Re: [asterisk-users] Whither asterisk-addons?

[Chris Brentano]

Correction, I did notice it for download at 
http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-addons-1.6.1.1.tar.gz

- Chris

On 16 Oct, 2009, at 4:06 PM, Chris Brentano wrote:

> I noticed that asterisk.org got a redesign, quite recently it seems,
> which is very nice, but the addons package isn't listed for download
> any longer, nor are releases posted to 
> http://downloads.asterisk.org/pub/telephony/
>
>
> That said, looks like it's still available in svn, 
> http://svnview.digium.com/svn/asterisk-addons/tags/1.6.1.1/
>
>
> So just wondering if addons will be around for the forseeable future?
>
> - Chris
>
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[asterisk-users] Nehalem & Digium Wildcard issues?

[Chris Brentano]

Just putting this out there to see if anyone else has seen any issues.  
May cross-post to asterisk-dev if it's indeed a bug (and not my own  
stupidity).

I've got a Digium TE220 (2xT1 interface w/Echo canceller) that in two  
separate Nehalem-based (Xeon E5520 "Gainestown") boxes (HP ProLiant  
ML350 G6; HP Z800 Workstation) has caused numerous kernel panics. This  
is only when the dahdi service is running with a very simple config  
(I've defined the first span, the bchans and the dchan, and that's  
about it). If dahdi is stopped, or the card is removed, everything's  
fine. I instead installed a Digium TE122P in the ML350 and haven't had  
any issues. I also haven't seen this in a pre-Nehalem Xeon server.

I'm using Asterisk 1.6.1.6, Dahdi 2.2.0 and LibPRI 1.4.10.1, running  
on CentOS 5.3 (2.6.18-164.el5).

Has anyone seen anything similar?

- Chris

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[asterisk-users] IVR

[Nazir Ahmed Vaid]

Ladies and Gentlemen,
We already have an Asterisk Call center suite installed at our contact
center. Now we wish to commence IVR services. We are offering Health
Information Services. Can someone help us to develop this Addon / Solution?

Best regards.

-- 
السلام عليكم ورحمة الله وبركاته


Nazir Ahmed Vaid
Cell:+92300-828

eHealth Services (Pvt) Ltd.
http://www.ehealth-services.com

NexSource Pakistan (Pvt) Ltd.
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[asterisk-users] Whither asterisk-addons?

[Chris Brentano]

I noticed that asterisk.org got a redesign, quite recently it seems,  
which is very nice, but the addons package isn't listed for download  
any longer, nor are releases posted to 
http://downloads.asterisk.org/pub/telephony/ 
.

That said, looks like it's still available in svn, 
http://svnview.digium.com/svn/asterisk-addons/tags/1.6.1.1/ 
.

So just wondering if addons will be around for the forseeable future?

- Chris

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Re: [asterisk-users] Linux/Asterisk on game consoles?

[C. Chad Wallace]

At 4:48 PM on 16 Oct 2009, Adam Moffett wrote:

> Out of curiosity why would you want to?

Because he can?  or, "Because it's there."
http://www.askoxford.com/worldofwords/quotations/quotefrom/mallory/

...but hopefully the OP doesn't end up bricking his console.


> > I don't know much about game consoles, and I was wondering if
> > someone had successfully ported Linux and Asterisk to the current
> > hardware, ie. Nintendo Wii, Sony PS3, or Microsoft XBox360?


-- 

C. Chad Wallace, B.Sc.
The Lodging Company
http://www.skihills.com/
OpenPGP Public Key ID: 0x262208A0



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Re: [asterisk-users] Linux/Asterisk on game consoles?

[Adam Moffett]

Out of curiosity why would you want to?

> Hello
>
> I don't know much about game consoles, and I was wondering if someone
> had successfully ported Linux and Asterisk to the current hardware,
> ie. Nintendo Wii, Sony PS3, or Microsoft XBox360?
>
> Thank you.
>
>
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>   


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Re: [asterisk-users] SIP to IAX to SIP

[Ivan Stepaniuk]

George Farris wrote:
> I have a machine running Ubuntu that I run Asterisk 1.4.x on and it runs
> very well.  On that machine I have a SIP phone.  I have configured a
> netgear wgt634u with asterisk and a SIP phone and linked the two systems
> together via IAX.  Audio from Ubuntu to netgear is not bad, audio from
> netgear to ubuntu is unintelligible.  Any clues as to whether this will
> work?  Configuration suggestions?  Is a 200MHz arm processor just too
> small?
what codec are you using?

-- 
Iván Stepaniuk
Alba Fotónica S. L.
www.albafotonica.com


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[asterisk-users] SIP to IAX to SIP

[George Farris]

Hi all,

I have a machine running Ubuntu that I run Asterisk 1.4.x on and it runs
very well.  On that machine I have a SIP phone.  I have configured a
netgear wgt634u with asterisk and a SIP phone and linked the two systems
together via IAX.  Audio from Ubuntu to netgear is not bad, audio from
netgear to ubuntu is unintelligible.  Any clues as to whether this will
work?  Configuration suggestions?  Is a 200MHz arm processor just too
small?

Any help appreciated.


sip phone <--> wgt634u  <- iax -> ubuntu <--> sip phone
wireless   200MHz arm 3GHz AMDhardwired 100MB

 100MB lan between systems

Hope this is clear enough.


Cheers
George



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Re: [asterisk-users] Queues with unavailable members

[C. Chad Wallace]

At 7:35 PM on 16 Oct 2009, Benny Amorsen wrote:

> "C. Chad Wallace"  writes:
> 
> > Also, if there is another agent available, the caller would be
> > connected immediately, and it wouldn't have to make any more
> > attempts.  With the Wait() solution, that caller would be waiting
> > for 30 seconds regardless of whether there's anyone else
> > available.  
> 
> This bit is solved by the ringall strategy.
> 
> > Of course, I don't know your business case, so you'll have to decide
> > which of the two problems is worse.
> 
> I'm fairly happy with the Wait(1000) solution for now. We'll see if
> testing shows any problems with it.

Oh yeah, I hadn't even considered the ringall strategy!  With that,
your Wait() solution sounds perfect to me.  Congrats!

-- 

C. Chad Wallace, B.Sc.
The Lodging Company
http://www.skihills.com/
OpenPGP Public Key ID: 0x262208A0



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Re: [asterisk-users] can i use Asterisk to send sms to my databaseusers?

[Danny Nicholas]

Absolutely.  Just set up the phone address (201...@cingular.net) in
users.conf and restart.  I'm not sure what apps will send besides voicemail,
but then that wasn't the question.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of kazabe
Sent: Friday, October 16, 2009 1:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] can i use Asterisk to send sms to my
databaseusers?

Hi.

that is the question.  I need send periodically some sms messages to
my users, stored in a database.  We are doing that process with a cell
service provider, but i wanna know if i can use my own server to do
that.

Any sugestion?

thanks in advance.

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[asterisk-users] can i use Asterisk to send sms to my database users?

[kazabe]

Hi.

that is the question.  I need send periodically some sms messages to
my users, stored in a database.  We are doing that process with a cell
service provider, but i wanna know if i can use my own server to do
that.

Any sugestion?

thanks in advance.

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Re: [asterisk-users] MWI for multiple voice mail boxes

[Danny Nicholas]

Preparing for the lightning bolt here (ready to duck!!), the way I have
things set up, tkeeley would have an entry in users.conf as 612 and 610
would have an entry in users.conf as 610.  There would be an entry in
sip.conf for tkeeley under 612 and no entry for 610 since it's just a
mailbox and not a physical extension.  Not necessarily best or even correct,
just works for me.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A.
Sullivan III
Sent: Friday, October 16, 2009 12:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] MWI for multiple voice mail boxes

No, probably my ignorance but why would I do that? I set up all the
users, extensions, and mailboxes manually by editing the config files in
order to have more control than the user.conf gives me (if I understand
the user.conf file properly - I've never used it based upon reading the
documentation).  Thanks - John

