Re: [asterisk-users] Where an extension really is (DUNDi woes)

[Kyle Sexton]

On 6/15/07, Anthony Francis <[EMAIL PROTECTED]> wrote:


Kyle Sexton wrote:
> I have two servers setup to do DUNDi lookups against each other.  The
> scenario is that on server A, I have a wildcard match for extensions
> 64XX that rings to a local extension on the server.  On server B I
> have a 6442 real extension that I would like to have ring if called.
> It seems that DUNDi is matching on the 64XX and not searching out to
> see if there is a *more* exact match than the pattern match.  Is there
> any way to get around this?

I don't think I am incorrect in saying that dundi doesn't look for
externally that which it knows about locally. I think thats pretty
standard of routing protocols.



I was afraid of that.  It just means I have to explicitly list every number
in the DID range (so hundreds of extensions).  I was hoping DUNDi would make
the dialplan simpler. :(

--
Kyle Sexton
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[asterisk-users] Asterisk to Panasonic TDA200 with Unicall

[Carlos Chavez]

 I am trying to set up a connection from an Asterisk server with a 2 port
E1 card to a Panasonic TDA200 pbx.  We are using Asterisk 1.4.4, Zaptel 1.4.3
with Unicall.  So far we can get the E1 link up and we can send calls from the
Panasonic to an IP phone on Asterisk and even through the other E1 to the
PSTN.  The problem is when trying to send calls from Asterisk to the
Panasonic, we can only hear one ring and then the connection drops. 
Unfortunately I do not have access to the log files at the moment so from
memory the error we got said that ther was an "Unexpected MF6 signal" before
the call was dropped.  

 I have several systemas talking to other Panasonic models like the D500.
 They all work fine.  We have basically the same configuration on those except
that we use Asterisk 1.2.X instead of 1.4.X on those.  Does anyone know what
that error message mean?  Tomorrow I will travel back to where the server is
and try again.  I will try to determine if the problem is caused by the
Panasonic or the Asterisk side.

--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001


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Re: [asterisk-users] combining AGI with dialplans

[james]

In my first message I included the example script:

> ***test.php***
> #!/usr/bin/php -q
>  require_once('phpagi.php');
> $agi = new AGI();
>
> $dialstr="IAX2/wayne/[EMAIL PROTECTED]";
> $agi->SetVar("JAMES",$dialstr);
> exit(0);
> ?>

I have found why it didn't work.  I needed to call $agi->set_variable()
not $agi->SetVar.  This does what I wanted and expected.


James

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Re: [asterisk-users] Unicall + MFC/R2 line dropped immediately afterconnect

[Steve Underwood]

Peter Gubis wrote:
> Hi,
>
> connection is already established (i can also hear called person for a
> while). Problem is, that the line is every time dropped after 1 second.
> I assume, that first billing pulse arrives immediately after link is
> established and it drops the line.
>
> Next week I'll be able to play around with timeouts and will try to
> change mfcr2->clear_back_persistence_check parameter as you advised in
> your second reply.
>   
What you are seeing maybe pulses similar to metering pulses, but for 
another purpose.

In some places it is common practice to create a pulse just after call 
connection. This will kick out any collect calls, and prevent nasty 
unexepected bills. The answer is exactly the same as for tolerating 
meter pulses - set a clearback timeout and things should work OK.

I didn't want to make a clearback timeout the default, as it slows down 
calls. However, people all over South America seem to be hitting this 
problem now. Maybe I should make a timeout the default for some national 
variants, and tell people how to remove it if they don't need it.

Steve


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Re: [asterisk-users] Que on A2Billing

[Nitesh Divecha]

Strange...
Got it working now... I can receive incoming call...

Changed following parameters in additional_a2billing_sip.conf of the DID 
to: -

qualify=yes
canreinvite=no

Cheers,
Nitesh



Guillermo Salas M. wrote:
> On Fri, 2007-06-15 at 17:42 -0400, Nitesh Divecha wrote:
>   
>> When I call from my cell to the above DID, it hits on the Asterisk and
>> I 
>> see A2Billing trying to call SIP/2486543210, but it fails because 
>> Asterisk says "Unable to create channel of type 'SIP' (cause 3 - No 
>> route to destination) ". 
>> 
>
> I know it, but the error is saying that you don't have one 2486543210
> user registred.
>
> Show us the output of:
>
> sip show peers
>
> Regards,
>
>   


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Re: [asterisk-users] Que on A2Billing

[Nitesh Divecha]

Here is my "sip show peers"

hyperion*CLI> sip show peers
Name/username  HostDyn Nat ACL Port Status   
2486543210/2486543210  86.14.22.128 D   N  61547LAGGED 
(66 ms)

Now here is the catch, before it used to show the status OK but now its 
showing LAGGED.
Dunno what does that means... Any suggestions...

Cheers,
Nitesh




Guillermo Salas M. wrote:
> On Fri, 2007-06-15 at 17:42 -0400, Nitesh Divecha wrote:
>   
>> When I call from my cell to the above DID, it hits on the Asterisk and
>> I 
>> see A2Billing trying to call SIP/2486543210, but it fails because 
>> Asterisk says "Unable to create channel of type 'SIP' (cause 3 - No 
>> route to destination) ". 
>> 
>
> I know it, but the error is saying that you don't have one 2486543210
> user registred.
>
> Show us the output of:
>
> sip show peers
>
> Regards,
>
>   


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[asterisk-users] Asterisk 1.4.anything on FreeBSD?

[Bruce Komito]

I was very pleased to learn that 1.4.5 has been released.  Unfortunately,
I have been beating my head against a wall trying to install 1.4.4 on
FreeBSD (6.2).

If you have been successful in building 1.4.anything (including addons and
zaptel-bsd-trunk), could you please respond, on- or off-list with the
secret?  Plain vanilla asterisk is ok (I think), but when I try to build
asterisk-addons-1.4.1 and zaptel-bsd-trunk, nothing seems to be in the
right place (or at least in the place where the build scripts expect it to
be) and the build fails.  I've tried moving stuff around, creating
symbolic links, you name it, but unwinding the build spaghetti is beyond
my capabilities, I'm afraid.

TIA

[If you don't have any experience with FreeBSD, please don't bother
responding!]

Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815



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[asterisk-users] Asterisk 1.2.19 and 1.4.5 released!

[The Asterisk Development Team]

The Asterisk development team is proud to release Asterisk versions 
1.2.19 and 1.4.5.

There has been a very large number of bugs fixed since the last release, 
including crashes and other critical issues.  There were 244 commits to 
the 1.4 source tree and 74 commits to the 1.2 source tree since the last 
set of releases.  Also, keep in mind that the release branches are only 
changed to fix problems.  In the same time frame, the development tree 
had 439 commits.  There is a ChangeLog available in the tarball of each 
release with a complete list of changes.

Keep in mind that the date for moving the 1.2 release series to security 
fix maintenance only will be coming within the next couple of months. 
We strongly encourage everyone to migrate to Asterisk 1.4.  Don't forget 
to read the UPGRADE.txt file in 1.4 for important information regarding 
upgrading from 1.2.

These releases are available for download from ftp.digium.com.  They are 
distributed as both tarballs and patch sets against the previous 
releases.  All release files have been signed with GPG keys from members 
of the Digium software development team to ensure authenticity.

As always, thank you very much for your support!

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[asterisk-users] Accessing Voicemail(Asterisk) -- Remotely either from Cell or Landline

[Deepak Naidu]

Hi, I was wondering if we can check the voicemails remotely from a cell or a 
landline number.

We have SIP 3 Digit Extensions connected to Asterisk server.

If users are away from Desk & need to access voicemails can they dial in to 
Asterisk PBX & check their messages.

I know one can check through web link & even have mailed.  Aslo I have checked 
regarding DISA, but I am not kind of OK in using DISA now for just voicemails.

Is their any other ways.  I am using Free PBX so can I do any thing from 
FreePBX to manager it, if not backend configs are fine.
  
--
Deepak


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Re: [asterisk-users] calling

[Carol McGeehon]

Dean,
I would like to use it for this application only.  Our phone
service is currently provided by our parent organization with is a
county government unit.  Thanks for your reply.
 
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dean
Collins
Sent: Friday, June 15, 2007 3:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] calling
 
Hi Carol,
Yes Asterisk can be set up to provide this service.
 
