questions regarding
your suggestions.
Then I will post to the list with a more meaningful subject and results.
On 4/6/2010 10:31 AM, Steve Edwards wrote:
> On Tue, 6 Apr 2010, bob gailer wrote:
>
>
>> I have2 Trixbox Servers. Each has an IAX trunks to the other. One works
>> the other fails:
>>
>> -- Executing [...@macro-dialout-trunk:19] Dial("SIP/526-09eec7c8",
>> "IAX2/InterOffice/210,300,tr") in new stack
>> -- Called InterOffice/210
>> -- Hungup 'IAX2/InterOffice-7578'
>>== Everyone is busy/congested at this time (1:0/0/1)
>>
>> The only difference I am aware of is that one server has a public IP
>> address, the other is behind a NAT.
>>
>> The trunk from the server with the public address works fine.
>>
>> I added nat=yes to the other's peer details - did not help.
>>
>> What should I do?
>>
> 1) Set "auth=plaintext" (only for the duration of the debug session)
Where / how do I do that. Is that in the trunk peer settings?
> enable iax2 debugging on the CLI.
>
I did that; I now get without any action on my part:
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW
Timestamp: 1ms SCall: 00714 DCall: 0 [67.228.218.114:4569]
Whence cometh those lines?
When I call I get:
VERSION : 2
CALLED NUMBER : 210
CODEC_PREFS : (ulaw|alaw|gsm)
CALLING NUMBER : 526
CALLING PRESNTN : 0
CALLING TYPEOFN : 0
CALLING TRANSIT : 0
CALLING NAME: Bob's Office
LANGUAGE: en
FORMAT : 4
CAPABILITY : 14
ADSICPE : 2
DATE TIME : 2010-04-06 10:54:08
> You should see the "NEW" request and the passwords in plaintext.
>
What passwords? Where / how does one specify passwords?
> Verify the username, password, context, and extension all exist.
>
> 2) Reply with the sanitized console output.
>
What do you mean by sanitized. I assume you want just the relevant
output. True?
Thanks in advance.
--
Bob Gailer
919-636-4239
Chapel Hill NC
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Re: [asterisk-users] IAX Problem
On Tue, 6 Apr 2010, bob gailer wrote:
> I have2 Trixbox Servers. Each has an IAX trunks to the other. One works
> the other fails:
>
> -- Executing [...@macro-dialout-trunk:19] Dial("SIP/526-09eec7c8",
> "IAX2/InterOffice/210,300,tr") in new stack
> -- Called InterOffice/210
> -- Hungup 'IAX2/InterOffice-7578'
> == Everyone is busy/congested at this time (1:0/0/1)
>
> The only difference I am aware of is that one server has a public IP
> address, the other is behind a NAT.
>
> The trunk from the server with the public address works fine.
>
> I added nat=yes to the other's peer details - did not help.
>
> What should I do?
0) Use a more descriptive subject.
1) Set "auth=plaintext" (only for the duration of the debug session) and
enable iax2 debugging on the CLI.
You should see the "NEW" request and the passwords in plaintext.
Verify the username, password, context, and extension all exist.
2) Reply with the sanitized console output.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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[asterisk-users] IAX Problem
I have2 Trixbox Servers. Each has an IAX trunks to the other. One works
the other fails:
-- Executing [...@macro-dialout-trunk:19] Dial("SIP/526-09eec7c8",
"IAX2/InterOffice/210,300,tr") in new stack
-- Called InterOffice/210
-- Hungup 'IAX2/InterOffice-7578'
== Everyone is busy/congested at this time (1:0/0/1)
The only difference I am aware of is that one server has a public IP
address, the other is behind a NAT.
The trunk from the server with the public address works fine.
I added nat=yes to the other's peer details - did not help.
What should I do?
--
Bob Gailer
919-636-4239
Chapel Hill NC
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
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asterisk-users mailing list
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Re: [asterisk-users] IAX problem through intermediate asterisk box
On Thu, 26 Mar 2009, Andrew Hakman wrote:
> So no one else has a problem routing IAX traffic through an
> intermediate Asterisk server? Does anyone else use Asterisk in such a
> configuration?
I do. Not had a problem apart from when Digium break the protocol.
1.2 -> Interweb -> 1.2 -> Interweb -> 1.2
Also now have 1.4 in the middle too.
I'm moving to SIP though because the last leg is stuck on 1.2 and carrying
the traffic is not something I want to keep on doing. (No "reinvite" in
IAX in 1.2)
Gordon
> On Thu, Mar 26, 2009 at 2:45 AM, Andrew Hakman
> wrote:
>> I'm having a problem with IAX running through an intermediate asterisk
>> box. Perhaps a small diagram will explain the situation better:
>>
>> *A --- [cloud (public internet)] --- *B [cloud
>> (private network)]--- *C
>>
>> Asterisk server's A, B, and C, are all connected together with IAX
>> All asterisk servers are 1.6.0.6
>> Server A and B are geographically close, but connected over the public
>> internet.
>> Server B and C are geographically far, but connected over a private network.
>> (the latency between A and B, and B and C are roughly equal)
>>
>> Each server has at least 1 phone hanging off of it, with A and C
>> having most of the phones (B only has a couple).
>> A's extensions are all 1XXX, B's are 2XXX, and C's are 3XXX
>>
>> Phoning from A to B (or vice versa) works well, as does phoning from B
>> to C (and vice versa). Calls can be placed for an indefinite amount of
>> time and everything works great.
>>
>> The problem arises when phoning from A through B to C (or vice versa).
>> For the first small amount of time (which can vary on a call to call
>> basis, and lasts from 0 seconds to 3 minutes or so) everything is
>> fine. After this, the audio in both directions gets garbled, and
>> starts arriving in spurts. Once this happens, it continues forever.
>> The audio never returns to normal no matter how long you wait.
>>
>> A to B uses IAX with trunking. B to C is not using trunking
>> (dahdi_dummy is not working well on C for some reason - the module
>> loads, but no /dev/dahdi is ever created). The same behavior happens
>> when A to B is not using trunking either.
>>
>> Usually only 1 call is being placed at a time. An interesting thing
>> happens when 2 testcalls are in progress at the same time though. If
>> there's a call from A to B, and a call from A to C is made, once the
>> call from A to C becomes garbled, so does the A to B call. When the A
>> to C call is ended, the A to B call clears up. Ending the A to B call
>> first does not improve the A to C call.
>>
>> The dialplans are setup so each server passes all non-local extensions
>> to it's neighbor.
>>
>> Hence, for A, the relevant part of the dialplan is
>>
>> exten => _2XXX,1,Verbose(1|Extension 2xxx)
>> exten => _2XXX,n,Dial(IAX2/asterisk_B/${EXTEN})
>> exten => _2XXX,n,Hangup()
>>
>> exten => _3XXX,1,Verbose(1|Extension 3xxx)
>> exten => _3xxx,n,Dial(IAX2/asterisk_B/${EXTEN})
>> exten => _3xxx,n,Hangup()
>>
>> For B:
>>
>> exten => _1XXX,1,NoOp()
>> exten => _1XXX,n,Dial(IAX2/asterisk_A/${EXTEN})
>> exten => _1XXX,n,Hangup()
>>
>> exten => _3xxx,1,NoOp()
>> exten => _3xxx,n,Dial(IAX2/asterisk_C/${EXTEN})
>> exten => _3xxx,n,Hangup()
>>
>>
>> For C:
>> exten => _2XXX,1,Verbose(1|Extension 2xxx)
>> exten => _2XXX,n,Dial(IAX2/asterisk_B/${EXTEN})
>> exten => _2XXX,n,Hangup()
>>
>> exten => _1XXX,1,Verbose(1|Extension 1xxx)
>> exten => _1XXX,n,Dial(IAX2/asterisk_B/${EXTEN})
>> exten => _1XXX,n,Hangup()
>>
>> Is this the proper way to set such a configuration up? Is there a
>> better way to call from A through B to C that would work better?