On Fri, 2009-10-16 at 12:07 -0500, Danny Nicholas wrote:
> I assume you have a 610 entry in users.conf?
> 
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A.
> Sullivan III
> Sent: Friday, October 16, 2009 12:05 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] MWI for multiple voice mail boxes
> 
> Alas, it does not work for me on 1.6.1.6.  That was my original
> configuration based upon the documentation.  It was slightly different
> than you have because I specified the context.  tkeeley is in context
> a10f but the mailboxes are in context a10. Thus, I had:
> 
> [tkeeley]
> mailbox=...@a10, 6...@a10
> 
> It then complains that it cannot find mailox 610 in context a10.
> However, it is there and it does receive voice mail.  Thanks - John
> 
> On Fri, 2009-10-16 at 10:14 -0500, Danny Nicholas wrote:
> > Let's stick a fork in this one - 
> > Here's the link I used
> > http://www.voip-info.org/wiki/view/Asterisk+sip+mailbox
> > 
> > if we make tkeely's sip.conf look like this
> > [tkeeley]
> > Type=peer
> > Context=a10
> > Mailbox=612, 610
> > 
> > He? Should be good to go.
> > 
> > This worked on 1.4.26.1
> > 
> > -Original Message-
> > From: asterisk-users-boun...@lists.digium.com
> > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A.
> > Sullivan III
> > Sent: Thursday, October 15, 2009 8:14 PM
> > To: jsm...@digium.com; Asterisk Users Mailing List - Non-Commercial
> > Discussion
> > Subject: Re: [asterisk-users] MWI for multiple voice mail boxes
> > 
> > On Thu, 2009-10-15 at 15:29 -0700, Jared Smith wrote:
> > > On Wed, 2009-10-14 at 22:41 -0400, John A. Sullivan III wrote:
> > > > Hello, all.  I have a user who needs to monitor their voice mail box
> > > > and
> > > > the general delivery voice mail box.  I defined them in sip.conf as
> > > > follows:
> > > > 
> > > > [tkeeley](a10f)
> > > > mailbox=...@a10, 6...@a10 
> > > 
> > > I think you've got the syntax wrong here... try
mailbox=...@a10&6...@a10
> > > instead.  Contrary to what others on this thread might lead you to
> > > believe, this should actually work. :-)
> > 
> > O - it really didn't like that:
> > 
> > mailbox=...@a10&6...@a10
> > 
> > app_voicemail.c:1630 messagecount: Couldn't find mailbox 612 in context
> > a10&6...@a10
> > 
> > It looks like it's interpreting everything after the @ as context.  I'm
> > running 1.6.1.6.  Thanks anyway - John
-- 
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com

http://www.spiritualoutreach.com
Making Christianity intelligible to secular society


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Re: [asterisk-users] Where to find IMAP storage doc ?

[Noah Miller]

>> We're also working fine with it but I also do not know what the
>> available imapflags are and what they mean. I have seen notls and
>> novalidatecert.  Out of curiosity, I spent the last 20 minutes googling
>> for information on c-client imapflags and didn't find any definitions or
>> even a simple list, either.  There is a list of flags in the c-client
>> man page but they seem to be a different set of flags.  Let me know what
>> you find as I would like to know what functionality and options they
>> give us.

I'd recommend compiling c-client from source.  I've never run Lenny
before, but I had a number of issues with various pre-compiled
versions of c-client.  I feel your pain on lack of documentation for
compiling from source, though.  The magic steps for me on CentOS were:

1. Modify the EXTRACFLAGS line of the uw-imap makefile:

EXTRACFLAGS=-DDISABLE_POP_PROXY=1 -DIGNORE_LOCK_EACCES_ERRORS=1
-I/usr/include/openssl -fPIC -fno-strict-aliasing -Wall
-Wno-pointer-sign -Wno-parentheses

(I think this is all I had to modify, but I can send you my complete
working Makefile, if you like).

2. Compile for your platform:

For lenny, I think it would be:
make ldb

3. For asterisk, manually configure the location of uw-imap:

./configure --with-imap=/path/to/imap


- Noah

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Re: [asterisk-users] Queues with unavailable members

[Benny Amorsen]

"C. Chad Wallace"  writes:

> It would only be trying one agent at a time for each waiting queue
> member...

Would it? Almost all our queues are on a ringall strategy.

> I don't know how expensive it is to open and close a Local channel and
> do a DB lookup, but I wouldn't expect it to be a real problem. You are
> at least avoiding multiple calls out to the cellular network.

Not that expensive, but still a bit of a waste when done every couple of
seconds. Especially if multiple agents are unavailable.

> Also, if there is another agent available, the caller would be connected
> immediately, and it wouldn't have to make any more attempts.  With the
> Wait() solution, that caller would be waiting for 30 seconds regardless
> of whether there's anyone else available.  

This bit is solved by the ringall strategy.

> Of course, I don't know your business case, so you'll have to decide
> which of the two problems is worse.

I'm fairly happy with the Wait(1000) solution for now. We'll see if
testing shows any problems with it.


/Benny

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Re: [asterisk-users] Linux/Asterisk on game consoles?

[Noah Miller]

>> I don't know much about game consoles, and I was wondering if someone
>> had successfully ported Linux and Asterisk to the current hardware,
>> ie. Nintendo Wii, Sony PS3, or Microsoft XBox360?

The Xbox is an x86 machine, so running linux and/or asterisk on it
should not be too difficult.  There's even a not-so-difficult method
of adding a USB port, which would allow you to attach Xorcom hardware
for PSTN connections.

The Xbox360 is a PowerPC machine.  I don't know what the status of
having it run *nix is, but there's a site dedicated to it here:
http://www.free60.org

For the wii, there's: http://wiibrew.org/wiki/Wii_Linux


- Noah

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Re: [asterisk-users] MWI for multiple voice mail boxes

[John A. Sullivan III]

No, probably my ignorance but why would I do that? I set up all the
users, extensions, and mailboxes manually by editing the config files in
order to have more control than the user.conf gives me (if I understand
the user.conf file properly - I've never used it based upon reading the
documentation).  Thanks - John

On Fri, 2009-10-16 at 12:07 -0500, Danny Nicholas wrote:
> I assume you have a 610 entry in users.conf?
> 
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A.
> Sullivan III
> Sent: Friday, October 16, 2009 12:05 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] MWI for multiple voice mail boxes
> 
> Alas, it does not work for me on 1.6.1.6.  That was my original
> configuration based upon the documentation.  It was slightly different
> than you have because I specified the context.  tkeeley is in context
> a10f but the mailboxes are in context a10. Thus, I had:
> 
> [tkeeley]
> mailbox=...@a10, 6...@a10
> 
> It then complains that it cannot find mailox 610 in context a10.
> However, it is there and it does receive voice mail.  Thanks - John
> 
> On Fri, 2009-10-16 at 10:14 -0500, Danny Nicholas wrote:
> > Let's stick a fork in this one - 
> > Here's the link I used
> > http://www.voip-info.org/wiki/view/Asterisk+sip+mailbox
> > 
> > if we make tkeely's sip.conf look like this
> > [tkeeley]
> > Type=peer
> > Context=a10
> > Mailbox=612, 610
> > 
> > He? Should be good to go.
> > 
> > This worked on 1.4.26.1
> > 
> > -Original Message-
> > From: asterisk-users-boun...@lists.digium.com
> > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A.
> > Sullivan III
> > Sent: Thursday, October 15, 2009 8:14 PM
> > To: jsm...@digium.com; Asterisk Users Mailing List - Non-Commercial
> > Discussion
> > Subject: Re: [asterisk-users] MWI for multiple voice mail boxes
> > 
> > On Thu, 2009-10-15 at 15:29 -0700, Jared Smith wrote:
> > > On Wed, 2009-10-14 at 22:41 -0400, John A. Sullivan III wrote:
> > > > Hello, all.  I have a user who needs to monitor their voice mail box
> > > > and
> > > > the general delivery voice mail box.  I defined them in sip.conf as
> > > > follows:
> > > > 
> > > > [tkeeley](a10f)
> > > > mailbox=...@a10, 6...@a10 
> > > 
> > > I think you've got the syntax wrong here... try mailbox=...@a10&6...@a10
> > > instead.  Contrary to what others on this thread might lead you to
> > > believe, this should actually work. :-)
> > 
> > O - it really didn't like that:
> > 
> > mailbox=...@a10&6...@a10
> > 
> > app_voicemail.c:1630 messagecount: Couldn't find mailbox 612 in context
> > a10&6...@a10
> > 
> > It looks like it's interpreting everything after the @ as context.  I'm
> > running 1.6.1.6.  Thanks anyway - John
-- 
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com

http://www.spiritualoutreach.com
Making Christianity intelligible to secular society


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Re: [asterisk-users] Queues with unavailable members

[C. Chad Wallace]

At 11:23 AM on 16 Oct 2009, Benny Amorsen wrote:

> I was going in the same direction at the end of my first mail, but I
> hadn't written any code. There is a problem though: The Queue
> application will keep sending calls to the Local channel, which have
> to be rejected, over and over.
> 
> Would it perhaps work to simply Wait(30) if the call is rejected by
> the phone? If the Queue assumes that the phone is busy for those 30
> seconds, I have accomplished my goal. It's worth a shot.

It would only be trying one agent at a time for each waiting queue
member...  I don't know how expensive it is to open and close a Local
channel and do a DB lookup, but I wouldn't expect it to be a real
problem.  You are at least avoiding multiple calls out to the cellular
network.  

Also, if there is another agent available, the caller would be connected
immediately, and it wouldn't have to make any more attempts.  With the
Wait() solution, that caller would be waiting for 30 seconds regardless
of whether there's anyone else available.  

Of course, I don't know your business case, so you'll have to decide
which of the two problems is worse.

-- 

C. Chad Wallace, B.Sc.
The Lodging Company
http://www.skihills.com/
OpenPGP Public Key ID: 0x262208A0



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Re: [asterisk-users] MWI for multiple voice mail boxes

[Danny Nicholas]

I assume you have a 610 entry in users.conf?