>From the tone of your question I'm guessing you may not be currently
using Asterisk.
 
Is this all you want Asterisk to do? (eg as an application service
rather than provide telephony for the rest of the library as well), or
are you looking to have it replace your existing telephony equipment?
 
As a suggestion if you google Trixbox and Nerd Vittles you will find a
fairly detailed explanation of how to set your Trixbox server (a version
of Asterisk) up to provide exactly what you are after. If this is too
complicated for you never fear, just post an email of your requirements
and ask for someone/body to 'build' you a version of Asterisk for a fee
and I'm sure you will have a few people offering bids in no time.
 
 
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
 +1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).
 


From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Carol
McGeehon
Sent: Friday, 15 June 2007 5:36 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] calling
 
I would like to find out if Asterisk will allow you to call multiple
phone numbers with a pre-recorded message.  I currently use a calling
system software to contact people to let them know they have a book
waiting for them at the public library.  I'd also like to know if the
software will allow you to use a report to load a list of phone numbers
to contact.  
 
Carol McGeehon
Technical Support Manager
Douglas County Library System
1409 NE Diamond Lake Blvd
Roseburg, OR   97470
541-440-6005
 
When it is dark enough, you can see the stars.
 
 

-- 
This message has been scanned for viruses and dangerous content by
MailScanner, and is believed to be clean. 

Douglas County, Oregon 
Douglas County, Oregon   
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Re: [asterisk-users] combining AGI with dialplans

[Lee Jenkins]

[EMAIL PROTECTED] wrote:
> On 2007-05-15 Tony Mountifield wrote (wrt using AGI scripts to dial out):
> 
>> Can't comment on this one, as I never use AGI to dial.
>> My AGIs just set the context, extension and priority,
>> and exit to the dialplan to do any dialling.
> 
> (http://article.gmane.org/gmane.comp.telephony.pbx.asterisk.user/185537)
> 
> 
> I would like to do this, but I am having trouble figuring out how. I have
> tried the following but it is not working for me:
> 
> 
> ***test.php***
> #!/usr/bin/php -q
>  require_once('phpagi.php');
> $agi = new AGI();
> 
> $dialstr="IAX2/wayne/[EMAIL PROTECTED]";
> $agi->SetVar("JAMES",$dialstr);
> exit(0);
> ?>
> 
> ***extensions.conf***
> [from-sip]
> exten => 111,1,DeadAGI(test.php)
> exten => 111,2,Dial(${JAMES})
> exten => 111,3,Hangup
> 
> 
> 
> Thanks in advance for any help.
> James
> 

Checkout the "h" hangup extension:
http://www.voip-info.org/wiki/index.php?page=Asterisk+h+extension

-- 

Warm Regards,

Lee




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Re: [asterisk-users] Where an extension really is (DUNDi woes)

[Anthony Francis]

Kyle Sexton wrote:
> I have two servers setup to do DUNDi lookups against each other.  The 
> scenario is that on server A, I have a wildcard match for extensions 
> 64XX that rings to a local extension on the server.  On server B I 
> have a 6442 real extension that I would like to have ring if called.  
> It seems that DUNDi is matching on the 64XX and not searching out to 
> see if there is a *more* exact match than the pattern match.  Is there 
> any way to get around this?
>
>
> -- 
> Kyle Sexton
> 
>
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> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
I don't think I am incorrect in saying that dundi doesn't look for 
externally that which it knows about locally. I think thats pretty 
standard of routing protocols.

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Re: [asterisk-users] calling

[Dean Collins]

Hi Carol,

Yes Asterisk can be set up to provide this service.

 

>From the tone of your question I'm guessing you may not be currently
using Asterisk.

 

Is this all you want Asterisk to do? (eg as an application service
rather than provide telephony for the rest of the library as well), or
are you looking to have it replace your existing telephony equipment?

 

As a suggestion if you google Trixbox and Nerd Vittles you will find a
fairly detailed explanation of how to set your Trixbox server (a version
of Asterisk) up to provide exactly what you are after. If this is too
complicated for you never fear, just post an email of your requirements
and ask for someone/body to 'build' you a version of Asterisk for a fee
and I'm sure you will have a few people offering bids in no time.

 

 

Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
 +1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Carol
McGeehon
Sent: Friday, 15 June 2007 5:36 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] calling

 

I would like to find out if Asterisk will allow you to call multiple
phone numbers with a pre-recorded message.  I currently use a calling
system software to contact people to let them know they have a book
waiting for them at the public library.  I'd also like to know if the
software will allow you to use a report to load a list of phone numbers
to contact.  

 

Carol McGeehon

Technical Support Manager

Douglas County Library System

1409 NE Diamond Lake Blvd

Roseburg, OR   97470

541-440-6005

 

When it is dark enough, you can see the stars.

 

 

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Re: [asterisk-users] Que on A2Billing

[Guillermo Salas M.]

On Fri, 2007-06-15 at 17:42 -0400, Nitesh Divecha wrote:
> When I call from my cell to the above DID, it hits on the Asterisk and
> I 
> see A2Billing trying to call SIP/2486543210, but it fails because 
> Asterisk says "Unable to create channel of type 'SIP' (cause 3 - No 
> route to destination) ". 

I know it, but the error is saying that you don't have one 2486543210
user registred.

Show us the output of:

sip show peers

Regards,

-- 
Guillermo Salas M.
Telconet S.A.
Calle 15 y Avenida 24 Esq
Edificio Barre #2 Primer Piso
Telefono : +593 5 262 8071
Celular  : +593 9 985 5138
e-mail   : [EMAIL PROTECTED]
www  : http://www.manta.telconet.net
   http://www.telcocarrier.net

Linux User: 255902

Beat me, whip me, make me use Windows!

Please avoid sending me Word or PowerPoint attachments.
See http://www.fsf.org/philosophy/no-word-attachments.html

Please avoid the Top Posting, see
http://es.wikipedia.org/wiki/Top-posting


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Re: [asterisk-users] Que on A2Billing

[Nitesh Divecha]

2486543210 is my SIP-Friend which I created manually and associated with 
one of the card number.
My ATA is registered to Asterisk using the about DID Number.
So I want when I call the above number, it should ring on the ATA.
When I call from my cell to the above DID, it hits on the Asterisk and I 
see A2Billing trying to call SIP/2486543210, but it fails because 
Asterisk says "Unable to create channel of type 'SIP' (cause 3 - No 
route to destination) ".

Any suggestion...

Cheers,
Nitesh





Guillermo Salas M. wrote:
> On Fri, 2007-06-15 at 17:01 -0400, Nitesh Divecha wrote:
>   
>> Thanks man... That really helped me to move couple of steps. Now I see
>> the incoming calls are going in proper direction... I know I am still
>> missing a small piece here... I did ADD the Destination as a
>> SIP/2486543210, assigned the card number, enabled VOIP_CALL, and
>> enabled Active. 
>>
>> 
>
>
> 2486543210 is your card number?
>
>
>   
>> When I dial the DID number, on the *CLI it shows the following: -
>>
>> a2billing.php|1|did: bug
>> -- AGI Script Executing Application: (DIAL) Options:
>> (SIP/2486543210|60|HL(360:61000:3))
>> -- Limit Data for this call: 
>> -- - timelimit = 360
>> -- - play_warning  = 61000
>> -- - play_to_caller= yes
>> -- - play_to_callee= no
>> -- - warning_freq  = 3
>> -- - start_sound   = UNDEF
>> -- - warning_sound = timeleft 
>> -- - end_sound = UNDEF
>> Destroying call '[EMAIL PROTECTED]'
>> Jun 15 16:41:34 NOTICE[15346]: app_dial.c:1069 dial_exec_full: Unable
>> to create channel of type 'SIP' (cause 3 - No route to destination) 
>>   == Everyone is busy/congested at this time (1:0/0/1)
>>
>> 
>
> I think that 2486543210 is not a customer, card number or SIP/IAX2
> friend, maybe is PSTN number. To redirect the call to any PSTN number
> you must need to set "voip call" to inactive and set the destination
> number to 2486543210.
>
>
>   
>> I bet I am missing something in extension.conf correct? I dont see any
>> examples in my package.
>>
>> 
>
>
> The context is fine don't worry about it.
>
>
>   
>> Any suggestion... Thanks once again... 
>> 
>
>
> Regards,
>
>   


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[asterisk-users] calling

[Carol McGeehon]

I would like to find out if Asterisk will allow you to call multiple
phone numbers with a pre-recorded message.  I currently use a calling
system software to contact people to let them know they have a book
waiting for them at the public library.  I'd also like to know if the
software will allow you to use a report to load a list of phone numbers
to contact.  
 