>> Anyone else experience total audio breakup after a while with a
>> similar arrangement? Why does it work initially for up to about 3
>> minutes, then completely fall apart?
>>
>> Thanks,
>> Andrew
>>
>
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Re: [asterisk-users] IAX problem through intermediate asterisk box
I initially had no trunking anywhere, and had the same behavior. I
thought trunking would help, but I can't figure out why the /dev/dahdi
device doesn't get created on C. The dahdi tools / modules don't seem
to have much error / debugging info available, or if they do, I sure
can't find it anywhere obvious.
Andrew
On Thu, Mar 26, 2009 at 11:39 PM, Brandon B. wrote:
> Here's my troubleshooting help -- since the problem sounds like a timing
> issue and part of the call is being trunked, then fix your timing problem,
> or remove the trunking from A and B then see if the problem goes away.
>
> On Thu, Mar 26, 2009 at 10:50 PM, Andrew Hakman
> wrote:
>>
>> So no one else has a problem routing IAX traffic through an
>> intermediate Asterisk server? Does anyone else use Asterisk in such a
>> configuration?
>>
>> On Thu, Mar 26, 2009 at 2:45 AM, Andrew Hakman
>> wrote:
>> > I'm having a problem with IAX running through an intermediate asterisk
>> > box. Perhaps a small diagram will explain the situation better:
>> >
>> > *A --- [cloud (public internet)] --- *B [cloud
>> > (private network)]--- *C
>> >
>> > Asterisk server's A, B, and C, are all connected together with IAX
>> > All asterisk servers are 1.6.0.6
>> > Server A and B are geographically close, but connected over the public
>> > internet.
>> > Server B and C are geographically far, but connected over a private
>> > network.
>> > (the latency between A and B, and B and C are roughly equal)
>> >
>> > Each server has at least 1 phone hanging off of it, with A and C
>> > having most of the phones (B only has a couple).
>> > A's extensions are all 1XXX, B's are 2XXX, and C's are 3XXX
>> >
>> > Phoning from A to B (or vice versa) works well, as does phoning from B
>> > to C (and vice versa). Calls can be placed for an indefinite amount of
>> > time and everything works great.
>> >
>> > The problem arises when phoning from A through B to C (or vice versa).
>> > For the first small amount of time (which can vary on a call to call
>> > basis, and lasts from 0 seconds to 3 minutes or so) everything is
>> > fine. After this, the audio in both directions gets garbled, and
>> > starts arriving in spurts. Once this happens, it continues forever.
>> > The audio never returns to normal no matter how long you wait.
>> >
>> > A to B uses IAX with trunking. B to C is not using trunking
>> > (dahdi_dummy is not working well on C for some reason - the module
>> > loads, but no /dev/dahdi is ever created). The same behavior happens
>> > when A to B is not using trunking either.
>> >
>> > Usually only 1 call is being placed at a time. An interesting thing
>> > happens when 2 testcalls are in progress at the same time though. If
>> > there's a call from A to B, and a call from A to C is made, once the
>> > call from A to C becomes garbled, so does the A to B call. When the A
>> > to C call is ended, the A to B call clears up. Ending the A to B call
>> > first does not improve the A to C call.
>> >
>> > The dialplans are setup so each server passes all non-local extensions
>> > to it's neighbor.
>> >
>> > Hence, for A, the relevant part of the dialplan is
>> >
>> > exten => _2XXX,1,Verbose(1|Extension 2xxx)
>> > exten => _2XXX,n,Dial(IAX2/asterisk_B/${EXTEN})
>> > exten => _2XXX,n,Hangup()
>> >
>> > exten => _3XXX,1,Verbose(1|Extension 3xxx)
>> > exten => _3xxx,n,Dial(IAX2/asterisk_B/${EXTEN})
>> > exten => _3xxx,n,Hangup()
>> >
>> > For B:
>> >
>> > exten => _1XXX,1,NoOp()
>> > exten => _1XXX,n,Dial(IAX2/asterisk_A/${EXTEN})
>> > exten => _1XXX,n,Hangup()
>> >
>> > exten => _3xxx,1,NoOp()
>> > exten => _3xxx,n,Dial(IAX2/asterisk_C/${EXTEN})
>> > exten => _3xxx,n,Hangup()
>> >
>> >
>> > For C:
>> > exten => _2XXX,1,Verbose(1|Extension 2xxx)
>> > exten => _2XXX,n,Dial(IAX2/asterisk_B/${EXTEN})
>> > exten => _2XXX,n,Hangup()
>> >
>> > exten => _1XXX,1,Verbose(1|Extension 1xxx)
>> > exten => _1XXX,n,Dial(IAX2/asterisk_B/${EXTEN})
>> > exten => _1XXX,n,Hangup()
>> >
>> > Is this the proper way to set such a configuration up? Is there a
>> > better way to call from A through B to C that would work better?
>> > Anyone else experience total audio breakup after a while with a
>> > similar arrangement? Why does it work initially for up to about 3
>> > minutes, then completely fall apart?
>> >
>> > Thanks,
>> > Andrew
>> >
>>
>> ___
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
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as
Re: [asterisk-users] IAX problem through intermediate asterisk box
Here's my troubleshooting help -- since the problem sounds like a timing
issue and part of the call is being trunked, then fix your timing problem,
or remove the trunking from A and B then see if the problem goes away.
On Thu, Mar 26, 2009 at 10:50 PM, Andrew Hakman wrote:
> So no one else has a problem routing IAX traffic through an
> intermediate Asterisk server? Does anyone else use Asterisk in such a
> configuration?
>
> On Thu, Mar 26, 2009 at 2:45 AM, Andrew Hakman
> wrote:
> > I'm having a problem with IAX running through an intermediate asterisk
> > box. Perhaps a small diagram will explain the situation better:
> >
> > *A --- [cloud (public internet)] --- *B [cloud
> > (private network)]--- *C
> >
> > Asterisk server's A, B, and C, are all connected together with IAX
> > All asterisk servers are 1.6.0.6
> > Server A and B are geographically close, but connected over the public
> internet.
> > Server B and C are geographically far, but connected over a private
> network.
> > (the latency between A and B, and B and C are roughly equal)
> >
> > Each server has at least 1 phone hanging off of it, with A and C
> > having most of the phones (B only has a couple).
> > A's extensions are all 1XXX, B's are 2XXX, and C's are 3XXX
> >
> > Phoning from A to B (or vice versa) works well, as does phoning from B
> > to C (and vice versa). Calls can be placed for an indefinite amount of
> > time and everything works great.
> >
> > The problem arises when phoning from A through B to C (or vice versa).
> > For the first small amount of time (which can vary on a call to call
> > basis, and lasts from 0 seconds to 3 minutes or so) everything is
> > fine. After this, the audio in both directions gets garbled, and
> > starts arriving in spurts. Once this happens, it continues forever.
> > The audio never returns to normal no matter how long you wait.
> >
> > A to B uses IAX with trunking. B to C is not using trunking
> > (dahdi_dummy is not working well on C for some reason - the module
> > loads, but no /dev/dahdi is ever created). The same behavior happens
> > when A to B is not using trunking either.
> >
> > Usually only 1 call is being placed at a time. An interesting thing
> > happens when 2 testcalls are in progress at the same time though. If
> > there's a call from A to B, and a call from A to C is made, once the
> > call from A to C becomes garbled, so does the A to B call. When the A
> > to C call is ended, the A to B call clears up. Ending the A to B call
> > first does not improve the A to C call.
> >
> > The dialplans are setup so each server passes all non-local extensions
> > to it's neighbor.