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A.
Sullivan III
Sent: Friday, October 16, 2009 12:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] MWI for multiple voice mail boxes

Alas, it does not work for me on 1.6.1.6.  That was my original
configuration based upon the documentation.  It was slightly different
than you have because I specified the context.  tkeeley is in context
a10f but the mailboxes are in context a10. Thus, I had:

[tkeeley]
mailbox=...@a10, 6...@a10

It then complains that it cannot find mailox 610 in context a10.
However, it is there and it does receive voice mail.  Thanks - John

On Fri, 2009-10-16 at 10:14 -0500, Danny Nicholas wrote:
> Let's stick a fork in this one - 
> Here's the link I used
> http://www.voip-info.org/wiki/view/Asterisk+sip+mailbox
> 
> if we make tkeely's sip.conf look like this
> [tkeeley]
> Type=peer
> Context=a10
> Mailbox=612, 610
> 
> He? Should be good to go.
> 
> This worked on 1.4.26.1
> 
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A.
> Sullivan III
> Sent: Thursday, October 15, 2009 8:14 PM
> To: jsm...@digium.com; Asterisk Users Mailing List - Non-Commercial
> Discussion
> Subject: Re: [asterisk-users] MWI for multiple voice mail boxes
> 
> On Thu, 2009-10-15 at 15:29 -0700, Jared Smith wrote:
> > On Wed, 2009-10-14 at 22:41 -0400, John A. Sullivan III wrote:
> > > Hello, all.  I have a user who needs to monitor their voice mail box
> > > and
> > > the general delivery voice mail box.  I defined them in sip.conf as
> > > follows:
> > > 
> > > [tkeeley](a10f)
> > > mailbox=...@a10, 6...@a10 
> > 
> > I think you've got the syntax wrong here... try mailbox=...@a10&6...@a10
> > instead.  Contrary to what others on this thread might lead you to
> > believe, this should actually work. :-)
> 
> O - it really didn't like that:
> 
> mailbox=...@a10&6...@a10
> 
> app_voicemail.c:1630 messagecount: Couldn't find mailbox 612 in context
> a10&6...@a10
> 
> It looks like it's interpreting everything after the @ as context.  I'm
> running 1.6.1.6.  Thanks anyway - John
-- 
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com

http://www.spiritualoutreach.com
Making Christianity intelligible to secular society


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Re: [asterisk-users] MWI for multiple voice mail boxes

[John A. Sullivan III]

Alas, it does not work for me on 1.6.1.6.  That was my original
configuration based upon the documentation.  It was slightly different
than you have because I specified the context.  tkeeley is in context
a10f but the mailboxes are in context a10. Thus, I had:

[tkeeley]
mailbox=...@a10, 6...@a10

It then complains that it cannot find mailox 610 in context a10.
However, it is there and it does receive voice mail.  Thanks - John

On Fri, 2009-10-16 at 10:14 -0500, Danny Nicholas wrote:
> Let's stick a fork in this one - 
> Here's the link I used
> http://www.voip-info.org/wiki/view/Asterisk+sip+mailbox
> 
> if we make tkeely's sip.conf look like this
> [tkeeley]
> Type=peer
> Context=a10
> Mailbox=612, 610
> 
> He? Should be good to go.
> 
> This worked on 1.4.26.1
> 
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A.
> Sullivan III
> Sent: Thursday, October 15, 2009 8:14 PM
> To: jsm...@digium.com; Asterisk Users Mailing List - Non-Commercial
> Discussion
> Subject: Re: [asterisk-users] MWI for multiple voice mail boxes
> 
> On Thu, 2009-10-15 at 15:29 -0700, Jared Smith wrote:
> > On Wed, 2009-10-14 at 22:41 -0400, John A. Sullivan III wrote:
> > > Hello, all.  I have a user who needs to monitor their voice mail box
> > > and
> > > the general delivery voice mail box.  I defined them in sip.conf as
> > > follows:
> > > 
> > > [tkeeley](a10f)
> > > mailbox=...@a10, 6...@a10 
> > 
> > I think you've got the syntax wrong here... try mailbox=...@a10&6...@a10
> > instead.  Contrary to what others on this thread might lead you to
> > believe, this should actually work. :-)
> 
> O - it really didn't like that:
> 
> mailbox=...@a10&6...@a10
> 
> app_voicemail.c:1630 messagecount: Couldn't find mailbox 612 in context
> a10&6...@a10
> 
> It looks like it's interpreting everything after the @ as context.  I'm
> running 1.6.1.6.  Thanks anyway - John
-- 
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com

http://www.spiritualoutreach.com
Making Christianity intelligible to secular society


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[asterisk-users] OT Old Sipura OK - Linksys (junk)

[Joseph]

I have two old Sipura 3000 original (green) boxes and they are running 
perfectly faxes going through, no echo.
Firmware 2.0.13(GWg), hardware  2.0.1(1813)
Does the number in the bracket means anything? 

In addition I have Linksys 3000 running the same firmware and same hardware 
(except the number in the bracket).
I can not get to work correctly.  The unit has a lot of echo, can not get rid 
of it, faxes will not go through.
Newer firmware 3.1.10 is buggy so I'll not try it; I've loaded 3.1.20 and it 
will not work correctly can not dial out without on default setting (have to 
set 
the SPA to PSTN high).
So in other words Linksys unit (at least this model) is a piece of junk.

Does anybody runs Linksys 3102 with Cisco firmware 5.1.10?  How do they run?

I have two units Linksys 3102 with firmware 5.1.7(GW) hardware: 1.4.5(a) 
and I'm not impress with them either, lot of echo as well. 
Maybe that is why they sold the unit to Cisco as they can not make it to work 
correctly :-/

Any recommendation where to move from here, which unit to buy for use with 
Asterisk? 

-- 
Joseph

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[asterisk-users] Origin of "Exceptionally long voice queue length queuing to IAX2/blahblah" messages

[daren ferreira]

Hello,

I'm using asterisk for a quite long period, i integrated a lot of stuff to make 
it behave like any carrier class system, so users can:

manage Call forward on busy
manage Call forward on no answer
manage unconditionnal call forward
call back missed calls


and a lot of such services that can be both configured by menus and dtmf or by 
web interfaces (PHP+SQL), in fact i recreated services everybody use to get 
with their classical operator.

All of theses functions are achieved throught dialplan commands, first MYSQL 
commands, conditions on results etc..

It worked as a charm but got wrong with the growing number of users, i began 
having "asterisk network connectivity loss", in fact qualified IAX2 friends 
became "UNREACHABLE" for seconds, connected calls were disturbed when not 
simply cut.

I first thought of IAX problems on charge, moved to SIP, but got the same 
problems, i then moved my MYSQL commands to ODBC functions, optimized my 
macros, it worked for a short while, i changed dialplan to reduce as much as 
possible LOCAL channels, it worked for a while but now the problem comes back 
with its IAX messages:

[Oct 16 17:55:40] WARNING[2514]: channel.c:951 __ast_queue_frame: Exceptionally 
long voice queue length queuing to IAX2/blahblah
[Oct 16 17:55:40] WARNING[2511]: channel.c:951 __ast_queue_frame: Exceptionally 
long voice queue length queuing to IAX2/blahblah
[Oct 16 17:57:38] NOTICE[2513]: chan_iax2.c:9186 __iax2_poke_noanswer: Peer 
'XXX' is now UNREACHABLE! Time: 9
[Oct 16 17:57:38] NOTICE[2505]: chan_iax2.c:9186 __iax2_poke_noanswer: Peer 
'YYY' is now UNREACHABLE! Time: 9
[Oct 16 17:57:55] WARNING[2505]: chan_iax2.c:804 jb_warning_output: Resyncing 
the jb. last_delay 0, this delay 22334, threshold 1000, new offset -22334
[Oct 16 17:57:57] WARNING[2510]: chan_iax2.c:804 jb_warning_output: Resyncing 
the jb. last_delay -135, this delay 1529, threshold 1000, new offset -1529
[Oct 16 17:58:05] NOTICE[2507]: chan_iax2.c:8264 socket_process: Peer 'XXX' is 
now REACHABLE! Time: 4
[Oct 16 17:58:07] NOTICE[2511]: chan_iax2.c:8264 socket_process: Peer 'YYY' is 
now REACHABLE! Time: 11

On some servers i used less advanced dialplan, with ODBC functions but geeting 
rid of LOCAL channels, and, until now, i didn't got any problem on them... so i 
suspect LOCAL channels, without proof of it, but what else? And for my advanced 
services, it will be hard to get rid of them without scripts so, i'm 
actually thinking of moving my dialplan commands to AGI script(s), so i may get 
rid of LOCAL channels and manage the processes more easily, but reading wikis 
and mailing lists, some of them consider dialplan commands better than scripts 
for performances... but if so, why are AGI used by so much projects 
(asterbilling or ccard systems), which seems to support very large amount of 
users

So, if somebody on the list has any idea or advice... please let me know...