Carol McGeehon
Technical Support Manager
Douglas County Library System
1409 NE Diamond Lake Blvd
Roseburg, OR   97470
541-440-6005
 
When it is dark enough, you can see the stars.
 
 
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[asterisk-users] scaling with SMP

[Mark Price]

Is there a way to cause asterisk to benefit from running on a machine
with more than two cores?  I only see two processes running, with one
at a very low priority and the other at a very high priority.  I'm
guessing one is managing the other.

Thanks,
Mark

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[asterisk-users] Where an extension really is (DUNDi woes)

[Kyle Sexton]

I have two servers setup to do DUNDi lookups against each other.  The
scenario is that on server A, I have a wildcard match for extensions 64XX
that rings to a local extension on the server.  On server B I have a 6442
real extension that I would like to have ring if called.  It seems that
DUNDi is matching on the 64XX and not searching out to see if there is a
*more* exact match than the pattern match.  Is there any way to get around
this?


--
Kyle Sexton
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Re: [asterisk-users] Que on A2Billing

[Nitesh Divecha]

Thanks man... That really helped me to move couple of steps. Now I see the
incoming calls are going in proper direction... I know I am still missing a
small piece here... I did ADD the Destination as a SIP/2486543210, assigned
the card number, enabled VOIP_CALL, and enabled Active.

When I dial the DID number, on the *CLI it shows the following: -

a2billing.php|1|did: bug
   -- AGI Script Executing Application: (DIAL) Options:
(SIP/2486543210|60|HL(360:61000:3))
   -- Limit Data for this call:
   -- - timelimit = 360
   -- - play_warning  = 61000
   -- - play_to_caller= yes
   -- - play_to_callee= no
   -- - warning_freq  = 3
   -- - start_sound   = UNDEF
   -- - warning_sound = timeleft
   -- - end_sound = UNDEF
Destroying call '[EMAIL PROTECTED]'
Jun 15 16:41:34 NOTICE[15346]: app_dial.c:1069 dial_exec_full: Unable to
create channel of type 'SIP' (cause 3 - No route to destination)
 == Everyone is busy/congested at this time (1:0/0/1)

I bet I am missing something in extension.conf correct? I dont see any
examples in my package.

Any suggestion... Thanks once again...

Cheers,
Nitesh






On 6/15/07, Guillermo Salas M. <[EMAIL PROTECTED]> wrote:


On Fri, 2007-06-15 at 14:20 -0400, Nitesh Divecha wrote:
> You said change the context for SIP Customers to
> "context=a2billing-did", do I have to create this context or
> A2Billing
> will generate by itself?
>


The a2billing package comes with some examples, you must have to create
the a2billing-did context :

[a2billing-did]
exten => _X.,1,NoOp,${CALLERID(all)}
exten => _X.,2,DeadAGI(a2billing.php|1|did)
exten => _X.,3,Hangup()

This will be the context for your DID provider and not for your
customers.

Check this link for more information:

http://forum.asterisk2billing.org/viewtopic.php?t=1784


Cheers!

--
Guillermo Salas M.
Telconet S.A.
Calle 15 y Avenida 24 Esq
Edificio Barre #2 Primer Piso
Telefono : +593 5 262 8071
Celular  : +593 9 985 5138
e-mail   : [EMAIL PROTECTED]
www  : http://www.manta.telconet.net
   http://www.telcocarrier.net

Linux User: 255902

Beat me, whip me, make me use Windows!

Please avoid sending me Word or PowerPoint attachments.
See http://www.fsf.org/philosophy/no-word-attachments.html

Please avoid the Top Posting, see
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Re: [asterisk-users] Que on A2Billing

[Guillermo Salas M.]

On Fri, 2007-06-15 at 17:01 -0400, Nitesh Divecha wrote:
> Thanks man... That really helped me to move couple of steps. Now I see
> the incoming calls are going in proper direction... I know I am still
> missing a small piece here... I did ADD the Destination as a
> SIP/2486543210, assigned the card number, enabled VOIP_CALL, and
> enabled Active. 
> 


2486543210 is your card number?


> When I dial the DID number, on the *CLI it shows the following: -
> 
> a2billing.php|1|did: bug
> -- AGI Script Executing Application: (DIAL) Options:
> (SIP/2486543210|60|HL(360:61000:3))
> -- Limit Data for this call: 
> -- - timelimit = 360
> -- - play_warning  = 61000
> -- - play_to_caller= yes
> -- - play_to_callee= no
> -- - warning_freq  = 3
> -- - start_sound   = UNDEF
> -- - warning_sound = timeleft 
> -- - end_sound = UNDEF
> Destroying call '[EMAIL PROTECTED]'
> Jun 15 16:41:34 NOTICE[15346]: app_dial.c:1069 dial_exec_full: Unable
> to create channel of type 'SIP' (cause 3 - No route to destination) 
>   == Everyone is busy/congested at this time (1:0/0/1)
> 

I think that 2486543210 is not a customer, card number or SIP/IAX2
friend, maybe is PSTN number. To redirect the call to any PSTN number
you must need to set "voip call" to inactive and set the destination
number to 2486543210.


> I bet I am missing something in extension.conf correct? I dont see any
> examples in my package.
> 


The context is fine don't worry about it.


> Any suggestion... Thanks once again... 


Regards,

-- 
Guillermo Salas M.
Telconet S.A.
Calle 15 y Avenida 24 Esq
Edificio Barre #2 Primer Piso
Telefono : +593 5 262 8071
Celular  : +593 9 985 5138
e-mail   : [EMAIL PROTECTED]
www  : http://www.manta.telconet.net
   http://www.telcocarrier.net

Linux User: 255902

Beat me, whip me, make me use Windows!

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See http://www.fsf.org/philosophy/no-word-attachments.html

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Re: [asterisk-users] Community PBX?

[Gonzalo Servat]

On 6/15/07, Kyle Sexton <[EMAIL PROTECTED]> wrote:


I'm wondering if anyone out there is running a community PBX for their
local Asterisk User Groups or area Linux groups.  I've been thinking of
setting one up but am stuck as to what services to provide that people would
actually find useful.  I know that I could setup simple SIP->SIP to allow
everyone to call each other, but that's not generally too fun.



.. and about 500 million other places already offer this :)

One suggestion would be to setup something like a "virtual office" where you
could allow users to point their DIDs to their SIP/IAX account on your
server and receive calls, maybe allow free calls to freecall destinations,
voicemail, stuff like that.

As an example of a community project, there's www.voipuser.org which, I
think, had the right idea. They turn their income from inbound calls into
outbound minutes for the community to use. Pretty smart!

- Gonzalo
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Re: [asterisk-users] Run as root?

[Remco Post]

Malcom Kemp wrote:
> In looking at the safe_asterisk script, it would appear that it is
> encouraging the running of the Asterisk application as root user.  My
> natural inclination is to run it as a non-privileged user.  What is
> recommendation?
> 

the recommendation is not to run as root, and have both the desired user
and group configure in your asterisk.conf

> 
> 
> 
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-- 

Remco Post

"I didn't write all this code, and I can't even pretend that all of it
makes sense." -- Glen Hattrup

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Re: [asterisk-users] app_rxfax vs (iaxmodem+hylafax)

[Paradise Dove]

On 6/15/07, Steve Underwood <[EMAIL PROTECTED]> wrote:
> Paradise Dove wrote:
> > can anybody help me to choose the most reliable fax solution for * .
> > after googling the net i  found that there are at least two solutions
> > for this, app_rxfax+spandsp  and iaxmodem+hylafax.
> >
> > - what's the differences between these two?
> > - which one's better? why?
> >
> app_rxfax+spandsp fits inside asterisk, and sends and receives TIFF
> files. 3rd party bits and pieces can integrate it with an e-mail
> environment.
>
> iaxmodem+spandsp+hylafax works as a IAX port on a network. It uses the
> same spandsp engine for its front end, but hylafax for its back end. If
> you want to use hylafax clients, this would be the appropriate choice.
> Also, it does a couple of things app_rxfax won't currently do, like
> colour faxing.
>
> Spandsp can do T.38, but not inside Asterisk, and currently not with
> Hylafax. T.38 for Hylafax might be added, but integrating T.38 with a
> class 1 FAX modem interface for Hylafax is an awful bodge. The t38modem
> program from openh323 does this, and it has to do some nasty things to
> work. :-\

...and which method is more reliable and is recommended?