> >
> > Hence, for A, the relevant part of the dialplan is
> >
> > exten => _2XXX,1,Verbose(1|Extension 2xxx)
> > exten => _2XXX,n,Dial(IAX2/asterisk_B/${EXTEN})
> > exten => _2XXX,n,Hangup()
> >
> > exten => _3XXX,1,Verbose(1|Extension 3xxx)
> > exten => _3xxx,n,Dial(IAX2/asterisk_B/${EXTEN})
> > exten => _3xxx,n,Hangup()
> >
> > For B:
> >
> > exten => _1XXX,1,NoOp()
> > exten => _1XXX,n,Dial(IAX2/asterisk_A/${EXTEN})
> > exten => _1XXX,n,Hangup()
> >
> > exten => _3xxx,1,NoOp()
> > exten => _3xxx,n,Dial(IAX2/asterisk_C/${EXTEN})
> > exten => _3xxx,n,Hangup()
> >
> >
> > For C:
> > exten => _2XXX,1,Verbose(1|Extension 2xxx)
> > exten => _2XXX,n,Dial(IAX2/asterisk_B/${EXTEN})
> > exten => _2XXX,n,Hangup()
> >
> > exten => _1XXX,1,Verbose(1|Extension 1xxx)
> > exten => _1XXX,n,Dial(IAX2/asterisk_B/${EXTEN})
> > exten => _1XXX,n,Hangup()
> >
> > Is this the proper way to set such a configuration up? Is there a
> > better way to call from A through B to C that would work better?
> > Anyone else experience total audio breakup after a while with a
> > similar arrangement? Why does it work initially for up to about 3
> > minutes, then completely fall apart?
> >
> > Thanks,
> > Andrew
> >
>
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>
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Re: [asterisk-users] IAX problem through intermediate asterisk box
I'll have to get some VPN's setup, but I will give it a try with SIP.
Thanks for the input - you saved me building 2 more asterisk servers
for testing this issue locally (rather than across 3 networks).
Andrew
On Thu, Mar 26, 2009 at 11:12 PM, Steve Totaro
wrote:
> On Fri, Mar 27, 2009 at 12:50 AM, Andrew Hakman
> wrote:
>> So no one else has a problem routing IAX traffic through an
>> intermediate Asterisk server? Does anyone else use Asterisk in such a
>> configuration?
>>
>> On Thu, Mar 26, 2009 at 2:45 AM, Andrew Hakman
>> wrote:
>>> I'm having a problem with IAX running through an intermediate asterisk
>>> box. Perhaps a small diagram will explain the situation better:
>>>
>>> *A --- [cloud (public internet)] --- *B [cloud
>>> (private network)]--- *C
>>>
>>> Asterisk server's A, B, and C, are all connected together with IAX
>>> All asterisk servers are 1.6.0.6
>>> Server A and B are geographically close, but connected over the public
>>> internet.
>>> Server B and C are geographically far, but connected over a private network.
>>> (the latency between A and B, and B and C are roughly equal)
>>>
>>> Each server has at least 1 phone hanging off of it, with A and C
>>> having most of the phones (B only has a couple).
>>> A's extensions are all 1XXX, B's are 2XXX, and C's are 3XXX
>>>
>>> Phoning from A to B (or vice versa) works well, as does phoning from B
>>> to C (and vice versa). Calls can be placed for an indefinite amount of
>>> time and everything works great.
>>>
>>> The problem arises when phoning from A through B to C (or vice versa).
>>> For the first small amount of time (which can vary on a call to call
>>> basis, and lasts from 0 seconds to 3 minutes or so) everything is
>>> fine. After this, the audio in both directions gets garbled, and
>>> starts arriving in spurts. Once this happens, it continues forever.
>>> The audio never returns to normal no matter how long you wait.
>>>
>>> A to B uses IAX with trunking. B to C is not using trunking
>>> (dahdi_dummy is not working well on C for some reason - the module
>>> loads, but no /dev/dahdi is ever created). The same behavior happens
>>> when A to B is not using trunking either.
>>>
>>> Usually only 1 call is being placed at a time. An interesting thing
>>> happens when 2 testcalls are in progress at the same time though. If
>>> there's a call from A to B, and a call from A to C is made, once the
>>> call from A to C becomes garbled, so does the A to B call. When the A
>>> to C call is ended, the A to B call clears up. Ending the A to B call
>>> first does not improve the A to C call.
>>>
>>> The dialplans are setup so each server passes all non-local extensions
>>> to it's neighbor.
>>>
>>> Hence, for A, the relevant part of the dialplan is
>>>
>>> exten => _2XXX,1,Verbose(1|Extension 2xxx)
>>> exten => _2XXX,n,Dial(IAX2/asterisk_B/${EXTEN})
>>> exten => _2XXX,n,Hangup()
>>>
>>> exten => _3XXX,1,Verbose(1|Extension 3xxx)
>>> exten => _3xxx,n,Dial(IAX2/asterisk_B/${EXTEN})
>>> exten => _3xxx,n,Hangup()
>>>
>>> For B:
>>>
>>> exten => _1XXX,1,NoOp()
>>> exten => _1XXX,n,Dial(IAX2/asterisk_A/${EXTEN})
>>> exten => _1XXX,n,Hangup()
>>>
>>> exten => _3xxx,1,NoOp()
>>> exten => _3xxx,n,Dial(IAX2/asterisk_C/${EXTEN})
>>> exten => _3xxx,n,Hangup()
>>>
>>>
>>> For C:
>>> exten => _2XXX,1,Verbose(1|Extension 2xxx)
>>> exten => _2XXX,n,Dial(IAX2/asterisk_B/${EXTEN})
>>> exten => _2XXX,n,Hangup()
>>>
>>> exten => _1XXX,1,Verbose(1|Extension 1xxx)
>>> exten => _1XXX,n,Dial(IAX2/asterisk_B/${EXTEN})
>>> exten => _1XXX,n,Hangup()
>>>
>>> Is this the proper way to set such a configuration up? Is there a
>>> better way to call from A through B to C that would work better?
>>> Anyone else experience total audio breakup after a while with a
>>> similar arrangement? Why does it work initially for up to about 3
>>> minutes, then completely fall apart?
>>>
>>> Thanks,
>>> Andrew
>>>
>>
>> ___
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
> I have had, seen, or fixed this problem more times than I can count.
>
> Use SIP.
>
> IAX2 has been a common problem that I have fixed many many times for
> people over the years.
>
> OR, "The latest version should fix it", which is the Digium tagline on IAX2.
>
> Please report back your results if you do use SIP.
>
> --
> Thanks,
> Steve Totaro
> +18887771888 (Toll Free)
> +12409381212 (Cell)
>
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asterisk-users ma
Re: [asterisk-users] IAX problem through intermediate asterisk box
On Fri, Mar 27, 2009 at 12:50 AM, Andrew Hakman wrote:
> So no one else has a problem routing IAX traffic through an
> intermediate Asterisk server? Does anyone else use Asterisk in such a
> configuration?
>
> On Thu, Mar 26, 2009 at 2:45 AM, Andrew Hakman
> wrote:
>> I'm having a problem with IAX running through an intermediate asterisk
>> box. Perhaps a small diagram will explain the situation better:
>>
>> *A --- [cloud (public internet)] --- *B [cloud
>> (private network)]--- *C
>>
>> Asterisk server's A, B, and C, are all connected together with IAX
>> All asterisk servers are 1.6.0.6
>> Server A and B are geographically close, but connected over the public
>> internet.
>> Server B and C are geographically far, but connected over a private network.
>> (the latency between A and B, and B and C are roughly equal)
>>
>> Each server has at least 1 phone hanging off of it, with A and C
>> having most of the phones (B only has a couple).
>> A's extensions are all 1XXX, B's are 2XXX, and C's are 3XXX
>>
>> Phoning from A to B (or vice versa) works well, as does phoning from B
>> to C (and vice versa). Calls can be placed for an indefinite amount of
>> time and everything works great.
>>
>> The problem arises when phoning from A through B to C (or vice versa).