Regards

Daren
  
_
A la recherche de bons plans pour une rentrée pas chère ? Bing ! Trouvez !
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Re: [asterisk-users] OT wanted old Sipura firmware 2.0.13

[Joseph]

On 10/16/09 10:23, Dave Fullerton wrote:
>Joseph wrote:
>> Does anybody know here I can find old Sipura firmware 2.0.13 for SPA-3000
>> I have Cisco 3.1.20 but it is not working as it suppose to.
>>
>
>
>http://www.totek.ca/index.php?option=com_content&task=view&id=151&Itemid=39

Thank you, that is a golden link.

-- 
Joseph

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Re: [asterisk-users] MWI for multiple voice mail boxes

[Danny Nicholas]

Let's stick a fork in this one - 
Here's the link I used
http://www.voip-info.org/wiki/view/Asterisk+sip+mailbox

if we make tkeely's sip.conf look like this
[tkeeley]
Type=peer
Context=a10
Mailbox=612, 610

He? Should be good to go.

This worked on 1.4.26.1

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A.
Sullivan III
Sent: Thursday, October 15, 2009 8:14 PM
To: jsm...@digium.com; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [asterisk-users] MWI for multiple voice mail boxes

On Thu, 2009-10-15 at 15:29 -0700, Jared Smith wrote:
> On Wed, 2009-10-14 at 22:41 -0400, John A. Sullivan III wrote:
> > Hello, all.  I have a user who needs to monitor their voice mail box
> > and
> > the general delivery voice mail box.  I defined them in sip.conf as
> > follows:
> > 
> > [tkeeley](a10f)
> > mailbox=...@a10, 6...@a10 
> 
> I think you've got the syntax wrong here... try mailbox=...@a10&6...@a10
> instead.  Contrary to what others on this thread might lead you to
> believe, this should actually work. :-)

O - it really didn't like that:

mailbox=...@a10&6...@a10

app_voicemail.c:1630 messagecount: Couldn't find mailbox 612 in context
a10&6...@a10

It looks like it's interpreting everything after the @ as context.  I'm
running 1.6.1.6.  Thanks anyway - John
-- 
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com

http://www.spiritualoutreach.com
Making Christianity intelligible to secular society


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Re: [asterisk-users] Asterisk with a Cisco AS5300 gateway

[Jonathan Thurman]

>>>  destination-pattern .T
>
>> What does "destination-pattern .T" mean? I'm not familiar with what
>> ".T" would match. I would suggest using a more specific pattern that
>> you expect to be coming down the line.

One or more characters (up to 31 characters), waiting "timeouts
inter-digit" before sending.

http://www.cisco.com/en/US/docs/ios/12_3/vvf_c/dial_peer/dp_plan.html

You could be more specific, if you know what is always going to be
coming down the line, like 503... if you only have Oregon numbers,
and get 10 digits from the provider.  T is useful for outbound calls
with a trunk number such as 9T because you never know what number
those crazy users will try to call.

-Jonathan

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Re: [asterisk-users] inquire if SIP connections are active or not

[Darrin Henshaw]

You could validate whether it has a physical connection I believe. Add
qualify=yes in the sip definition and use something like:

/usr/sbin/asterisk -rx "sip show peer " | grep "UNREACHABLE" | wc -l

Where  is the name of the sip definition on your system. If the
return is 0 then all is well, if the return is 1 then you have a
connection issue. Not sure how to do any other type of validation, but
no doubt it's possible.

On Fri, Oct 16, 2009 at 11:40 AM, Jerry Geis  wrote:
> Is there a way to ask asterisk from a shell script if its connection (SIP)
> is valid to another system. Lets say for example to cisco call manager?
>
> Thanks,
>
> Jerry
>
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Re: [asterisk-users] Check if a variable is set

[Ishfaq Malik]

We have a winner!

Thanks Danny.

My excuse of not thinking of that myself is working wholly in realtime 
and mysql where you have to use priorities apart from when I'm writing 
Macros. Did I get away with that?

Ish

Danny Nicholas wrote:
> Why not this?
> [macro-extcall]
> ;Macro created by Ish to handle external national calls
> exten => s,1,Set(CALLERID(all)=${ARG2})
> exten => s,n,Gotoif($["${ARG3}" != "1"]?dialit)
> exten => s,n,ExecIf(${ARG3}=1|Monitor|My monitor arguments)
> exten => s,n(dialit),Dial(SIP/44${ar...@carrier,45)
> exten => s,n,Hangup
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik
> Sent: Friday, October 16, 2009 9:08 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Check if a variable is set
>
> Hi
>
> Here it is
>
> [macro-extcall]
> ;Macro created by Ish to handle external national calls
> exten => s,1,Set(CALLERID(all)=${ARG2})
> exten => s,2,ExecIf(${ARG3}=1|Monitor|My monitor arguments)
> exten => s,3,Dial(SIP/44${ar...@carrier,45)
> exten => s,4,Hangup
>
> Execution id Macro(extcall|Dialled Number|Caller CLI)
>
> Now to enable monitoring for an outgoing line it would be
>
> Macro(extcall|Dialled Number|Caller CLI|1)
>
> But I have 79 of these in the system already with just the 2 arguments and
> we use realtime so they are all in a DB rather than text file so I'd rather
> not go through all the existing ones and change them to
>
> Macro(extcall|Dialled Number|Caller CLI|0)
>
> If I don't have to.
>
> Also, I don't like getting bested by machines!
>
> Ish
>
>
>
>
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>   

-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062

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[asterisk-users] inquire if SIP connections are active or not

[Jerry Geis]

Is there a way to ask asterisk from a shell script if its connection (SIP)
is valid to another system. Lets say for example to cisco call manager?

Thanks,

Jerry

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Re: [asterisk-users] Queues with unavailable members

[Benny Amorsen]

Benny Amorsen  writes:

> Would it perhaps work to simply Wait(30) if the call is rejected by the
> phone? If the Queue assumes that the phone is busy for those 30 seconds,
> I have accomplished my goal. It's worth a shot.

This works! Actually I tried out Wait(1000), but that worked fine. After
30 seconds (the timeout in the queue) the Local channel was closed, and
a short while later a new call attempt was made. Just as I was hoping.

It would still be neat to have a min_dial_interval option, so that Queue
never overwhelms the server with failing dial attempts.


/Benny


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Re: [asterisk-users] OT wanted old Sipura firmware 2.0.13

[Dave Fullerton]

Joseph wrote:
> Does anybody know here I can find old Sipura firmware 2.0.13 for SPA-3000
> I have Cisco 3.1.20 but it is not working as it suppose to.
> 


http://www.totek.ca/index.php?option=com_content&task=view&id=151&Itemid=39

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Re: [asterisk-users] Check if a variable is set

[Danny Nicholas]

Why not this?
[macro-extcall]
;Macro created by Ish to handle external national calls
exten => s,1,Set(CALLERID(all)=${ARG2})
exten => s,n,Gotoif($["${ARG3}" != "1"]?dialit)
exten => s,n,ExecIf(${ARG3}=1|Monitor|My monitor arguments)
exten => s,n(dialit),Dial(SIP/44${ar...@carrier,45)
exten => s,n,Hangup


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik
Sent: Friday, October 16, 2009 9:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Check if a variable is set

Hi

Here it is

[macro-extcall]
;Macro created by Ish to handle external national calls
exten => s,1,Set(CALLERID(all)=${ARG2})
exten => s,2,ExecIf(${ARG3}=1|Monitor|My monitor arguments)
exten => s,3,Dial(SIP/44${ar...@carrier,45)
exten => s,4,Hangup

Execution id Macro(extcall|Dialled Number|Caller CLI)

Now to enable monitoring for an outgoing line it would be

Macro(extcall|Dialled Number|Caller CLI|1)

But I have 79 of these in the system already with just the 2 arguments and
we use realtime so they are all in a DB rather than text file so I'd rather
not go through all the existing ones and change them to

Macro(extcall|Dialled Number|Caller CLI|0)

If I don't have to.

Also, I don't like getting bested by machines!

Ish




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Re: [asterisk-users] Check if a variable is set

[Ishfaq Malik]

Hi

Here it is

[macro-extcall]
;Macro created by Ish to handle external national calls
exten => s,1,Set(CALLERID(all)=${ARG2})
exten => s,2,ExecIf(${ARG3}=1|Monitor|My monitor arguments)
exten => s,3,Dial(SIP/44${ar...@carrier,45)
exten => s,4,Hangup

Execution id Macro(extcall|Dialled Number|Caller CLI)

Now to enable monitoring for an outgoing line it would be

Macro(extcall|Dialled Number|Caller CLI|1)

But I have 79 of these in the system already with just the 2 arguments and we 
use realtime so they are all in a DB rather than text file so I'd rather not go 
through all the existing ones and change them to

Macro(extcall|Dialled Number|Caller CLI|0)

If I don't have to.

Also, I don't like getting bested by machines!