>
> Steve
>
>
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[asterisk-users] Community PBX?

[Kyle Sexton]

I'm wondering if anyone out there is running a community PBX for their local
Asterisk User Groups or area Linux groups.  I've been thinking of setting
one up but am stuck as to what services to provide that people would
actually find useful.  I know that I could setup simple SIP->SIP to allow
everyone to call each other, but that's not generally too fun.

--
Kyle Sexton
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Re: [asterisk-users] Que on A2Billing

[Guillermo Salas M.]

On Fri, 2007-06-15 at 14:20 -0400, Nitesh Divecha wrote:
> You said change the context for SIP Customers to 
> "context=a2billing-did", do I have to create this context or
> A2Billing 
> will generate by itself?
> 


The a2billing package comes with some examples, you must have to create
the a2billing-did context :

[a2billing-did]
exten => _X.,1,NoOp,${CALLERID(all)}
exten => _X.,2,DeadAGI(a2billing.php|1|did)
exten => _X.,3,Hangup()

This will be the context for your DID provider and not for your
customers.

Check this link for more information:

http://forum.asterisk2billing.org/viewtopic.php?t=1784


Cheers!

-- 
Guillermo Salas M.
Telconet S.A.
Calle 15 y Avenida 24 Esq
Edificio Barre #2 Primer Piso
Telefono : +593 5 262 8071
Celular  : +593 9 985 5138
e-mail   : [EMAIL PROTECTED]
www  : http://www.manta.telconet.net
   http://www.telcocarrier.net

Linux User: 255902

Beat me, whip me, make me use Windows!

Please avoid sending me Word or PowerPoint attachments.
See http://www.fsf.org/philosophy/no-word-attachments.html

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http://es.wikipedia.org/wiki/Top-posting


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[asterisk-users] combining AGI with dialplans

[james]

On 2007-05-15 Tony Mountifield wrote (wrt using AGI scripts to dial out):

> Can't comment on this one, as I never use AGI to dial.
> My AGIs just set the context, extension and priority,
> and exit to the dialplan to do any dialling.

(http://article.gmane.org/gmane.comp.telephony.pbx.asterisk.user/185537)


I would like to do this, but I am having trouble figuring out how. I have
tried the following but it is not working for me:


***test.php***
#!/usr/bin/php -q
SetVar("JAMES",$dialstr);
exit(0);
?>

***extensions.conf***
[from-sip]
exten => 111,1,DeadAGI(test.php)
exten => 111,2,Dial(${JAMES})
exten => 111,3,Hangup



Thanks in advance for any help.
James






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[asterisk-users] FXS card with 3-way call, transfer and call waiting.

[Paulo Garcia]

Hi,

I would like to understand how those features (subject) work on fxs ports.
Unfortunately I don't have a digium card with this kind of port, then any
help will be appreciated. I tried to gather some information from google and
this list history, but I still need some help.

3-way-call - As I could understand, when you are talking to A-person, you
can press *flash*, call to B-person and press *flash* again to put all three
in conference. Is it right?

In this case if I drop my call, will all other participants be dropped or
they'll keep talking with each other?

Here I have another doubt. If the participants will keep talking, does this
a extension transfer? How the *transfer* keyword in zapata.conf will affect
this behavior?

Supposing my idea of transferring is right, does this can have conflicts
with #2 feature (attendant transfer) ? Both features (#2 and flash) can live
together?

Thanks in advance!


--

Paulo Garcia
Pika Technologies
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Re: [asterisk-users] Que on A2Billing

[Nitesh Divecha]

Thanks Man...

Do I need to change my context in sip.conf to "context=a2billing" or 
should I leave it to "context=default"?

You said change the context for SIP Customers to 
"context=a2billing-did", do I have to create this context or A2Billing 
will generate by itself?

Cheers,
Nitesh



Guillermo Salas M. wrote:
> On Fri, 2007-06-15 at 12:19 -0400, Nitesh Divecha wrote:
>   
>> Thanks everyone,
>>
>> OK, I got everything working... I manage to create a SIP Customer with a 
>> real DID number and configured an ATA with the DID number. ATA can login 
>> and can make calls out without any issues.
>>
>> But incoming calls are failing... As soon as the call hits Asterisk, 
>> A2Billing script runs and ask for PIN Number... I checked the context 
>> for my DID it shows "context=a2billing" and under sip.conf 
>> "context=a2billing".
>>
>> If I change the default context under sip.conf to "context=default", 
>> then the calls are failing... meaning I do not get any response back, 
>> but on *CLI debug show that its failing to look for the DID number. 
>> Well, I know this is due to my DID is in  "context=a2billing".
>>
>> Anyone can suggest how can I fix this... I want to ring my incoming to 
>> that ATA which has DID assigned.
>> 
>
> You need to setup the DID on the DID section of a2billing.
>
> First create one SIP/IAX2 configuration for your DID provider and assign
> the context a2billing-did.
>
> Later on the DID section, add the DID Provider, add the DID number and
> asign one destination to the DID (your ata card number) or any SIP
> extension enabling the "voip call" radius button.
>
> Try it.
>
> Regards,
>
>
>   


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Re: [asterisk-users] Transfer caller direct to voicemail

[Drew Gibson]

Hi Wes,

thanks for the suggestion but I have gone a simpler route suggested by 
Leonardo Kamache with

exten => _*[1-3]XX,1,Wait(1)
exten => _*[1-3]XX,n,Voicemail(${EXTEN:[EMAIL PROTECTED]|u)
exten => _*[1-3]XX,n,Hangup()

I had assumed the "*" would have been eaten by features in features.conf 
but there is nothing configured there to use "*"!

regards,

Drew


Wes Baehr wrote:
> Drew,
>
> I've written a tiny patch that duplicates the functionality of a blind
> transfer but sets a variable (VMXFER) before transferring. The dialplan
> simply looks for the VMXFER variable, and if found, will direct the call
> directly to voicemail.
>
> This way, instead of making the operators learn they have to push ADDITIONAL
> digits to complete the transfer, they just use a different # sequence. (I
> have mine set up as #8 -- 8 for V)
>
> Email me directly and I can make up a patch for you (for 1.4 anyway).
>
> Wes Baehr
>
>   
>> -Original Message-
>> From: [EMAIL PROTECTED] [mailto:asterisk-users-
>> [EMAIL PROTECTED] On Behalf Of Drew Gibson
>> Sent: Tuesday, June 12, 2007 11:15 AM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: [asterisk-users] Transfer caller direct to voicemail
>>
>> Hi,
>>
>> Our operator frequently gets requests to transfer a call directly to
>> voicemail in order for the caller to leave a message without disturbing
>> the callee. Basicly, I'm looking for a blindxfer to vm.
>>
>> My first thought was to prepend a digit (eg 7) to the extension but this
>> does not fit well with our dialplan.
>>
>> According to an article on voip-info.org [EMAIL PROTECTED] appears to
>> implement this as #*XXX. I assume they are using an application map in
>> features.conf but I cannot see a way to pass the required extension to
>> the VoiceMail() application.
>>
>> Can this be done in features.conf?
>>
>> regards,
>>
>> Drew
>>
>> --
>> Drew Gibson
>>
>> Systems Administrator
>> OANDA Corporation
>> 416-593-6767 x322
>> www.oanda.com
>>
>> 
-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


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Re: [asterisk-users] Recomender Server specs for 250 con-current calls

[Vamsi Pottangi]

Transcoding plays a major role you can find some info @ voip-info ...
http://www.voip-info.org/wiki-Asterisk+dimensioning

~Vamsi

On 6/6/07, Vidura Senadeera <[EMAIL PROTECTED]> wrote:


Dear All,

I looking to implement asterisk solution for 2000 sip registrations and
expecting con-current call about 250.

Can some one provide me guide line that what kind of server will fullfil
the requirment.

what is the Processor, RAM ???

--
Thanks & Regards,
Vidura Senadeera,
Network Engineer,
Debug Solutions
Sri Lanka.
Tel - +94114520036
Mobile - +9466596
Web - www.debug.lk

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Re: [asterisk-users] Que on A2Billing

[Guillermo Salas M.]