>> For the first small amount of time (which can vary on a call to call
>> basis, and lasts from 0 seconds to 3 minutes or so) everything is
>> fine. After this, the audio in both directions gets garbled, and
>> starts arriving in spurts. Once this happens, it continues forever.
>> The audio never returns to normal no matter how long you wait.
>>
>> A to B uses IAX with trunking. B to C is not using trunking
>> (dahdi_dummy is not working well on C for some reason - the module
>> loads, but no /dev/dahdi is ever created). The same behavior happens
>> when A to B is not using trunking either.
>>
>> Usually only 1 call is being placed at a time. An interesting thing
>> happens when 2 testcalls are in progress at the same time though. If
>> there's a call from A to B, and a call from A to C is made, once the
>> call from A to C becomes garbled, so does the A to B call. When the A
>> to C call is ended, the A to B call clears up. Ending the A to B call
>> first does not improve the A to C call.
>>
>> The dialplans are setup so each server passes all non-local extensions
>> to it's neighbor.
>>
>> Hence, for A, the relevant part of the dialplan is
>>
>> exten => _2XXX,1,Verbose(1|Extension 2xxx)
>> exten => _2XXX,n,Dial(IAX2/asterisk_B/${EXTEN})
>> exten => _2XXX,n,Hangup()
>>
>> exten => _3XXX,1,Verbose(1|Extension 3xxx)
>> exten => _3xxx,n,Dial(IAX2/asterisk_B/${EXTEN})
>> exten => _3xxx,n,Hangup()
>>
>> For B:
>>
>> exten => _1XXX,1,NoOp()
>> exten => _1XXX,n,Dial(IAX2/asterisk_A/${EXTEN})
>> exten => _1XXX,n,Hangup()
>>
>> exten => _3xxx,1,NoOp()
>> exten => _3xxx,n,Dial(IAX2/asterisk_C/${EXTEN})
>> exten => _3xxx,n,Hangup()
>>
>>
>> For C:
>> exten => _2XXX,1,Verbose(1|Extension 2xxx)
>> exten => _2XXX,n,Dial(IAX2/asterisk_B/${EXTEN})
>> exten => _2XXX,n,Hangup()
>>
>> exten => _1XXX,1,Verbose(1|Extension 1xxx)
>> exten => _1XXX,n,Dial(IAX2/asterisk_B/${EXTEN})
>> exten => _1XXX,n,Hangup()
>>
>> Is this the proper way to set such a configuration up? Is there a
>> better way to call from A through B to C that would work better?
>> Anyone else experience total audio breakup after a while with a
>> similar arrangement? Why does it work initially for up to about 3
>> minutes, then completely fall apart?
>>
>> Thanks,
>> Andrew
>>
>
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> asterisk-users mailing list
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>
I have had, seen, or fixed this problem more times than I can count.
Use SIP.
IAX2 has been a common problem that I have fixed many many times for
people over the years.
OR, "The latest version should fix it", which is the Digium tagline on IAX2.
Please report back your results if you do use SIP.
--
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
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Re: [asterisk-users] IAX problem through intermediate asterisk box
So no one else has a problem routing IAX traffic through an
intermediate Asterisk server? Does anyone else use Asterisk in such a
configuration?
On Thu, Mar 26, 2009 at 2:45 AM, Andrew Hakman wrote:
> I'm having a problem with IAX running through an intermediate asterisk
> box. Perhaps a small diagram will explain the situation better:
>
> *A --- [cloud (public internet)] --- *B [cloud
> (private network)]--- *C
>
> Asterisk server's A, B, and C, are all connected together with IAX
> All asterisk servers are 1.6.0.6
> Server A and B are geographically close, but connected over the public
> internet.
> Server B and C are geographically far, but connected over a private network.
> (the latency between A and B, and B and C are roughly equal)
>
> Each server has at least 1 phone hanging off of it, with A and C
> having most of the phones (B only has a couple).
> A's extensions are all 1XXX, B's are 2XXX, and C's are 3XXX
>
> Phoning from A to B (or vice versa) works well, as does phoning from B
> to C (and vice versa). Calls can be placed for an indefinite amount of
> time and everything works great.
>
> The problem arises when phoning from A through B to C (or vice versa).
> For the first small amount of time (which can vary on a call to call
> basis, and lasts from 0 seconds to 3 minutes or so) everything is
> fine. After this, the audio in both directions gets garbled, and
> starts arriving in spurts. Once this happens, it continues forever.
> The audio never returns to normal no matter how long you wait.
>
> A to B uses IAX with trunking. B to C is not using trunking
> (dahdi_dummy is not working well on C for some reason - the module
> loads, but no /dev/dahdi is ever created). The same behavior happens
> when A to B is not using trunking either.
>
> Usually only 1 call is being placed at a time. An interesting thing
> happens when 2 testcalls are in progress at the same time though. If
> there's a call from A to B, and a call from A to C is made, once the
> call from A to C becomes garbled, so does the A to B call. When the A
> to C call is ended, the A to B call clears up. Ending the A to B call
> first does not improve the A to C call.
>
> The dialplans are setup so each server passes all non-local extensions
> to it's neighbor.
>
> Hence, for A, the relevant part of the dialplan is
>
> exten => _2XXX,1,Verbose(1|Extension 2xxx)
> exten => _2XXX,n,Dial(IAX2/asterisk_B/${EXTEN})
> exten => _2XXX,n,Hangup()
>
> exten => _3XXX,1,Verbose(1|Extension 3xxx)
> exten => _3xxx,n,Dial(IAX2/asterisk_B/${EXTEN})
> exten => _3xxx,n,Hangup()
>
> For B:
>
> exten => _1XXX,1,NoOp()
> exten => _1XXX,n,Dial(IAX2/asterisk_A/${EXTEN})
> exten => _1XXX,n,Hangup()
>
> exten => _3xxx,1,NoOp()
> exten => _3xxx,n,Dial(IAX2/asterisk_C/${EXTEN})
> exten => _3xxx,n,Hangup()
>
>
> For C:
> exten => _2XXX,1,Verbose(1|Extension 2xxx)
> exten => _2XXX,n,Dial(IAX2/asterisk_B/${EXTEN})
> exten => _2XXX,n,Hangup()
>
> exten => _1XXX,1,Verbose(1|Extension 1xxx)
> exten => _1XXX,n,Dial(IAX2/asterisk_B/${EXTEN})
> exten => _1XXX,n,Hangup()
>
> Is this the proper way to set such a configuration up? Is there a
> better way to call from A through B to C that would work better?
> Anyone else experience total audio breakup after a while with a
> similar arrangement? Why does it work initially for up to about 3
> minutes, then completely fall apart?
>
> Thanks,
> Andrew
>
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[asterisk-users] IAX problem through intermediate asterisk box
I'm having a problem with IAX running through an intermediate asterisk
box. Perhaps a small diagram will explain the situation better:
*A --- [cloud (public internet)] --- *B [cloud
(private network)]--- *C
Asterisk server's A, B, and C, are all connected together with IAX
All asterisk servers are 1.6.0.6
Server A and B are geographically close, but connected over the public internet.
Server B and C are geographically far, but connected over a private network.
(the latency between A and B, and B and C are roughly equal)
Each server has at least 1 phone hanging off of it, with A and C
having most of the phones (B only has a couple).
A's extensions are all 1XXX, B's are 2XXX, and C's are 3XXX
Phoning from A to B (or vice versa) works well, as does phoning from B
to C (and vice versa). Calls can be placed for an indefinite amount of
time and everything works great.
The problem arises when phoning from A through B to C (or vice versa).
For the first small amount of time (which can vary on a call to call
basis, and lasts from 0 seconds to 3 minutes or so) everything is
fine. After this, the audio in both directions gets garbled, and
starts arriving in spurts. Once this happens, it continues forever.
The audio never returns to normal no matter how long you wait.