Ish


Darrin Henshaw wrote:
> Mind posting the macro itself? I think we might need to store the
> return value of isnull then test with execif.
>
> On 16/10/2009, Ishfaq Malik  wrote:
>   
>> That fails to execute in both conditions
>>
>> ABBAS SHAKEEL wrote:
>> 
>>> Please try this
>>>
>>> xten => s,2,ExecIf( 0EXISTS(${ARG3})=1 & 0${ARG3}=1|
>>>
>>> On Fri, Oct 16, 2009 at 3:45 PM, Ishfaq Malik >> > wrote:
>>>
>>> I'm basically trying to make an argument optional in a macro, I'm
>>> starting to think it's not possible
>>>
>>> If I do this in my macro
>>> exten => s,2,ExecIf(EXISTS(${ARG3})=1 & ${ARG3}=1|>> to do>
>>>
>>> I see this in the console
>>> Executing [...@macro-extcall:2] ExecIf("SIP/PACK501-08222428",
>>> "EXISTS()=1
>>> & =1|
>>>
>>> As I didn't pass a third argument.
>>>
>>> Essentially, what I'm trying to do in php terms would be this
>>> if(isset($var) && $var==1)
>>>
>>> Ish
>>>
>>> ABBAS SHAKEEL wrote:
>>> > Sorry its macro I called it a function.
>>> >
>>> > This link will be helpful to you
>>> > http://www.voip-info.org/wiki/index.php?page=Asterisk+variables
>>> >
>>> >
>>> > On Fri, Oct 16, 2009 at 3:13 PM, ABBAS SHAKEEL
>>> > >> 
>>> >> >> wrote:
>>> >
>>> > If you want to check in Console then NOOP can be used .
>>> > if in case of function call you can check its length if there
>>> > exists some thing
>>> >
>>> >
>>> > On Fri, Oct 16, 2009 at 3:04 PM, Ishfaq Malik
>>> mailto:i...@pack-net.co.uk>
>>> > >> wrote:
>>> >
>>> > Hi
>>> >
>>> > Is there any way to check if a variable is set in asterisk?
>>> > I've had a
>>> > look around and can't find a purpose built function for it.
>>> >
>>> > I'm going to be using it to see if an argument has been
>>> passed
>>> > with a
>>> > macro or not (e.g. see if ${ARG3} is set or not)
>>> >
>>> > Thanks in advance
>>> >
>>> > Ish
>>> > --
>>> > Ishfaq Malik
>>> > Software Developer
>>> > PackNet Ltd
>>> >
>>> > Office:   0161 660 3062
>>> >
>>> > ___
>>> > -- Bandwidth and Colocation Provided by
>>> > http://www.api-digital.com --
>>> >
>>> > AstriCon 2009 - October 13 - 15 Phoenix, Arizona
>>> > Register Now: http://www.astricon.net
>>> >
>>> > asterisk-users mailing list
>>> > To UNSUBSCRIBE or update options visit:
>>> >   http://lists.digium.com/mailman/listinfo/asterisk-users
>>> >
>>> >
>>> >
>>> >
>>> > --
>>> > Best Regards
>>> > Shakeel Abbas
>>> >
>>> >
>>> >
>>> >
>>> > --
>>> > Best Regards
>>> > Shakeel Abbas
>>> >
>>> >
>>>
>>> 
>>> >
>>> > ___
>>> > -- Bandwidth and Colocation Provided by
>>> http://www.api-digital.com --
>>> >
>>> > AstriCon 2009 - October 13 - 15 Phoenix, Arizona
>>> > Register Now: http://www.astricon.net
>>> >
>>> > asterisk-users mailing list
>>> > To UNSUBSCRIBE or update options visit:
>>> >http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>> --
>>> Ishfaq Malik
>>> Software Developer
>>> PackNet Ltd
>>>
>>> Office:   0161 660 3062
>>>
>>> ___
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>
>>> AstriCon 2009 - October 13 - 15 Phoenix, Arizona
>>> Register Now: http://www.astricon.net
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>>
>>>

Re: [asterisk-users] Check if a variable is set

[Ishfaq Malik]

Grrr, none of it works, and the ExecIf's default position in the case of 
confusion is to execute rather than to not

Darrin Henshaw wrote:
> Actually just noticed a typo try:
>
> exten => s,1,ExecIf($[${ISNULL(${ARG3})} = 1]|Set,ARG3=1)
>
> Had { instead of [ in the ExecIf.
>
> On Fri, Oct 16, 2009 at 10:26 AM, Darrin Henshaw
>  wrote:
>   
>> Something like:
>>
>> exten => s,1,ExecIf(${${ISNULL(${ARG3})} = 1]|Set,ARG3=1)
>>
>> Should work from what I read on voip-info.org.
>>
>> On Fri, Oct 16, 2009 at 10:19 AM, Darrin Henshaw
>>  wrote:
>> 
>>> Mind posting the macro itself? I think we might need to store the
>>> return value of isnull then test with execif.
>>>
>>> On 16/10/2009, Ishfaq Malik  wrote:
>>>   
 That fails to execute in both conditions

 ABBAS SHAKEEL wrote:
 
> Please try this
>
> xten => s,2,ExecIf( 0EXISTS(${ARG3})=1 & 0${ARG3}=1|
>
> On Fri, Oct 16, 2009 at 3:45 PM, Ishfaq Malik  > wrote:
>
> I'm basically trying to make an argument optional in a macro, I'm
> starting to think it's not possible
>
> If I do this in my macro
> exten => s,2,ExecIf(EXISTS(${ARG3})=1 & ${ARG3}=1| to do>
>
> I see this in the console
> Executing [...@macro-extcall:2] ExecIf("SIP/PACK501-08222428",
> "EXISTS()=1
> & =1|
>
> As I didn't pass a third argument.
>
> Essentially, what I'm trying to do in php terms would be this
> if(isset($var) && $var==1)
>
> Ish
>
> ABBAS SHAKEEL wrote:
> > Sorry its macro I called it a function.
> >
> > This link will be helpful to you
> > http://www.voip-info.org/wiki/index.php?page=Asterisk+variables
> >
> >
> > On Fri, Oct 16, 2009 at 3:13 PM, ABBAS SHAKEEL
> >  
>  >> wrote:
> >
> > If you want to check in Console then NOOP can be used .
> > if in case of function call you can check its length if there
> > exists some thing
> >
> >
> > On Fri, Oct 16, 2009 at 3:04 PM, Ishfaq Malik
> mailto:i...@pack-net.co.uk>
> > >> 
> wrote:
> >
> > Hi
> >
> > Is there any way to check if a variable is set in asterisk?
> > I've had a
> > look around and can't find a purpose built function for it.
> >
> > I'm going to be using it to see if an argument has been
> passed
> > with a
> > macro or not (e.g. see if ${ARG3} is set or not)
> >
> > Thanks in advance
> >
> > Ish
> > --
> > Ishfaq Malik
> > Software Developer
> > PackNet Ltd
> >
> > Office:   0161 660 3062
> >
> > ___
> > -- Bandwidth and Colocation Provided by
> > http://www.api-digital.com --
> >
> > AstriCon 2009 - October 13 - 15 Phoenix, Arizona
> > Register Now: http://www.astricon.net
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
> >
> > --
> > Best Regards
> > Shakeel Abbas
> >
> >
> >
> >
> > --
> > Best Regards
> > Shakeel Abbas
> >
> >
>
> 
> >
> > ___
> > -- Bandwidth and Colocation Provided by
> http://www.api-digital.com --
> >
> > AstriCon 2009 - October 13 - 15 Phoenix, Arizona
> > Register Now: http://www.astricon.net
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> Ishfaq Malik
> Software Developer
> PackNet Ltd
>
> Office:   0161 660 3062
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2009 - October 13 - 15 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:

Re: [asterisk-users] app_swift issue

[Jeremy Kister]

On Tue, 25 Aug 2009 23:37:12 -0700, Abbas Shakeel wrote:
 > when i try to execute make command on app_swift-1.6.2
 >
 > I get the following error
 >
 > [r...@asterisk app_swift-1.6.2]# make
 > gcc -I/opt/swift/include -g -Wall -D_REENTRANT -D_GNU_SOURCE -fPIC   -c -o
 > app_swift.o app_swift.c
 > app_swift.c: In function ‘engine’:
 > app_swift.c:402: error: incompatible types in assignment
 > app_swift.c: In function ‘load_module’:
 > app_swift.c:546: error: ‘AST_MODULE’ undeclared (first use in this function)

try the patch at http://jeremy.kister.net/code/app_swift-1.6.2.patch



-- 

Jeremy Kister
http://jeremy.kister.net./

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Re: [asterisk-users] question on SIP and call manager

[Danny Nicholas]

On Asterisk 1.4, Call doesn't line Channel: A&B.  
You could put the second dialplan snippet into a context and do your
callfile like this:
[callccm]
exten => s,1,Dial(SIP/CCMMAIN,10,KkTt)
Exten => s,n,Dial(SIP/CCMSLAVE,10,KkTt)

--
Channel: SIP/104
CallerID: SIP/104
MaxRetries: 1
WaitTime: 60
retryTime: 5
Context: callccm
Extension: s

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Thursday, October 15, 2009 7:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] question on SIP and call manager

>
> Here are two ways to address this
>
> 1. Dial(SIP/CCMMAIN&SIP/CCMSLAVE) - this tries both at once
>
> 2. exten => s,1,Dial(SIP/CCMMAIN,10,KkTt)
>Exten => s,n,Dial(SIP/CCMSLAVE,10,KkTt)
>
> CCMSLAVE only gets called if no one answers CCMMAIN in 10 seconds (2-3
> rings)
>
>   
Danny thats good to know for extensions.conf
but
I am using call files.

echo "Channel: SIP/CCMMAIN/5551212" >  /tmp/call
echo "Context: smvoice-test" >> /tmp/call

Can I do the Channel: SIP/CCMMAIN/5551212&SIP/CCMSLAVE/5551212
in the Channel for the call file?