On Fri, 2007-06-15 at 12:19 -0400, Nitesh Divecha wrote:
> Thanks everyone,
> 
> OK, I got everything working... I manage to create a SIP Customer with a 
> real DID number and configured an ATA with the DID number. ATA can login 
> and can make calls out without any issues.
> 
> But incoming calls are failing... As soon as the call hits Asterisk, 
> A2Billing script runs and ask for PIN Number... I checked the context 
> for my DID it shows "context=a2billing" and under sip.conf 
> "context=a2billing".
> 
> If I change the default context under sip.conf to "context=default", 
> then the calls are failing... meaning I do not get any response back, 
> but on *CLI debug show that its failing to look for the DID number. 
> Well, I know this is due to my DID is in  "context=a2billing".
> 
> Anyone can suggest how can I fix this... I want to ring my incoming to 
> that ATA which has DID assigned.

You need to setup the DID on the DID section of a2billing.

First create one SIP/IAX2 configuration for your DID provider and assign
the context a2billing-did.

Later on the DID section, add the DID Provider, add the DID number and
asign one destination to the DID (your ata card number) or any SIP
extension enabling the "voip call" radius button.

Try it.

Regards,


-- 
Guillermo Salas M.
Telconet S.A.
Calle 15 y Avenida 24 Esq
Edificio Barre #2 Primer Piso
Telefono : +593 5 262 8071
Celular  : +593 9 985 5138
e-mail   : [EMAIL PROTECTED]
www  : http://www.manta.telconet.net
   http://www.telcocarrier.net

Linux User: 255902

Beat me, whip me, make me use Windows!

Please avoid sending me Word or PowerPoint attachments.
See http://www.fsf.org/philosophy/no-word-attachments.html

Please avoid the Top Posting, see
http://es.wikipedia.org/wiki/Top-posting


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Re: [asterisk-users] app_rxfax vs (iaxmodem+hylafax)

[Rodrigo Gonzalez]

Olivier wrote:
>
> 2007/6/15, Steve Underwood <[EMAIL PROTECTED] 
> >:
>
> ...
>
> The t38modem
> program from openh323 does this, and it has to do some nasty
> things to
> work. :-\
>
> Steve
>
> Is this openh323 project alive ?
> Latest news date from 2003 (http://www.openh323.org/) !
> 
>
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yes, in sourceforge

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Re: [asterisk-users] app_rxfax vs (iaxmodem+hylafax)

[Tzafrir Cohen]

On Fri, Jun 15, 2007 at 05:41:36PM +0200, Olivier wrote:
> 2007/6/15, Steve Underwood <[EMAIL PROTECTED]>:
> >
> >...
> >
> The t38modem
> >program from openh323 does this, and it has to do some nasty things to
> >work. :-\
> >
> >Steve
> >
> >Is this openh323 project alive ?
> Latest news date from 2003 (http://www.openh323.org/) !

Because the newer versions are at http://www.voxgratia.org/

Google still hasn't picked that up, sadly.

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] Que on A2Billing

[Nitesh Divecha]

Thanks everyone,

OK, I got everything working... I manage to create a SIP Customer with a 
real DID number and configured an ATA with the DID number. ATA can login 
and can make calls out without any issues.

But incoming calls are failing... As soon as the call hits Asterisk, 
A2Billing script runs and ask for PIN Number... I checked the context 
for my DID it shows "context=a2billing" and under sip.conf 
"context=a2billing".

If I change the default context under sip.conf to "context=default", 
then the calls are failing... meaning I do not get any response back, 
but on *CLI debug show that its failing to look for the DID number. 
Well, I know this is due to my DID is in  "context=a2billing".

Anyone can suggest how can I fix this... I want to ring my incoming to 
that ATA which has DID assigned.

Cheers,
Nitesh







Guillermo Salas M. wrote:
> On Thu, 2007-06-14 at 14:46 -0400, Nitesh Divecha wrote:
>   
>> Hello All,
>>
>> I got one quick question on A2Billing.
>>
>> Specs: -
>> - A2Billing v1.3
>> - OS CentOS 4.5
>> - Asterisk 1.2
>> - Zaptel 1.2
>>
>> Did the installation and everything is working as it suppose to...
>>
>> Using the A2Billing documentation, I created the RateCard, SIP Trunks, 
>> and SIP Customers. I was also able to login using XLite Dialer and was 
>> able to call out to my SIP Trunk also.
>>
>> Now how can I remove the IVR Prompt... Meaning from my XLite dialer I 
>> want to dial directly and let A2Billing do the billing part. Right now 
>> is something like when I dial any number from XLite, A2Billing script is 
>> invoked and it will announce "You have XXX amount, please enter the 
>> number you wish to call followed by #". And then I have to enter the 
>> number again and then the call is initiated... Its kinda annoying to do 
>> that every time you want to call.
>>
>> Is there anyway to modify config some where, so it will do the billing 
>> in background when the phone call is hangup.
>>
>> 
>
>
> Yes, is possible using the a2billing.conf file in the right way.
>
> I don't have the v1.3 installed, but in the previous release 1.2.3 you
> must have to modify :
>
> use_dnid=YES
> number_try=1
> say_balance_after_auth=NO
> say_balance_after_call=NO
> say_rateinitial=NO
> say_timetocall=NO
>
> Regards,
>
>   


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Re: [asterisk-users] WaitExten not responding on key presses

[Vitell Listmaster]

Jack wrote:
> Hi,
> 
> I have the problem that WaitExten is not responding to key presses.

You haven't answered the call when announce=0 - you need to answer the
call before you can get input information! Change your code to:

[hotline]
exten => _X.,1,Set(CALLERID(name)=Hotline)
exten => _X.,n,Set(original_extension=${EXTEN})
exten => _X.,n,GotoIf($[${announce}=1]?announce:noannounce)
exten => _X.,n(announce),Answer
exten => _X.,n,NoOp(Ansage: Das Problem XYZ ist bereits bekannt und wird
bearbeitet)
exten => _X.,n,NoOp(Ansage: Druecken Sie die Taste 1 falls Sie wegen
einem anderen Problem anrufen)
exten => _X.,n,NoOp(Ansage: Ansonsten druecken Sie eine andere Taste
oder legen Sie bitte auf)
exten => _X.,n,WaitExten(5)
exten => _X.,n,Goto(18)
exten => _X.,n(noannounce),Set(menu=1)
exten => _X.,n,Answer
exten => _X.,n,NoOp(Ansage: Das Gespraech wird aus Qualitaetsgruenden
aufgezeichnet)
exten => _X.,n,NoOp(Ansage: Falls Sie damit nicht einverstanden sind
druecken Sie bitte die Taste 1)
.
.
.

Changed lines marked with , additional line marked with  (note
the change to labels rather than numbers as well - makes it much more
readable!)

Nick.



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Re: [asterisk-users] Scaling Asterisk: Dual-Core CPUs not yielding gains at high call volumes

[Russell Bryant]

Matthew J. Roth wrote:
> In the meantime, I'm looking for insights as to what would cause 
> Asterisk (or any other process) to idle at the same value, despite 
> having similar workloads and twice as many CPUs available to it.  I'll 
> be working on benchmarking Asterisk from very low to very high call 
> volumes so any suggestions or tips, such as how to generate a large 
> number of calls or what statistics I should gather, would also be 
> appreciated.

I am very curious if using this library on your system will help 
increase the load you are able to put on the dual core system.

http://www.hoard.org/

People that are running Asterisk on Solaris have noted that using the 
mtmalloc library allows for much higher call density.  I am hoping that 
hoard will let the people running Asterisk on Linux see similar 
performance improvements, but I have yet to convince anyone to give it a 
try and let me know how it goes.  :)

-- 
Russell Bryant
Software Engineer
Digium, Inc.

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Re: [asterisk-users] app_rxfax vs (iaxmodem+hylafax)

[Olivier]

2007/6/15, Steve Underwood <[EMAIL PROTECTED]>:


...


The t38modem

program from openh323 does this, and it has to do some nasty things to
work. :-\

Steve

Is this openh323 project alive ?