A to B uses IAX with trunking. B to C is not using trunking
(dahdi_dummy is not working well on C for some reason - the module
loads, but no /dev/dahdi is ever created). The same behavior happens
when A to B is not using trunking either.
Usually only 1 call is being placed at a time. An interesting thing
happens when 2 testcalls are in progress at the same time though. If
there's a call from A to B, and a call from A to C is made, once the
call from A to C becomes garbled, so does the A to B call. When the A
to C call is ended, the A to B call clears up. Ending the A to B call
first does not improve the A to C call.
The dialplans are setup so each server passes all non-local extensions
to it's neighbor.
Hence, for A, the relevant part of the dialplan is
exten => _2XXX,1,Verbose(1|Extension 2xxx)
exten => _2XXX,n,Dial(IAX2/asterisk_B/${EXTEN})
exten => _2XXX,n,Hangup()
exten => _3XXX,1,Verbose(1|Extension 3xxx)
exten => _3xxx,n,Dial(IAX2/asterisk_B/${EXTEN})
exten => _3xxx,n,Hangup()
For B:
exten => _1XXX,1,NoOp()
exten => _1XXX,n,Dial(IAX2/asterisk_A/${EXTEN})
exten => _1XXX,n,Hangup()
exten => _3xxx,1,NoOp()
exten => _3xxx,n,Dial(IAX2/asterisk_C/${EXTEN})
exten => _3xxx,n,Hangup()
For C:
exten => _2XXX,1,Verbose(1|Extension 2xxx)
exten => _2XXX,n,Dial(IAX2/asterisk_B/${EXTEN})
exten => _2XXX,n,Hangup()
exten => _1XXX,1,Verbose(1|Extension 1xxx)
exten => _1XXX,n,Dial(IAX2/asterisk_B/${EXTEN})
exten => _1XXX,n,Hangup()
Is this the proper way to set such a configuration up? Is there a
better way to call from A through B to C that would work better?
Anyone else experience total audio breakup after a while with a
similar arrangement? Why does it work initially for up to about 3
minutes, then completely fall apart?
Thanks,
Andrew
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RE: [asterisk-users] IAX problem
You are correct, I don't have entry for them to be able and authenticate! how should I define it? Should it be a peer or a user? Could you please add an example? Thanks a million... Itamar Lavender IT Manager Direct: +1 646 485 1828 __ Traiana, Inc 51 E. 42nd St., 10th Fl New York, NY 10017 Main: +1 212 404 1714 Fax:+1 656 536 4900 www.traiana.com The information contained in this e-mail is confidential and may be legally privileged. It is intended solely for the use of the individual or entity to whom it is addressed and others explicitly authorized to receive it. If you have received this e-mail in error, please destroy it and delete it from your computer. Any disclosure, copying or distribution of the information is strictly prohibited and may be unlawful. No responsibility can be accepted to any end users for any action taken on the basis of the information. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Panton Sent: Thursday, November 02, 2006 04:10 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IAX problem On 2 Nov 2006, at 02:38, Itamar Lavender wrote: > Hi All, > > > > I'm having problem with IAX, I'm trying to connect to speex.co.il > from asterisk using: > > register => username:[EMAIL PROTECTED] What does the rest of iax.conf look like ? Auth is a 2 way thing - you have sucessfully registered with them, but when they send you a 'new' your box fails to authenticate them as it can't find a matching user/friend entry in iax.conf. Tim Panton www.mexuar.com ___ --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 ___ --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] IAX problem
On 2 Nov 2006, at 02:38, Itamar Lavender wrote: Hi All, I'm having problem with IAX, I'm trying to connect to speex.co.il from asterisk using: register => username:[EMAIL PROTECTED] What does the rest of iax.conf look like ? Auth is a 2 way thing - you have sucessfully registered with them, but when they send you a 'new' your box fails to authenticate them as it can't find a matching user/friend entry in iax.conf. Tim Panton www.mexuar.com ___ --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
[asterisk-users] IAX problem
Hi All, I'm having problem with IAX, I'm trying to connect to speex.co.il from asterisk using: register => username:[EMAIL PROTECTED] and I cant get it to work. Maybe someone who already got this to work will help… When dialing my speex extension I see the next output from consol: IAX2 Debugging Enabled *CLI> Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00016ms SCall: 00101 DCall: 0 [212.29.199.163:4569] VERSION : 2 CALLED NUMBER : s CALLING NUMBER : 0546558780 CALLING NAME : 0546558780 LANGUAGE : en USERNAME : ilavender FORMAT : 2 CAPABILITY : 63490 ADSICPE : 2 DATE TIME : 2006-11-02 03:47:36 [Nov 2 04:26:27] NOTICE[5179]: chan_iax2.c:6788 socket_process: Rejected connect attempt from 212.29.199.163, who was trying to reach 's@' Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 00016ms SCall: 2 DCall: 00101 [212.29.199.163:4569] Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REJECT Timestamp: 00019ms SCall: 2 DCall: 00101 [212.29.199.163:4569] CAUSE : No authority found CAUSE CODE : 50 Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 00019ms SCall: 00101 DCall: 2 [212.29.199.163:4569] Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 00011ms SCall: 3 DCall: 0 [212.29.199.163:4569] USERNAME : ilavender REFRESH : 60 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGAUTH Timestamp: 00017ms SCall: 00170 DCall: 3 [212.29.199.163:4569] AUTHMETHODS : 3 CHALLENGE : 101355226 USERNAME : ilavender Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: REGREQ Timestamp: 00017ms SCall: 3 DCall: 00170 [212.29.199.163:4569] USERNAME : ilavender REFRESH : 60 MD5 RESULT : c5bf720545e784b2d6e28d9dfd734a11 Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: REGACK Timestamp: 00025ms SCall: 00170 DCall: 3 [212.29.199.163:4569] USERNAME : ilavender DATE TIME : 2006-11-02 03:47:48 REFRESH : 60 APPARENT ADDRES : IPV4 194.90.67.1:29628 CALLING NAME : ilavender Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00025ms SCall: 3 DCall: 00170 [212.29.199.163:4569] Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 6ms SCall: 1 DCall: 0 [212.29.199.163:4569] USERNAME : ilavender REFRESH : 60 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGAUTH Timestamp: 00018ms SCall: 00279 DCall: 1 [212.29.199.163:4569] AUTHMETHODS : 3 CHALLENGE : 151436469 USERNAME : ilavender Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: REGREQ Timestamp: 00016ms SCall: 1 DCall: 00279 [212.29.199.163:4569] USERNAME : ilavender REFRESH : 60 MD5 RESULT : b7bf725df38fa5ba42a11de765bc8f9c Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: REGACK Timestamp: 00034ms SCall: 00279 DCall: 1 [212.29.199.163:4569] USERNAME : ilavender DATE TIME : 2006-11-02 03:48:38 REFRESH : 60 APPARENT ADDRES : IPV4 194.90.67.1:29644 CALLING NAME : ilavender -- Registered IAX2 to '212.29.199.163', who sees us as 194.90.67.1:29644 with no messages waiting Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00034ms SCall: 1 DCall: 00279 [212.29.199.163:4569] Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 00014ms SCall: 2 DCall: 0 [212.29.199.163:4569] USERNAME : ilavender REFRESH : 60 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGAUTH Timestamp: 7ms SCall: 00234 DCall: 2 [212.29.199.163:4569] AUTHMETHODS : 3 CHALLENGE : 208585322 USERNAME : ilavender Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: REGREQ Timestamp: 00024ms SCall: 2 DCall: 00234 [212.29.199.163:4569] USERNAME : ilavender REFRESH : 60 MD5 RESULT : 0216f88113948a5950428a02437a5d1a Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: REGACK Timestamp: 00017ms SCall: 00234 DCall: 2 [212.29.199.163:4569] USERNAME : ilavender DATE TIME : 2006-11-02 03:49:28 REFRESH : 60 APPARENT ADDRES : IPV4 194.90.67.1
Re: [Asterisk-Users] IAX problem - Bug or Compatibility issue?