Jerry


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Re: [asterisk-users] Soft phone not registering

[Darrin Henshaw]

First suggestion is if this Asterisk server is accessible from the
internet put a secret in the peer definition. What you have now is
wide open. Second thing is if I understand it you are going:

PC(Soft Phone) > ADSL Router > Internet > Asterisk box. Is that
correct? If not, can you descibe it better.

On Fri, Oct 16, 2009 at 7:56 AM, Rakesh Sabharwal
 wrote:
>
> HI All,
>
> I have installed Asterisk 1.4.26.2 on a centOS box on a public IP and trying 
> to connect from softphone behind ADSL router.
>
> The softphone is not able to register, we get some SIP messages on the 
> server, which look like below.
>
> Please advise where could be the issue.
>
> Thnx
> Rakesh
>
> ---
> Retransmitting #3 (NAT) to x.x.x.x:38155:
> OPTIONS sip:te...@192.168.1.7:5060;rinstance=5b19b87f10954011;transport=UDP 
> SIP/2.0
> Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK39432aec;rport
> From: "asterisk" ;tag=as7d8aae9d
> To: 
> Contact: 
> Call-ID: 3c92389c5e72d3e92fd8d20b70055...@x.x.x.x
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Fri, 16 Oct 2009 10:47:56 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
> Supported: replaces
> Content-Length: 0
>
>
> ---
> Retransmitting #4 (NAT) to x.x.x.x:38155:
> OPTIONS sip:te...@192.168.1.7:5060;rinstance=5b19b87f10954011;transport=UDP 
> SIP/2.0
> Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK39432aec;rport
> From: "asterisk" ;tag=as7d8aae9d
> To: 
> Contact: 
> Call-ID: 3c92389c5e72d3e92fd8d20b70055...@x.x.x.x
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Fri, 16 Oct 2009 10:47:56 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
> Supported: replaces
> Content-Length: 0
>
> 
>
> sip.conf 
>
> [general]
> context = tutorial
> bindport = 5060
> bindaddr =0.0.0.0
> domain = x.x.x.x
> nat=yes
> disallow = all
> allow = alaw
> keeprtpalive = yes
> notifyringing = yes
> canreinvite = no
> type = peer
> realm = asterisk
> qualify = yes
>
> [test2]
> type = peer
> host = dynamic
> username = test2
> context = tutorial
> port = 5060
> notifyringing = yes
> nat = yes
> type = friend
> canreinvite = no
> realm = asterisk
> qualify = yes
> mailbox=...@mb_tutorial
>
> ---
>
>
>
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2009 - October 13 - 15 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>

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Re: [asterisk-users] Check if a variable is set

[Darrin Henshaw]

Actually just noticed a typo try:

exten => s,1,ExecIf($[${ISNULL(${ARG3})} = 1]|Set,ARG3=1)

Had { instead of [ in the ExecIf.

On Fri, Oct 16, 2009 at 10:26 AM, Darrin Henshaw
 wrote:
> Something like:
>
> exten => s,1,ExecIf(${${ISNULL(${ARG3})} = 1]|Set,ARG3=1)
>
> Should work from what I read on voip-info.org.
>
> On Fri, Oct 16, 2009 at 10:19 AM, Darrin Henshaw
>  wrote:
>> Mind posting the macro itself? I think we might need to store the
>> return value of isnull then test with execif.
>>
>> On 16/10/2009, Ishfaq Malik  wrote:
>>> That fails to execute in both conditions
>>>
>>> ABBAS SHAKEEL wrote:
 Please try this

 xten => s,2,ExecIf( 0EXISTS(${ARG3})=1 & 0${ARG3}=1|

 On Fri, Oct 16, 2009 at 3:45 PM, Ishfaq Malik >>> > wrote:

     I'm basically trying to make an argument optional in a macro, I'm
     starting to think it's not possible

     If I do this in my macro
     exten => s,2,ExecIf(EXISTS(${ARG3})=1 & ${ARG3}=1|>>>     to do>

     I see this in the console
     Executing [...@macro-extcall:2] ExecIf("SIP/PACK501-08222428",
     "EXISTS()=1
     & =1|

     As I didn't pass a third argument.

     Essentially, what I'm trying to do in php terms would be this
     if(isset($var) && $var==1)

     Ish

     ABBAS SHAKEEL wrote:
     > Sorry its macro I called it a function.
     >
     > This link will be helpful to you
     > http://www.voip-info.org/wiki/index.php?page=Asterisk+variables
     >
     >
     > On Fri, Oct 16, 2009 at 3:13 PM, ABBAS SHAKEEL
     > >>>     
     >> wrote:
     >
     >     If you want to check in Console then NOOP can be used .
     >     if in case of function call you can check its length if there
     >     exists some thing
     >
     >
     >     On Fri, Oct 16, 2009 at 3:04 PM, Ishfaq Malik
     mailto:i...@pack-net.co.uk>
     >     >> wrote:
     >
     >         Hi
     >
     >         Is there any way to check if a variable is set in asterisk?
     >         I've had a
     >         look around and can't find a purpose built function for it.
     >
     >         I'm going to be using it to see if an argument has been
     passed
     >         with a
     >         macro or not (e.g. see if ${ARG3} is set or not)
     >
     >         Thanks in advance
     >
     >         Ish
     >         --
     >         Ishfaq Malik
     >         Software Developer
     >         PackNet Ltd
     >
     >         Office:   0161 660 3062
     >
     >         ___
     >         -- Bandwidth and Colocation Provided by
     >         http://www.api-digital.com --
     >
     >         AstriCon 2009 - October 13 - 15 Phoenix, Arizona
     >         Register Now: http://www.astricon.net
     >
     >         asterisk-users mailing list
     >         To UNSUBSCRIBE or update options visit:
     >           http://lists.digium.com/mailman/listinfo/asterisk-users
     >
     >
     >
     >
     >     --
     >     Best Regards
     >     Shakeel Abbas
     >
     >
     >
     >
     > --
     > Best Regards
     > Shakeel Abbas
     >
     >

 
     >
     > ___
     > -- Bandwidth and Colocation Provided by
     http://www.api-digital.com --
     >
     > AstriCon 2009 - October 13 - 15 Phoenix, Arizona
     > Register Now: http://www.astricon.net
     >
     > asterisk-users mailing list
     > To UNSUBSCRIBE or update options visit:
     >    http://lists.digium.com/mailman/listinfo/asterisk-users

     --
     Ishfaq Malik
     Software Developer
     PackNet Ltd

     Office:   0161 660 3062

     ___
     -- Bandwidth and Colocation Provided by http://www.api-digital.com --

     AstriCon 2009 - October 13 - 15 Phoenix, Arizona
     Register Now: http://www.astricon.net

     asterisk-users mailing list
     To UNSUBSCRIBE or update options visit:
       http://lists.digium.com/mailman/listinfo/asterisk-users




 --
 Best Regards
 Shakeel Abbas

 

 ___
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Re: [asterisk-users] Check if a variable is set

[Darrin Henshaw]

Something like:

exten => s,1,ExecIf(${${ISNULL(${ARG3})} = 1]|Set,ARG3=1)

Should work from what I read on voip-info.org.