Latest news date from 2003 (http://www.openh323.org/) !
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Re: [asterisk-users] mISDN problem

[Ex Vitorino]

On 6/13/07, Josu Lazkano <[EMAIL PROTECTED]> wrote:
>
> How can I saw the status of the ISDN???
>

  ...try "misdn show stacks" or "misdn show config". You can also increase
  debug level in /etc/misdn-init.conf... Output will end up in
/var/log/asterisk/misdn.log

  Cheers,
--
  Ex Vito

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Re: [asterisk-users] app_rxfax vs (iaxmodem+hylafax)

[Steve Underwood]

Paradise Dove wrote:
> can anybody help me to choose the most reliable fax solution for * .
> after googling the net i  found that there are at least two solutions
> for this, app_rxfax+spandsp  and iaxmodem+hylafax.
>
> - what's the differences between these two?
> - which one's better? why?
>   
app_rxfax+spandsp fits inside asterisk, and sends and receives TIFF 
files. 3rd party bits and pieces can integrate it with an e-mail 
environment.

iaxmodem+spandsp+hylafax works as a IAX port on a network. It uses the 
same spandsp engine for its front end, but hylafax for its back end. If 
you want to use hylafax clients, this would be the appropriate choice. 
Also, it does a couple of things app_rxfax won't currently do, like 
colour faxing.

Spandsp can do T.38, but not inside Asterisk, and currently not with 
Hylafax. T.38 for Hylafax might be added, but integrating T.38 with a 
class 1 FAX modem interface for Hylafax is an awful bodge. The t38modem 
program from openh323 does this, and it has to do some nasty things to 
work. :-\

Steve


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Re: [asterisk-users] g729 codec

[Tzafrir Cohen]

On Fri, Jun 15, 2007 at 09:12:54AM -0400, Kevin Smith wrote:
> Hi everyone,
> 
> Simple question that I haven't been able to find a direct answer to. We 
> currently have call recording with our asterisk system. The files, I am 
> assuming since that is the codec we are using, are being recorded in the 
> g729 codec. Is there a way to listen to these calls, say on windows 
> media player or another audio program? Or do I need to convert the files 
> to a different format to listen to them outside of Asterisk?

ue 'convert' from Asterisk when the load is low?

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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[asterisk-users] can ENUMLOOKUP query multiple DNS servers without having to replicate the same code for each server?

[rjcarvalho]

Hi all,

Does ENUMLOOKUP can query multiple DNS servers without having to  
replicate the same code in which the only thing replaced is the server?


If I use ENUMLOOKUP(${exten}), Asterisk will parse enum.conf file to  
find the list of DNS servers in order of preference to be queried,  
but, I pretend to use something like this:  
${ENUMLOOKUP(+${ARG1:2:},sip,c) which does not seam to care about the  
existence of enum.conf file! May I force Asterisk to care about the  
servers I wrote in enum.conf?


To let you understand better, I wish to use just a block of code that  
is able to query multiple DNS servers, instead of repeating like in  
the following example the same code for each DNS server I wish to  
lookup for:


; Start first with e164.arpa zone:
exten => _X.,1,Set(sipcount=${ENUMLOOKUP(+${EXTEN},sip,c)}|counter=0)
exten => _X.,2,GotoIf($["${counter}"<"${sipcount}"]?3:6)
exten => _X.,3,Set(counter=$[${counter}+1])
exten => _X.,4,Dial(SIP/${ENUMLOOKUP(+${EXTEN},sip,${counter})})
exten => _X.,5,GotoIf($["${counter}"<"${sipcount}"]?3:6)
; ...then also try e164.org:
exten => _X.,6,Set(sipcount=${ENUMLOOKUP(+${EXTEN},sip,c)}|counter=0)
exten => _X.,7,GotoIf($["${counter}"<"${sipcount}"]?8:11)
exten => _X.,8,Set(counter=$[${counter}+1])
exten => _X.,9,Dial(SIP/${ENUMLOOKUP(+${EXTEN},sip,${counter})})
exten => _X.,10,GotoIf($["${counter}"<"${sipcount}"]?8:11)
; ...in case of no route by IP, then send out PRI:
exten => _X.,11,Dial(Zap/g1/${EXTEN})

Regards,
Ricardo.


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Re: [asterisk-users] Run as root?

[gincantalupo]

Hi Malcom,
my advice is to run asterisk as non-privileged user. And do not user 
safe_asterisk script if u can...it cannot check if it running and you 
can have many safe_asterisk running in memory...moreover this fills your 
CLI with a lot of annoying "remote unix connection" messages.

Giorgio


Malcom Kemp wrote:
>
> In looking at the safe_asterisk script, it would appear that it is 
> encouraging the running of the Asterisk application as root user.  My 
> natural inclination is to run it as a non-privileged user.  What is 
> recommendation?
>
> 
>
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Re: [asterisk-users] app_rxfax vs (iaxmodem+hylafax)

[Doug Lytle]

Paradise Dove wrote:
> can anybody help me to choose the most reliable fax solution for * .
> after googling the net i  found that there are at least two solutions
> for this, app_rxfax+spandsp  and iaxmodem+hylafax.
>
> - what's the differences between these two?
> - which one's better? why?
>   

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

Doug

-- 
 
Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety."



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[asterisk-users] g729 codec

[Kevin Smith]

Hi everyone,

Simple question that I haven't been able to find a direct answer to. We 
currently have call recording with our asterisk system. The files, I am 
assuming since that is the codec we are using, are being recorded in the 
g729 codec. Is there a way to listen to these calls, say on windows 
media player or another audio program? Or do I need to convert the files 
to a different format to listen to them outside of Asterisk?

Thanks,
Kevin

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Re: [asterisk-users] Unicall + MFC/R2 line dropped immediately afterconnect

[Peter Gubis]

Hi,

connection is already established (i can also hear called person for a
while). Problem is, that the line is every time dropped after 1 second.
I assume, that first billing pulse arrives immediately after link is
established and it drops the line.

Next week I'll be able to play around with timeouts and will try to
change mfcr2->clear_back_persistence_check parameter as you advised in
your second reply.


Many thanks and best regards,
Peter.