Nevermind. I think I found it. I didn't realise that Ethereal can decode IAX2 protocol. When IAX does not work, sometimes I got the following message as well: DEBUG[2213] chan_iax2.c: Immediately destroying 16384, having received INVAL What else can we say about that a part from my asterisk received frame with INVAL (Invalid call) or in other word incompatible frame? I also found a very quick and dirty workaround which is enough for me at the moment. - Original Message - From: Aryanto Rachmad To: asterisk-users@lists.digium.com Sent: Friday, December 30, 2005 6:38 PM Subject: [Asterisk-Users] IAX problem - Bug or Compatibility issue? Hello All, I am looking for more thorough debug than the one provided by the command "iax2 debug". Could anybody point me a good documentation about this? I have a issue with IAX connection. Sometimes it stucked. If so, I have to restart my asterisk through CLI command "restart now". Comparing the debug messages of working and non working sequences, I have noticed that when it does not work, the following debug messages are missing: Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 003 Type: CONTROL Subclass: (14?) Timestamp: 01581ms SCall: 00052 DCall: 16385 [213.61.187.157:4569]Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 003 Type: IAX Subclass: ACK Timestamp: 01581ms SCall: 16385 DCall: 00052 [213.61.187.157:4569] -- IAX2/sipdiscount_outbound-16385 is making progress passing it to Zap/1-1Dec 30 17:12:31 DEBUG[12600]: chan_zap.c:4791 zt_indicate: Requested indication 14 on channel Zap/1-1Dec 30 17:12:31 DEBUG[12600]: chan_zap.c:4857 zt_indicate: Received AST_CONTROL_PROGRESS on Zap/1-1Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 003 Type: VOICE Subclass: 4 Timestamp: 01732ms SCall: 00052 DCall: 16385 [213.61.187.157:4569]Dec 30 17:12:31 DEBUG[12569]: chan_iax2.c:6653 socket_read: Ooh, voice format changed to 4Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 004 Type: IAX Subclass: ACK Timestamp: 01732ms SCall: 16385 DCall: 00052 [213.61.187.157:4569] I have a few questions, especially about the following message: Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 003 Type: CONTROL Subclass: (14?) 1. Is the number 14 in (14?), in decimal or hexadecimal? 2. If that is in decimal, why is it not translated into its descriptions, i.e. Call Progress, according to the IAX2 protocol document I have (Internet-Draft, Expires: July 5, 2005). 3. Why is that number question marked? Is it because asterisk was not sure? 4. If asterisk was not sure, so sometimes it decodes the message sometimes it could not, is there any debug to confirm this? Or, am I looking at the wrong place? Which maybe the problem is so obvious and I missed that? I am running asterisk on IBM xSeries 330 with the following detail: CLI> show versionAsterisk 1.2.1 built by root @ atvie-asterisk on a i686 running Linux on 2005-12-28 07:52:36 UTC# uname -aLinux atvie-asterisk 2.6.14-1.1653_FC4smp #1 SMP Tue Dec 13 21:46:01 EST 2005 i686 i686 i386 GNU/Linux Please find also below the detail of IAX debug messages. Cheers, Anto MESSAGES WHEN IAX DOES NOT WORK -- Call accepted by 213.61.187.147 (format ulaw) -- Format for call is ulawTx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00057ms SCall: 16384 DCall: 00070 [213.61.187.147:4569]Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 002 Type: VOICE Subclass: 4 Timestamp: 00080ms SCall: 16384 DCall: 00070 [213.61.187.147:4569]Dec 30 17:04:25 DEBUG[12488]: chan_iax2.c:3699 find_tpeer: Created trunk peer for '213.61.187.147:4569'Dec 30 17:04:25 DEBUG[12488]: chan_iax2.c:3725 iax2_trunk_queue: Expanded trunk '213.61.187.147:4569' to 6400 bytesRx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 003 Type: IAX Subclass: ACK Timestamp: 00080ms SCall: 00070 DCall: 16384 [213.61.187.147:4569] <--- Some messages are missing hereTx-Frame Retry[000] -- OSeqno: 003 ISeqno: 002 Type: IAX Subclass: LAGRQ Timestamp: 10008ms SCall: 16384 DCall: 00070 [213.61.187.147:4569]Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 003 Type: IAX Subclass: LAGRQ Timestamp: 10016ms SCall: 00070 DCall: 16384 [213.61.187.147:4569]Tx-Frame Retry[000] -- OSeqno: 004 ISeqno: 003 Type: IAX Subclass: LAGRP Timestamp: 10016ms SCall: 16384 DCall: 00070 [213.61.187.147:4569]Rx-Frame
[Asterisk-Users] IAX problem - Bug or Compatibility issue?
Hello All, I am looking for more thorough debug than the one provided by the command "iax2 debug". Could anybody point me a good documentation about this? I have a issue with IAX connection. Sometimes it stucked. If so, I have to restart my asterisk through CLI command "restart now". Comparing the debug messages of working and non working sequences, I have noticed that when it does not work, the following debug messages are missing: Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 003 Type: CONTROL Subclass: (14?) Timestamp: 01581ms SCall: 00052 DCall: 16385 [213.61.187.157:4569]Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 003 Type: IAX Subclass: ACK Timestamp: 01581ms SCall: 16385 DCall: 00052 [213.61.187.157:4569] -- IAX2/sipdiscount_outbound-16385 is making progress passing it to Zap/1-1Dec 30 17:12:31 DEBUG[12600]: chan_zap.c:4791 zt_indicate: Requested indication 14 on channel Zap/1-1Dec 30 17:12:31 DEBUG[12600]: chan_zap.c:4857 zt_indicate: Received AST_CONTROL_PROGRESS on Zap/1-1Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 003 Type: VOICE Subclass: 4 Timestamp: 01732ms SCall: 00052 DCall: 16385 [213.61.187.157:4569]Dec 30 17:12:31 DEBUG[12569]: chan_iax2.c:6653 socket_read: Ooh, voice format changed to 4Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 004 Type: IAX Subclass: ACK Timestamp: 01732ms SCall: 16385 DCall: 00052 [213.61.187.157:4569] I have a few questions, especially about the following message: Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 003 Type: CONTROL Subclass: (14?) 1. Is the number 14 in (14?), in decimal or hexadecimal? 2. If that is in decimal, why is it not translated into its descriptions, i.e. Call Progress, according to the IAX2 protocol document I have (Internet-Draft, Expires: July 5, 2005). 