On Fri, Oct 16, 2009 at 10:19 AM, Darrin Henshaw
 wrote:
> Mind posting the macro itself? I think we might need to store the
> return value of isnull then test with execif.
>
> On 16/10/2009, Ishfaq Malik  wrote:
>> That fails to execute in both conditions
>>
>> ABBAS SHAKEEL wrote:
>>> Please try this
>>>
>>> xten => s,2,ExecIf( 0EXISTS(${ARG3})=1 & 0${ARG3}=1|
>>>
>>> On Fri, Oct 16, 2009 at 3:45 PM, Ishfaq Malik >> > wrote:
>>>
>>>     I'm basically trying to make an argument optional in a macro, I'm
>>>     starting to think it's not possible
>>>
>>>     If I do this in my macro
>>>     exten => s,2,ExecIf(EXISTS(${ARG3})=1 & ${ARG3}=1|>>     to do>
>>>
>>>     I see this in the console
>>>     Executing [...@macro-extcall:2] ExecIf("SIP/PACK501-08222428",
>>>     "EXISTS()=1
>>>     & =1|
>>>
>>>     As I didn't pass a third argument.
>>>
>>>     Essentially, what I'm trying to do in php terms would be this
>>>     if(isset($var) && $var==1)
>>>
>>>     Ish
>>>
>>>     ABBAS SHAKEEL wrote:
>>>     > Sorry its macro I called it a function.
>>>     >
>>>     > This link will be helpful to you
>>>     > http://www.voip-info.org/wiki/index.php?page=Asterisk+variables
>>>     >
>>>     >
>>>     > On Fri, Oct 16, 2009 at 3:13 PM, ABBAS SHAKEEL
>>>     > >>     
>>>     >>     >> wrote:
>>>     >
>>>     >     If you want to check in Console then NOOP can be used .
>>>     >     if in case of function call you can check its length if there
>>>     >     exists some thing
>>>     >
>>>     >
>>>     >     On Fri, Oct 16, 2009 at 3:04 PM, Ishfaq Malik
>>>     mailto:i...@pack-net.co.uk>
>>>     >     >> wrote:
>>>     >
>>>     >         Hi
>>>     >
>>>     >         Is there any way to check if a variable is set in asterisk?
>>>     >         I've had a
>>>     >         look around and can't find a purpose built function for it.
>>>     >
>>>     >         I'm going to be using it to see if an argument has been
>>>     passed
>>>     >         with a
>>>     >         macro or not (e.g. see if ${ARG3} is set or not)
>>>     >
>>>     >         Thanks in advance
>>>     >
>>>     >         Ish
>>>     >         --
>>>     >         Ishfaq Malik
>>>     >         Software Developer
>>>     >         PackNet Ltd
>>>     >
>>>     >         Office:   0161 660 3062
>>>     >
>>>     >         ___
>>>     >         -- Bandwidth and Colocation Provided by
>>>     >         http://www.api-digital.com --
>>>     >
>>>     >         AstriCon 2009 - October 13 - 15 Phoenix, Arizona
>>>     >         Register Now: http://www.astricon.net
>>>     >
>>>     >         asterisk-users mailing list
>>>     >         To UNSUBSCRIBE or update options visit:
>>>     >           http://lists.digium.com/mailman/listinfo/asterisk-users
>>>     >
>>>     >
>>>     >
>>>     >
>>>     >     --
>>>     >     Best Regards
>>>     >     Shakeel Abbas
>>>     >
>>>     >
>>>     >
>>>     >
>>>     > --
>>>     > Best Regards
>>>     > Shakeel Abbas
>>>     >
>>>     >
>>>
>>> 
>>>     >
>>>     > ___
>>>     > -- Bandwidth and Colocation Provided by
>>>     http://www.api-digital.com --
>>>     >
>>>     > AstriCon 2009 - October 13 - 15 Phoenix, Arizona
>>>     > Register Now: http://www.astricon.net
>>>     >
>>>     > asterisk-users mailing list
>>>     > To UNSUBSCRIBE or update options visit:
>>>     >    http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>>     --
>>>     Ishfaq Malik
>>>     Software Developer
>>>     PackNet Ltd
>>>
>>>     Office:   0161 660 3062
>>>
>>>     ___
>>>     -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>
>>>     AstriCon 2009 - October 13 - 15 Phoenix, Arizona
>>>     Register Now: http://www.astricon.net
>>>
>>>     asterisk-users mailing list
>>>     To UNSUBSCRIBE or update options visit:
>>>       http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>>
>>>
>>>
>>> --
>>> Best Regards
>>> Shakeel Abbas
>>>
>>> 
>>>
>>> ___
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>
>>> AstriCon 2009 - October 13 - 15 Phoenix, Arizona
>>> Register Now: http://www.astricon.net
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>    http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>> --
>> Ishfaq Malik
>> Software Developer
>> PackNet Ltd
>>
>> Office:   0161 660 3062
>>

Re: [asterisk-users] Check if a variable is set

[Danny Nicholas]

This might work:
exten => s,2,ExecIf($["LEN(${ARG3})" >"0"] & ${ARG3}=1|

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik
Sent: Friday, October 16, 2009 7:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Check if a variable is set

That fails to execute in both conditions

ABBAS SHAKEEL wrote:
> Please try this 
>
> xten => s,2,ExecIf( 0EXISTS(${ARG3})=1 & 0${ARG3}=1|
>
> On Fri, Oct 16, 2009 at 3:45 PM, Ishfaq Malik  > wrote:
>
> I'm basically trying to make an argument optional in a macro, I'm
> starting to think it's not possible
>
> If I do this in my macro
> exten => s,2,ExecIf(EXISTS(${ARG3})=1 & ${ARG3}=1| to do>
>
> I see this in the console
> Executing [...@macro-extcall:2] ExecIf("SIP/PACK501-08222428",
> "EXISTS()=1
> & =1|
>
> As I didn't pass a third argument.
>
> Essentially, what I'm trying to do in php terms would be this
> if(isset($var) && $var==1)
>
> Ish
>
> ABBAS SHAKEEL wrote:
> > Sorry its macro I called it a function.
> >
> > This link will be helpful to you
> > http://www.voip-info.org/wiki/index.php?page=Asterisk+variables
> >
> >
> > On Fri, Oct 16, 2009 at 3:13 PM, ABBAS SHAKEEL
> >  
>  >> wrote:
> >
> > If you want to check in Console then NOOP can be used .
> > if in case of function call you can check its length if there
> > exists some thing
> >
> >
> > On Fri, Oct 16, 2009 at 3:04 PM, Ishfaq Malik
> mailto:i...@pack-net.co.uk>
> > >> wrote:
> >
> > Hi
> >
> > Is there any way to check if a variable is set in asterisk?
> > I've had a
> > look around and can't find a purpose built function for it.
> >
> > I'm going to be using it to see if an argument has been
> passed
> > with a
> > macro or not (e.g. see if ${ARG3} is set or not)
> >
> > Thanks in advance
> >
> > Ish
> > --
> > Ishfaq Malik
> > Software Developer
> > PackNet Ltd
> >
> > Office:   0161 660 3062
> >
> > ___
> > -- Bandwidth and Colocation Provided by
> > http://www.api-digital.com --
> >
> > AstriCon 2009 - October 13 - 15 Phoenix, Arizona
> > Register Now: http://www.astricon.net
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
> >
> > --
> > Best Regards
> > Shakeel Abbas
> >
> >
> >
> >
> > --
> > Best Regards
> > Shakeel Abbas
> >
> >
>

> >
> > ___
> > -- Bandwidth and Colocation Provided by
> http://www.api-digital.com --
> >
> > AstriCon 2009 - October 13 - 15 Phoenix, Arizona
> > Register Now: http://www.astricon.net
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> Ishfaq Malik
> Software Developer
> PackNet Ltd
>
> Office:   0161 660 3062
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2009 - October 13 - 15 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>
> -- 
> Best Regards
> Shakeel Abbas
>
> 
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2009 - October 13 - 15 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062

___
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AstriCon 2009 - October 13 - 15 Phoenix, Arizona
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Re: [asterisk-users] Check if a variable is set

[Darrin Henshaw]

Mind posting the macro itself? I think we might need to store the
return value of isnull then test with execif.

On 16/10/2009, Ishfaq Malik  wrote:
> That fails to execute in both conditions
>
> ABBAS SHAKEEL wrote:
>> Please try this
>>
>> xten => s,2,ExecIf( 0EXISTS(${ARG3})=1 & 0${ARG3}=1|
>>
>> On Fri, Oct 16, 2009 at 3:45 PM, Ishfaq Malik > > wrote:
>>
>> I'm basically trying to make an argument optional in a macro, I'm
>> starting to think it's not possible
>>
>> If I do this in my macro
>> exten => s,2,ExecIf(EXISTS(${ARG3})=1 & ${ARG3}=1|> to do>
>>
>> I see this in the console
>> Executing [...@macro-extcall:2] ExecIf("SIP/PACK501-08222428",
>> "EXISTS()=1
>> & =1|
>>
>> As I didn't pass a third argument.
>>
>> Essentially, what I'm trying to do in php terms would be this
>> if(isset($var) && $var==1)
>>
>> Ish
>>
>> ABBAS SHAKEEL wrote:
>> > Sorry its macro I called it a function.
>> >
>> > This link will be helpful to you
>> > http://www.voip-info.org/wiki/index.php?page=Asterisk+variables
>> >
>> >
>> > On Fri, Oct 16, 2009 at 3:13 PM, ABBAS SHAKEEL
>> > > 
>> > >> wrote:
>> >
>> > If you want to check in Console then NOOP can be used .
>> > if in case of function call you can check its length if there
>> > exists some thing
>> >
>> >
>> > On Fri, Oct 16, 2009 at 3:04 PM, Ishfaq Malik
>> mailto:i...@pack-net.co.uk>
>> > >> wrote:
>> >
>> > Hi
>> >
>> > Is there any way to check if a variable is set in asterisk?
>> > I've had a
>> > look around and can't find a purpose built function for it.
>> >
>> > I'm going to be using it to see if an argument has been
>> passed
>> > with a
>> > macro or not (e.g. see if ${ARG3} is set or not)
>> >
>> > Thanks in advance
>> >
>> > Ish
>> > --
>> > Ishfaq Malik
>> > Software Developer
>> > PackNet Ltd
>> >
>> > Office:   0161 660 3062
>> >
>> > ___
>> > -- Bandwidth and Colocation Provided by
>> > http://www.api-digital.com --
>> >
>> > AstriCon 2009 - October 13 - 15 Phoenix, Arizona
>> > Register Now: http://www.astricon.net
>> >
>> > asterisk-users mailing list
>> > To UNSUBSCRIBE or update options visit:
>> >   http://lists.digium.com/mailman/listinfo/asterisk-users
>> >
>> >
>> >
>> >
>> > --
>> > Best Regards
>> > Shakeel Abbas
>> >
>> >
>> >
>> >
>> > --
>> > Best Regards
>> > Shakeel Abbas
>> >
>> >
>>
>> 
>> >
>> > ___
>> > -- Bandwidth and Colocation Provided by
>> http://www.api-digital.com --
>> >
>> > AstriCon 2009 - October 13 - 15 Phoenix, Arizona
>> > Register Now: http://www.astricon.net
>> >
>> > asterisk-users mailing list
>> > To UNSUBSCRIBE or update options visit:
>> >http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>> --
>> Ishfaq Malik
>> Software Developer
>> PackNet Ltd
>>
>> Office:   0161 660 3062
>>
>> ___
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> AstriCon 2009 - October 13 - 15 Phoenix, Arizona
>> Register Now: http://www.astricon.net
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>>
>>
>> --
>> Best Regards
>> Shakeel Abbas
>>
>> 
>>
>> ___
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> AstriCon 2009 - October 13 - 15 Phoenix, Arizona
>> Register Now: http://www.astricon.net
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> Ishfaq Malik
> Software Developer
> PackNet Ltd
>
> Office:   0161 660 3062
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2009 - October 13 - 15 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asteris