Oscar Carriles wrote:
> Hi,
>
> Clearback signal due to billing pulses normally drops calls after a fixed
> amount of time 2 minutes or so, Can you stablish an outbound call and after
> a while it drops? Or it never succeds?
>
>
> -Mensaje original-
> De: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] En nombre de Peter Gubis
> Enviado el: jueves, 14 de junio de 2007 18:50
> Para: asterisk-users@lists.digium.com
> Asunto: [asterisk-users] Unicall + MFC/R2 line dropped immediately
> afterconnect
>
> Hi,
>
> I am trying to set up an E1 line with CAS signaling using available unicall
> patches with libmfcr2 implementation. Inbound calls works well, I am able to
> get DNIS and ANI from incoming call, but I am still not able to make an
> outbound call with our local carrier.
>
> After tweaking of protocolvariant parameter in unicall.conf I was able to
> find proper values for inbound and outbound calls, but when outbound
> connection is created, it is immediately dropped by opposite side
> immediately after connection is made. I think, that we're receiving billing
> pulses on the line and they're interpreted as a cleaning request.
> I found similar problem in the list before:
> http://lists.digium.com/pipermail/asterisk-users/2006-June/156732.html.
> Steve suggested to use some kind of timeout to workaround this problem.
> I've tried to change many timeouts in libmfcr2 library, without any success.
> Can somebody help me to go around this line drop-down problem?
>
> Relevant part of debug log file:
>
> Jun  9 15:19:27 WARNING[19398]: chan_unicall.c:627 unicall_report: MFC/R2
> UniCall/21  <- 6 off [2/  40/Group I   /DNIS]
> Jun  9 15:19:27 WARNING[19398]: chan_unicall.c:2644 handle_uc_event:
> Unicall/21 event Alerting
> Jun  9 15:19:41 WARNING[19398]: chan_unicall.c:627 unicall_report: MFC/R2
> UniCall/21  <- 0101  [1/ 200/Await answer  /DNIS]
> Jun  9 15:19:41 WARNING[19398]: chan_unicall.c:2644 handle_uc_event:
> Unicall/21 event Connected
>   -- UniCall/21-1 answered UniCall/55-1
> Jun  9 15:19:41 WARNING[19398]: chan_unicall.c:1411 unicall_answer: Answer
> Call Jun  9 15:19:41 WARNING[19398]: chan_unicall.c:627 unicall_report:
> MFC/R2 UniCall/55 Call control(5) Jun  9 15:19:41 WARNING[19398]:
> chan_unicall.c:627 unicall_report: MFC/R2 UniCall/55 Answer call
> Jun  9 15:19:41 WARNING[19398]: chan_unicall.c:627 unicall_report: MFC/R2
> UniCall/55 0101  ->  [1/  20/Group B   /Accepted Paid]
> -- Attempting native bridge of UniCall/55-1 and UniCall/21-1 Jun  9
> 15:19:41 WARNING[19398]: chan_unicall.c:1620 unicall_bridge: unicall_bridge
> called Jun  9 15:19:41 WARNING[19398]: chan_unicall.c:2644 handle_uc_event:
> Unicall/55 event Answered
> Jun  9 15:19:42 WARNING[19398]: chan_unicall.c:627 unicall_report: MFC/R2
> UniCall/21  <- 0001  [1/ 400/Answered  /DNIS]
> Jun  9 15:19:42 WARNING[19398]: chan_unicall.c:627 unicall_report: MFC/R2
> UniCall/21 R2 prot. err. [1/ 400/Answered  /DNIS] cause 32773 -
> Unexpected CAS bit pattern
> Jun  9 15:19:42 WARNING[19398]: chan_unicall.c:627 unicall_report: MFC/R2
> UniCall/21 1001  ->  [1/   1/Idle  /Idle]
> Jun  9 15:19:42 WARNING[19398]: chan_unicall.c:2644 handle_uc_event:
> Unicall/21 event Protocol failure
>  -- Unicall/21 protocol error. Cause 32773 Jun  9 15:19:42 WARNING[19398]:
> chan_unicall.c:627 unicall_report: MFC/R2 UniCall/21 Channel gains Jun  9
> 15:19:42 WARNING[19398]: chan_unicall.c:627 unicall_report: MFC/R2
> UniCall/21 Channel switching
>  -- Hungup 'UniCall/21-1'
>   == Spawn extension (from_merlin, 87, 1) exited non-zero on 'UniCall/55-1'
> Jun  9 15:19:42 WARNING[19398]: chan_unicall.c:627 unicall_report: MFC/R2
> UniCall/55 Channel gains Jun  9 15:19:42 WARNING[19398]: chan_unicall.c:627
> unicall_report: MFC/R2 UniCall/55 Channel switching Jun  9 15:19:42
> WARNING[19398]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/55 Call
> control(6) Jun  9 15:19:42 WARNING[19398]: chan_unicall.c:627
> unicall_report: MFC/R2 UniCall/55 Drop call(cause=Normal Clearing [16])
> Jun  9 15:19:42 WARNING[19398]: chan_unicall.c:627 unicall_report: MFC/R2
> UniCall/55 1101  ->  [1/ 400/Answer/Accepted Paid]
> -- Hungup 'UniCall/55-1'
> Jun  9 15:19:42 WARNING[19390]: chan_unicall.c:627 unicall_report: MFC/R2
> UniCall/21  <- 0101  [1/   1/Idle  /Idle]
> Jun  9 15:19:42 WARNING[19390]: chan_unicall.c:627 unicall_report: MFC/R2
> UniCall/21 R2 prot. err. [1/   1/Idle  /Idle] cause 32773 -

Re: [asterisk-users] Slightly OT:CSU on Digium cards, and it's requirement

[Jeff Davis]

C F wrote:
> No, I installed the system, it goes from smart jack to the PRI card in
> the Panasonic KX-TDA200 thru the 0290 card
> 
>> When I first started working with T1's, most CSU's were external.  I still
>> have several of them in storage in fact... and I still use external
>> CSU/DSU's on my production network today. :-)  I'm typing this message and
>> it will be sent over a T1 connected to 2 external CSU's before it reaches
>> the internet.
>>
>> Bottom line is, no matter what the FCC says... and if somehow you managed
>> to get it to work without a CSU... I believe the phone company would have
>> a fit if they knew you connected equipment to their network without a CSU
>> on it.  They're very big on standards-compliance and stuff like that.
>> Sometime look into their rules and regs about colocating equipment inside
>> one of their CO's...  it's very very strict.
> 
> The last thing you say is why I am asking this question. The
> compliance doesn't realy bother me that much, what I am afraid is if
> the provider notices this and decides to cut it because of that.

Several points:

Your card does not contain a CSU, but you probably already know that. I 
just mention that so that everyone can stop scratching their collective 
heads. There was an FCC requirement for the functions of a CSU, but I 
don't know if it's still in effect or if it matters for you. (see below)

The CSU is as much for your protection as the telco's. The CSU helps to 
isolate you from things like lightning strikes, and electrical hazards.

Most, if not all, of the functions of the CSU are handled by the smart 
jack these days. Line conditioning being the exception. Most smart jacks 
provide loopback, diagnostics, and in some cases line monitoring.

You may not get any help from your provider if you don't have a CSU on 
the line, but you will probably not be cut off since the smart jack is 
providing the functions required. This is a very different situation 
from the past when diagnostic and loopback functions were only available 
through a CSU.

I mentioned line conditioning above. Keep in mind that your service is 
working now, but if conditions should change, and the signal degrades, 
you may experience an outage. It's my understanding that a CSU will help 
prevent that.

So, no, you don't absolutely HAVE to have a CSU these day if the 
conditions are just right, but it's cheap insurance against some types 
of damage and signal loss, and will increase the likely-hood that your 
carrier will actually help you if there's a problem.

--
Jeff Davis
Netsource Consulting

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[asterisk-users] Run as root?

[Malcom Kemp]

In looking at the safe_asterisk script, it would appear that it is
encouraging the running of the Asterisk application as root user.  My
natural inclination is to run it as a non-privileged user.  What is
recommendation?



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Re: [asterisk-users] Slightly OT:CSU on Digium cards, and it's requirement

[C F]

On 6/15/07, Nick Seraphin <[EMAIL PROTECTED]> wrote:
>
> On Thu, 14 Jun 2007, C F wrote:
>
> > > Bottom line is, no matter what the FCC says... and if somehow you managed
> > > to get it to work without a CSU... I believe the phone company would have
> > > a fit if they knew you connected equipment to their network without a CSU
> > > on it.  They're very big on standards-compliance and stuff like that.
> > > Sometime look into their rules and regs about colocating equipment inside
> > > one of their CO's...  it's very very strict.
> >
> > The last thing you say is why I am asking this question. The
> > compliance doesn't realy bother me that much, what I am afraid is if
> > the provider notices this and decides to cut it because of that.
>
>
> Whether they would actually cut you off or not probably depends on A) if
> they find out about it, and B) whoever finds out about it is a strict
> play-by-the-rules kinda guy and/or has a grudge against you or is having a
> bad day.  A lot of telco employees tend to "look the other way"...
> especially if it's not their job to care about it.
>
> But... they would have every right to terminate the service if you don't
> have proper equipment connected to their network.  So if they DID decide
> to terminate it, they would legally have the right to do so, and you would
> have no recourse other than possibly to purchase the correct equipment and
> maybe pay a reconnect fee to get service turned back on, which may take
> days/weeks/whatever time frame to do so.
>
> So it's basically a question of, can you afford the downtime caused by
> them shutting you off if/when they ever found out and/or cared enough to
> "follow the rules".
>
> The other possibility, considering it is working for you now, is that
> there IS a CSU built in but they don't want to tell you... maybe for
> example because it's not FCC certified... or so that they can charge you
> for an external CSU.

It's quite possible that the what you say about FCC certified (or any
other certification on their built in CSU) is why they want you to buy
an external one. They don't make their own (they recommend Adtran
ACE). The other thing, if it would have a CSU it should have LBO
settings and it doesn't.

For the rest, I am going to get external CSUs since I can't afford the downtime.

>
> -- Nick
>
>
>
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Re: [asterisk-users] My Kernel

[Dovid B]

Did you reboot your box after the kernel upgrade ?

- Original Message - 
From: "bilal ghayyad" <[EMAIL PROTECTED]>
To: 
Sent: Friday, June 15, 2007 1:02 AM
Subject: [asterisk-users] My Kernel


Hi List;

I did yum install kernel and yum install kernel-devel,
now when I type 'uname' -a I have the following:

[EMAIL PROTECTED] /]# 'uname' -a
Linux localhost.localdomain 2.6.15-1.2054_FC5smp #1
SMP Tue Mar 14 16:05:46 EST 2006 i686 i686 i386
GNU/Linux

And when I type rpm -q kernel, then I have the
followig:

[EMAIL PROTECTED] /]# rpm - q kernel
kernel-2.6.20-1.2319.fc5

So the question now is: what is my kernel that my
system is using it? And how I can make my system use
the latest updated kernel?