3. Why is that number question marked? Is it because asterisk was not sure? 4. If asterisk was not sure, so sometimes it decodes the message sometimes it could not, is there any debug to confirm this? Or, am I looking at the wrong place? Which maybe the problem is so obvious and I missed that? I am running asterisk on IBM xSeries 330 with the following detail: CLI> show versionAsterisk 1.2.1 built by root @ atvie-asterisk on a i686 running Linux on 2005-12-28 07:52:36 UTC# uname -aLinux atvie-asterisk 2.6.14-1.1653_FC4smp #1 SMP Tue Dec 13 21:46:01 EST 2005 i686 i686 i386 GNU/Linux Please find also below the detail of IAX debug messages. Cheers, Anto MESSAGES WHEN IAX DOES NOT WORK -- Call accepted by 213.61.187.147 (format ulaw) -- Format for call is ulawTx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00057ms SCall: 16384 DCall: 00070 [213.61.187.147:4569]Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 002 Type: VOICE Subclass: 4 Timestamp: 00080ms SCall: 16384 DCall: 00070 [213.61.187.147:4569]Dec 30 17:04:25 DEBUG[12488]: chan_iax2.c:3699 find_tpeer: Created trunk peer for '213.61.187.147:4569'Dec 30 17:04:25 DEBUG[12488]: chan_iax2.c:3725 iax2_trunk_queue: Expanded trunk '213.61.187.147:4569' to 6400 bytesRx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 003 Type: IAX Subclass: ACK Timestamp: 00080ms SCall: 00070 DCall: 16384 [213.61.187.147:4569] <--- Some messages are missing hereTx-Frame Retry[000] -- OSeqno: 003 ISeqno: 002 Type: IAX Subclass: LAGRQ Timestamp: 10008ms SCall: 16384 DCall: 00070 [213.61.187.147:4569]Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 003 Type: IAX Subclass: LAGRQ Timestamp: 10016ms SCall: 00070 DCall: 16384 [213.61.187.147:4569]Tx-Frame Retry[000] -- OSeqno: 004 ISeqno: 003 Type: IAX Subclass: LAGRP Timestamp: 10016ms SCall: 16384 DCall: 00070 [213.61.187.147:4569]Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 004 Type: IAX Subclass: LAGRP Timestamp: 10008ms SCall: 00070 DCall: 16384 [213.61.187.147:4569]Tx-Frame Retry[-01] -- OSeqno: 004 ISeqno: 004 Type: IAX Subclass: ACK Timestamp: 10008ms SCall: 16384 DCall: 00070 [213.61.187.147:4569]Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 005 Type: IAX Subclass: ACK Timestamp: 10016ms SCall: 00070 DCall: 16384 [213.61.187.147:4569]Rx-Frame Retry[ No] -- OSeqno: 004 ISeqno: 005 Type: IAX Subclass: HANGUP Timestamp: 10262ms SCall: 00070 DCall: 16384 [213.61.187.147:4569] CAUSE CODE : 0 MESSAGES AFTER ISSUING "CLI> restart now" command
[Asterisk-Users] IAX problem
Hi all,
I have 2 servers and I'm trying to configure iax to call from
Server2 (fxo) to Server1 (sip extension)
Server1: 2 sip's extension (123 and 321)
Server2: TDM-400 1 fxo(extension 999, channel 3) and 1 fxs (channel 4)
from 123 to 321 it's all right in both ways
from 999 to 123 no audio in both ways (the telephone of 123 rings but no sound)
Here's my configuration:
-- SERVER1 sip.conf -
[general]
context=sip
port=5060
bindaddr=192.168.0.1
disallow=all
allow=g729
allow=ulaw
language=en
[123]
type=friend
host=dynamic
username=123
secret=f599
callerid="123"<123>
context=sip
canreinvite=no
dtmfmode=rfc2833
mailbox=123
allow=g729
allow=ulaw
nat=yes
qualify=800
[321]
type=friend
host=dynamic
username=321
secret=f599
callerid="321"<321>
context=sip
canreinvite=no
dtmfmode=rfc2833
mailbox=321
allow=g729
allow=ulaw
nat=yes
qualify=800
- SERVER1 extensions.conf
[general]
static=yes
writeprotect=no
[default]
exten => 123,1,Dial(SIP/123,20,tr)
exten => 123,2,Voicemail,u123
exten => 123,102,Voicemail,b123
exten => 321,1,Dial(SIP/321,20,tr)
exten => 321,2,Voicemail,u321
exten => 321,102,Voicemail,b321
- SERVER1 iax.conf --
[general]
bandwidth=low
jitterbuffer=no
tos=lowdelay
[123]
type=user
host=192.168.0.2
context=default
dtmfmode=rfc2833
--- SERVER2 extension.conf ---
exten => 999,1,Dial(IAX2/[EMAIL PROTECTED]/123)
no iax.conf at SERVER2
no sip.conf at SERVER2
Debug at SERVER1
SERVER1> iax2 debug
IAX2 Debugging Enabled
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW
Timestamp: 8ms SCall: 1 DCall: 0 [192.168.0.2:4569]
VERSION : 2
CALLED NUMBER : 123
CALLING NUMBER : 403
CALLING NAME : Channel 3
LANGUAGE : en
USERNAME : 123
FORMAT : 2
CAPABILITY : 65283
ADSICPE : 2
DATE TIME : 189032149
-- Accepting unauthenticated call from 192.168.0.2, requested format = 2, actual format = 2
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACCEPT
Timestamp: 7ms SCall: 1 DCall: 1 [192.168.0.2:4569]
FORMAT : 2
-- Executing Dial("IAX2/[EMAIL PROTECTED]:4569/1", "SIP/123|20|tr") in new stack
-- Called 123
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW
Timestamp: 8ms SCall: 1 DCall: 0 [192.168.0.2:4569]
VERSION : 2
CALLED NUMBER : 123
CALLING NUMBER : 403
CALLING NAME : Channel 3
LANGUAGE : en
USERNAME : 123
FORMAT : 2
CAPABILITY : 65283
ADSICPE : 2
DATE TIME : 189032149
-- Accepting unauthenticated call from 192.168.0.2, requested format = 2, actual format = 2
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACCEPT
Timestamp: 7ms SCall: 1 DCall: 1 [192.168.0.2:4569]
FORMAT : 2
-- Executing Dial("IAX2/[EMAIL PROTECTED]:4569/1", "SIP/123|20|tr") in new stack
-- Called 123
Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: CONTROL Subclass: RINGING
Timestamp: 00010ms SCall: 1 DCall: 1 [192.168.0.2:4569]
Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK
Timestamp: 7ms SCall: 1 DCall: 1 [192.168.0.2:4569]
-- SIP/123-541f is ringing
Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: CONTROL Subclass: RINGING
Timestamp: 00010ms SCall: 1 DCall: 1 [192.168.0.2:4569]
Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK
Timestamp: 7ms SCall: 1 DCall: 1 [192.168.0.2:4569]
-- SIP/123-541f is ringing
Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 001 Type: CONTROL Subclass: RINGING
Timestamp: 00013ms SCall: 1 DCall: 1 [192.168.0.2:4569]
Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: VOICE Subclass: 2
Timestamp: 00040ms SCall: 1 DCall: 1 [192.168.0.2:4569]
Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACK
Timestamp: 00040ms SCall: 1 DCall: 1 [192.168.0.2:4569]
Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACK
Timestamp: 00010ms SCall: 1 DCall: 1 [192.168.0.2:4569]
Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 003 Type: IAX Subclass: ACK
Timestamp: 00013ms SCall: 1 DCall: 1 [192.168.0.2:4569]
Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 003 Type: IAX Subclass: LAGRQ
Timestamp: 10008ms SCall: 1 DCall: 1 [192.168.0.2:4569]
Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 003 Type: IAX Subclass: LAGRP
Timestamp: 10008ms SCall: 1 DCall: 1 [192.168.0.2:4569]
Tx-Frame Retry[000] -- OSeqno: 004 ISeqno: 003 Type: IAX Subclass: LAGRQ
Timestamp: 10011ms SCall: 1 DCall: 1 [192.168.0.2:4569]
Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 004 Type: IAX Subclass: ACK
Timestamp: 10008ms S
Re: [Asterisk-Users] iax problem
> > > > For #2, incoming calls would be handled with:
> > > > exten => 6789,1,Dial(SIP/1235)
> > > >
> > > Besides that :
> > >
> > > *CLI> iax2 show registry
> > > Host UsernamePerceived Refresh State
> > > X.X.X.X:4569 Username1 [MYIP]:456960 Registered
> > > X.X.X.X:4569 Username2 [MYIP]:456960 Registered
> > > X.X.X.X:4569 Username3 [MYIP]:456960 Registered
> > >
> > > source and destination ports for all 3 iax registrations are the same ,
> > > and my isp see only one, becouse rest is overwriten.
> >
> > Have you tried using three different contexts for those in iax.conf?