Regards
Bilal




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[asterisk-users] hangup during voicemail announcement drops all calls

[gincantalupo]

Hi,
I'm using Asterisk 1.2.18 on a Debian Etch box.
I have two phones a Gigaset C450IP and a Snom 360. Suppose someone is 
calling the Gigaset phone and a second call comes and is redirected to 
the voicemail: if the new caller hangs up during voicemail announcement, 
Asterisk drops the first call.
This does not happen if the first called party answers the incoming call 
using the SNOM phone.

TIA

Giorgio Incantalupo




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[asterisk-users] Error: Unable to allocate RTCP socket: Too many open files

[Yusuf]

Hi,

I have a Intel Xeon Dual Core server, with 3 GB RAM, running Centos 5.0, 
Asterisk 1.4.4 
and mysql 5.0.  It is a kinda high-traffic box, with about 60 concurrent calls.

The profile of calls on this box are:
Incoming:
via a Sangoma A101
via SIP from anothjer SIP server

Outgoing
all calls that come in are sent out via SIP to yet another SIP server.

This morning I has this error: (edited)

  Executing [EMAIL PROTECTED]:37] Dial("Zap/11-1", 
"SIP/[EMAIL PROTECTED]|40|L(360)") in new stack
 -- Setting call duration limit to 3600 seconds.
 -- Called [EMAIL PROTECTED]
 -- Call on SIP/10.65.138.105-0a67bbd8 left from hold
 -- SIP/10.65.138.105-0a67bbd8 answered SIP/sipCloverCSC-b7eba8a8
 -- Packet2Packet bridging SIP/sipClCSC-b7eba8a8 and 
SIP/10.65.138.105-0a67bbd8
[Jun 15 09:21:48] WARNING[5306]: channel.c:768 ast_channel_alloc: Channel 
allocation 
failed: Can't create alert pipe!
[Jun 15 09:21:48] WARNING[5306]: chan_sip.c:3783 sip_new: Unable to allocate 
AST channel 
structure for SIP channel
[Jun 15 09:21:48] NOTICE[5306]: chan_sip.c:13662 handle_request_invite: Unable 
to 
create/find SIP channel for this INVITE
 -- SIP/iswitch-0a69fb70 is ringing
 -- Call on SIP/iswitch-0a69fb70 left from hold
 -- SIP/iswitch-0a69fb70 is making progress passing it to 
SIP/sipClCSC-b7e2ec78
 -- Call on SIP/iswitch-0a569528 left from hold
 -- SIP/iswitch-0a569528 answered Zap/9-1
[Jun 15 09:21:49] WARNING[5306]: channel.c:768 ast_channel_alloc: Channel 
allocation 
failed: Can't create alert pipe!
[Jun 15 09:21:49] WARNING[5306]: chan_sip.c:3783 sip_new: Unable to allocate 
AST channel 
structure for SIP channel
[Jun 15 09:21:49] NOTICE[5306]: chan_sip.c:13662 handle_request_invite: Unable 
to 
create/find SIP channel for this INVITE
 -- SIP/10.65.138.103-0a8c4000 is ringing
 -- Call on SIP/10.65.138.103-0a8c4000 left from hold
 -- SIP/10.65.138.103-0a8c4000 is making progress passing it to 
SIP/sipClCSC-b7e62f28
 -- SIP/10.65.138.103-0a8c4000 is ringing
 -- Call on SIP/10.65.138.103-0a8c4000 left from hold
 -- SIP/10.65.138.103-0a8c4000 is making progress passing it to 
SIP/sipClCSC-b7e62f28
 -- Call on SIP/10.65.138.103-0a8c4000 left from hold
 -- SIP/10.65.138.103-0a8c4000 answered SIP/sipCloverCSC-b7e62f28
 -- Packet2Packet bridging SIP/sipCloverCSC-b7e62f28 and 
SIP/10.65.138.103-0a8c4000
   == Spawn extension (iaxClover, 0722269331, 37) exited non-zero on 
'SIP/sipClCSC-b7e4cd58'

 -- Executing [EMAIL PROTECTED]:52] GotoIf("Zap/1-1", "0 ? 60") in new stack
 -- Executing [EMAIL PROTECTED]:53] Dial("Zap/1-1", 
"SIP/iswitch/27117973000|40|L(360)") in new stack
 -- Setting call duration limit to 3600 seconds.
 -- Called iswitch/27117973000
[Jun 15 09:22:04] WARNING[5306]: acl.c:378 ast_ouraddrfor: Cannot create socket
[Jun 15 09:22:04] ERROR[5306]: rtp.c:1861 ast_rtp_new_with_bindaddr: Unable to 
allocate 
socket: Too many open files
[Jun 15 09:22:04] WARNING[5306]: chan_sip.c:4242 sip_alloc: Unable to create 
RTP audio 
session: Too many open files
[Jun 15 09:22:04] WARNING[5306]: acl.c:378 ast_ouraddrfor: Cannot create socket
[Jun 15 09:22:05] WARNING[5306]: acl.c:378 ast_ouraddrfor: Cannot create socket
[Jun 15 09:22:05] ERROR[5306]: rtp.c:1861 ast_rtp_new_with_bindaddr: Unable to 
allocate 
socket: Too many open files
[Jun 15 09:22:05] WARNING[5306]: chan_sip.c:4242 sip_alloc: Unable to create 
RTP audio 
session: Too many open files
[Jun 15 09:22:05] WARNING[5306]: acl.c:378 ast_ouraddrfor: Cannot create socket
[Jun 15 09:22:06] WARNING[5306]: acl.c:378 ast_ouraddrfor: Cannot create socket
[Jun 15 09:22:06] ERROR[5306]: rtp.c:1861 ast_rtp_new_with_bindaddr: Unable to 
allocate 
socket: Too many open files


So I stopped Asterisk.  I am going to

increase the ulimit,
also increasing the RTP range, from the default of 1 - 2.
I had SElinux on permissive, should I rather just disable it?

Can anyone give me pointers as to what has gone wrong, and what I can do, other 
than the 
above to fix it?

Also, as as aside, what it Packet2PAcket? Reading some of Olle's posts, I 
gather there is 
two types of brigding technologies?












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Re: [asterisk-users] Sending text to a phone that's no in-use ...

[Gordon Henderson]

On Fri, 15 Jun 2007, Philipp von Klitzing wrote:

> Hi!
>
 Anyone know if it's possible to send a line of text to a phone that's
 not currently in-use?

 What I want is:
   SendText(SIP/101, "Hello World")
 but that doesn't exist ...
>>
>> Snom's or Grandstream GXP2000's I'm afraid... Sending text to them while
>> in a call works fine (although reading the text on the Snom was
>> "challenging" until I'd worked out I needed to reprogram one of the
>> function keys :)
>
> Read the part about ".call" file here:
> http://www.voip-info.org/wiki/view/Asterisk+cmd+SendText
>
> Or look at the FAQ at the bottom of this page, look for "sipsak":
> http://www.voip-info.org/wiki/index.php?page=Asterisk+phone+snom
>
> Concept: Configure an auto-answer line on your phone and use a .call file
> together with SendText(),

I'd loked at that and thought "Hmm..." and wondered if there was a "newer" 
way...

> or use sipsak instead (i.e. external tool,
> outside of Asterisk).

And I must have glossed over that...

Job done:

   sipsak -M -v -s sip:[EMAIL PROTECTED] -B "Lunch time!"
   warning: ignoring -i option when in usrloc mode
   usrloc for 108 completed successful

   All usrloc tests completed successful.
   received last message 4.696 ms after first request (test duration).

At least for Grandstream phones - the Snom300 I have rejects it, but I'm 
sure it's just a matter of reading the manual now ;-)

And I can probably extract the IP addresses of each phone out of the astdb 
without too much bother too. (everything here on the same LAN, so not much 
of an issue)

Thanks!

> Third option if you have a SNOM 360 or 370: Send a SIP NOTIFY and push a
> XML message along with that (mini-browser style). Read more about this on
> the SNOM knowledgebase/wiki.

Just Snom300's for now.

Cheers,

Gordon

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