> >
> >
> Yes and result is as I suppose :
>
> -- Accepting UNAUTHENTICATED call from X.X.X.X:
>> requested format = ilbc,
>> requested prefs = (ilbc|gsm|ulaw|alaw),
>> actual format = ilbc,
>> host prefs = (),
>> priority = caller
> -- Executing Dial("IAX2/[EMAIL PROTECTED]:4569-1", "IAX2/1237") in new
> stack
> -- Called 1237
> -- Call accepted by 192.168.57.238 (format gsm)
> -- Format for call is gsm
> -- IAX2/1237-8 is ringing
> -- Hungup 'IAX2/1237-8'
>
> Everything enters via last registred username 'Username3'.
I'm out of ideas other then to open a feature request to add the
/1234 syntax to the register statement for iax.
___
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Re: [Asterisk-Users] iax problem
On Mon, Sep 26, 2005 at 11:02:47AM -0600, Rich Adamson wrote:
>
> > > For #2, incoming calls would be handled with:
> > > exten => 6789,1,Dial(SIP/1235)
> > >
> > Besides that :
> >
> > *CLI> iax2 show registry
> > Host UsernamePerceived Refresh State
> > X.X.X.X:4569 Username1 [MYIP]:456960 Registered
> > X.X.X.X:4569 Username2 [MYIP]:456960 Registered
> > X.X.X.X:4569 Username3 [MYIP]:456960 Registered
> >
> > source and destination ports for all 3 iax registrations are the same ,
> > and my isp see only one, becouse rest is overwriten.
>
> Have you tried using three different contexts for those in iax.conf?
>
>
Yes and result is as I suppose :
-- Accepting UNAUTHENTICATED call from X.X.X.X:
> requested format = ilbc,
> requested prefs = (ilbc|gsm|ulaw|alaw),
> actual format = ilbc,
> host prefs = (),
> priority = caller
-- Executing Dial("IAX2/[EMAIL PROTECTED]:4569-1", "IAX2/1237") in new stack
-- Called 1237
-- Call accepted by 192.168.57.238 (format gsm)
-- Format for call is gsm
-- IAX2/1237-8 is ringing
-- Hungup 'IAX2/1237-8'
Everything enters via last registred username 'Username3'.
/pch
--
Dyslexia bug unpatched since 1977 ...
exploit has been leaked to the underground.
___
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Re: [Asterisk-Users] iax problem
> > Two approaches that have been rather common are: > > 1. use the separate contexts for each did, > > 2. in the register statement, add /1234 at the end; like > > register => username:[EMAIL PROTECTED]/6789 > > > I don't think it will work , iax statement don't have > exten on end. > > [..] > register [:] @ [:] To register with > another IAX server. > [..] Ops... > This is true for SIP but not for IAX. > > > > For #2, incoming calls would be handled with: > > exten => 6789,1,Dial(SIP/1235) > > > Besides that : > > *CLI> iax2 show registry > Host UsernamePerceived Refresh State > X.X.X.X:4569 Username1 [MYIP]:456960 Registered > X.X.X.X:4569 Username2 [MYIP]:456960 Registered > X.X.X.X:4569 Username3 [MYIP]:456960 Registered > > source and destination ports for all 3 iax registrations are the same , > and my isp see only one, becouse rest is overwriten. Have you tried using three different contexts for those in iax.conf? ___ --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] iax problem
On Sun, Sep 25, 2005 at 07:26:12AM -0600, Rich Adamson wrote: > > Two approaches that have been rather common are: > 1. use the separate contexts for each did, > 2. in the register statement, add /1234 at the end; like > register => username:[EMAIL PROTECTED]/6789 > I don't think it will work , iax statement don't have exten on end. [..] register [:] @ [:] To register with another IAX server. [..] This is true for SIP but not for IAX. > For #2, incoming calls would be handled with: > exten => 6789,1,Dial(SIP/1235) > Besides that : *CLI> iax2 show registry Host UsernamePerceived Refresh State X.X.X.X:4569 Username1 [MYIP]:456960 Registered X.X.X.X:4569 Username2 [MYIP]:456960 Registered X.X.X.X:4569 Username3 [MYIP]:456960 Registered source and destination ports for all 3 iax registrations are the same , and my isp see only one, becouse rest is overwriten. /pch -- Dyslexia bug unpatched since 1977 ... exploit has been leaked to the underground. ___ --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] iax problem
> I've 3 iax connections to my provider , each of them have own DID , > > PH1<| > | >\/ > PH2<-->|-| <---> ||<-- DID1 >| A1 | <---> |ISP |<-- DID2 > PH3<-->|-| <---> ||<-- DID3 > > I had iax phone on each of this connection , but now I want > to terminate all on my asterisk box , and send calls to phones connected > to my asterisk depending to incoming username/DID . > > for example : > > Call to DID1 must be directed to PH1 , DID2 to PH2 and DID3 to PH3 etc > > In iax.conf I have : > > [Username1] ;DID1 > type=user > username=Username11 > ;secret=blah > host=X.X.X.X > context=fromisp1 > > [Username2] ;DID2 > type=user > username=Username2 > host=X.X.X.X > context=fromisp1 > > > [Username3] ;DID3 > type=user > username=Username3 > host=X.X.X.X > context=fromisp1 > > For each of the iax connection I have defined section with type user. > > In extension.conf I have : > > [fromisp1] > exten => s,1,Dial(SIP/1235) > exten => _X.,1,Dial(SIP/1235) > exten => h,1,Hangup > > Every incoming call enters context fromisp1 with exten 's' . > I can't distinguish incoming DID or username, of couse I've figure out > that I can create context for each iax connection , but for me I would > be wast of cpu cycles :) > > Some other ideas for my problem ?:) Two approaches that have been rather common are: 1. use the separate contexts for each did, 2. in the register statement, add /1234 at the end; like register => username:[EMAIL PROTECTED]/6789 For #2, incoming calls would be handled with: exten => 6789,1,Dial(SIP/1235) ___ --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] iax problem
Hi I've 3 iax connections to my provider , each of them have own DID , PH1<| | \/ PH2<-->|-| <---> ||<-- DID1 | A1 | <---> |ISP |<-- DID2 PH3<-->|-| <---> ||<-- DID3 I had iax phone on each of this connection , but now I want to terminate all on my asterisk box , and send calls to phones connected to my asterisk depending to incoming username/DID . for example : Call to DID1 must be directed to PH1 , DID2 to PH2 and DID3 to PH3 etc In iax.conf I have : [Username1] ;DID1 type=user username=Username11 ;secret=blah host=X.X.X.X context=fromisp1 [Username2] ;DID2 type=user username=Username2 host=X.X.X.X context=fromisp1 [Username3] ;DID3 type=user username=Username3 host=X.X.X.X context=fromisp1 For each of the iax connection I have defined section with type user. In extension.conf I have : [fromisp1] exten => s,1,Dial(SIP/1235) exten => _X.,1,Dial(SIP/1235) exten => h,1,Hangup Every incoming call enters context fromisp1 with exten 's' . I can't distinguish incoming DID or username, of couse I've figure out that I can create context for each iax connection , but for me I would be wast of cpu cycles :) Some other ideas for my problem ?:) /pch PS: This is my first post , don't shot me :) -- Dyslexia bug unpatched since 1977 ... exploit has been leaked to the underground. ___ --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] IAX problem; one end sounds like on fast forward
Hi, I have some issues with communication between to * servers. They are connected over DSL (3Mbps). One is behind NAT and the other on routable network. Almost every time caller will hear the other end like fast forward while the other end will have perfect quality. It doesn't matter if we use SIP phones (Cisco and Grandstreams) or analog sets via Sipura-2K. If I call the city through Mediatrix 1204 the quality is perfect. I am suspecting that this problem is related to jitter, but can not resolve it. I've tried using ulaw and ilbc with similar results. Both sites are configured to use IAX trunking and both have X101P to provide clocking (on one end the X101P is in red-alarm state as the line is not plugged in into X101P). I am tempted to switch to SIP for interoffice communication but first I want to try few more things.. Any suggestions? Regards, Wojtek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