Re: [asterisk-users] Check if a variable is set

[Ishfaq Malik]

That fails to execute in both conditions

ABBAS SHAKEEL wrote:
> Please try this 
>
> xten => s,2,ExecIf( 0EXISTS(${ARG3})=1 & 0${ARG3}=1|
>
> On Fri, Oct 16, 2009 at 3:45 PM, Ishfaq Malik  > wrote:
>
> I'm basically trying to make an argument optional in a macro, I'm
> starting to think it's not possible
>
> If I do this in my macro
> exten => s,2,ExecIf(EXISTS(${ARG3})=1 & ${ARG3}=1| to do>
>
> I see this in the console
> Executing [...@macro-extcall:2] ExecIf("SIP/PACK501-08222428",
> "EXISTS()=1
> & =1|
>
> As I didn't pass a third argument.
>
> Essentially, what I'm trying to do in php terms would be this
> if(isset($var) && $var==1)
>
> Ish
>
> ABBAS SHAKEEL wrote:
> > Sorry its macro I called it a function.
> >
> > This link will be helpful to you
> > http://www.voip-info.org/wiki/index.php?page=Asterisk+variables
> >
> >
> > On Fri, Oct 16, 2009 at 3:13 PM, ABBAS SHAKEEL
> >  
>  >> wrote:
> >
> > If you want to check in Console then NOOP can be used .
> > if in case of function call you can check its length if there
> > exists some thing
> >
> >
> > On Fri, Oct 16, 2009 at 3:04 PM, Ishfaq Malik
> mailto:i...@pack-net.co.uk>
> > >> wrote:
> >
> > Hi
> >
> > Is there any way to check if a variable is set in asterisk?
> > I've had a
> > look around and can't find a purpose built function for it.
> >
> > I'm going to be using it to see if an argument has been
> passed
> > with a
> > macro or not (e.g. see if ${ARG3} is set or not)
> >
> > Thanks in advance
> >
> > Ish
> > --
> > Ishfaq Malik
> > Software Developer
> > PackNet Ltd
> >
> > Office:   0161 660 3062
> >
> > ___
> > -- Bandwidth and Colocation Provided by
> > http://www.api-digital.com --
> >
> > AstriCon 2009 - October 13 - 15 Phoenix, Arizona
> > Register Now: http://www.astricon.net
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
> >
> > --
> > Best Regards
> > Shakeel Abbas
> >
> >
> >
> >
> > --
> > Best Regards
> > Shakeel Abbas
> >
> >
> 
> >
> > ___
> > -- Bandwidth and Colocation Provided by
> http://www.api-digital.com --
> >
> > AstriCon 2009 - October 13 - 15 Phoenix, Arizona
> > Register Now: http://www.astricon.net
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> Ishfaq Malik
> Software Developer
> PackNet Ltd
>
> Office:   0161 660 3062
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2009 - October 13 - 15 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>
> -- 
> Best Regards
> Shakeel Abbas
>
> 
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2009 - October 13 - 15 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Check if a variable is set

[Darrin Henshaw]

My first though is using the isnull function.

http://www.voip-info.org/wiki/view/Asterisk+func+isnull

On 16/10/2009, Ishfaq Malik  wrote:
> I'm basically trying to make an argument optional in a macro, I'm
> starting to think it's not possible
>
> If I do this in my macro
> exten => s,2,ExecIf(EXISTS(${ARG3})=1 & ${ARG3}=1|
>
> I see this in the console
> Executing [...@macro-extcall:2] ExecIf("SIP/PACK501-08222428", "EXISTS()=1
> & =1|
>
> As I didn't pass a third argument.
>
> Essentially, what I'm trying to do in php terms would be this
> if(isset($var) && $var==1)
>
> Ish
>
> ABBAS SHAKEEL wrote:
>> Sorry its macro I called it a function.
>>
>> This link will be helpful to you
>> http://www.voip-info.org/wiki/index.php?page=Asterisk+variables
>>
>>
>> On Fri, Oct 16, 2009 at 3:13 PM, ABBAS SHAKEEL
>> mailto:shakeel.abbas@gmail.com>> wrote:
>>
>> If you want to check in Console then NOOP can be used .
>> if in case of function call you can check its length if there
>> exists some thing
>>
>>
>> On Fri, Oct 16, 2009 at 3:04 PM, Ishfaq Malik > > wrote:
>>
>> Hi
>>
>> Is there any way to check if a variable is set in asterisk?
>> I've had a
>> look around and can't find a purpose built function for it.
>>
>> I'm going to be using it to see if an argument has been passed
>> with a
>> macro or not (e.g. see if ${ARG3} is set or not)
>>
>> Thanks in advance
>>
>> Ish
>> --
>> Ishfaq Malik
>> Software Developer
>> PackNet Ltd
>>
>> Office:   0161 660 3062
>>
>> ___
>> -- Bandwidth and Colocation Provided by
>> http://www.api-digital.com --
>>
>> AstriCon 2009 - October 13 - 15 Phoenix, Arizona
>> Register Now: http://www.astricon.net
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>>
>>
>> --
>> Best Regards
>> Shakeel Abbas
>>
>>
>>
>>
>> --
>> Best Regards
>> Shakeel Abbas
>>
>> 
>>
>> ___
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> AstriCon 2009 - October 13 - 15 Phoenix, Arizona
>> Register Now: http://www.astricon.net
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> Ishfaq Malik
> Software Developer
> PackNet Ltd
>
> Office:   0161 660 3062
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2009 - October 13 - 15 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>

-- 
Sent from my mobile device

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Re: [asterisk-users] Check if a variable is set

[ABBAS SHAKEEL]

Please try this
xten => s,2,ExecIf( 0EXISTS(${ARG3})=1 & 0${ARG3}=1|

On Fri, Oct 16, 2009 at 3:45 PM, Ishfaq Malik  wrote:

> I'm basically trying to make an argument optional in a macro, I'm
> starting to think it's not possible
>
> If I do this in my macro
> exten => s,2,ExecIf(EXISTS(${ARG3})=1 & ${ARG3}=1|
>
> I see this in the console
> Executing [...@macro-extcall:2] ExecIf("SIP/PACK501-08222428", "EXISTS()=1
> & =1|
>
> As I didn't pass a third argument.
>
> Essentially, what I'm trying to do in php terms would be this
> if(isset($var) && $var==1)
>
> Ish
>
> ABBAS SHAKEEL wrote:
> > Sorry its macro I called it a function.
> >
> > This link will be helpful to you
> > http://www.voip-info.org/wiki/index.php?page=Asterisk+variables
> >
> >
> > On Fri, Oct 16, 2009 at 3:13 PM, ABBAS SHAKEEL
> > mailto:shakeel.abbas@gmail.com>>
> wrote:
> >
> > If you want to check in Console then NOOP can be used .
> > if in case of function call you can check its length if there
> > exists some thing
> >
> >
> > On Fri, Oct 16, 2009 at 3:04 PM, Ishfaq Malik  > > wrote:
> >
> > Hi
> >
> > Is there any way to check if a variable is set in asterisk?
> > I've had a
> > look around and can't find a purpose built function for it.
> >
> > I'm going to be using it to see if an argument has been passed
> > with a
> > macro or not (e.g. see if ${ARG3} is set or not)
> >
> > Thanks in advance
> >
> > Ish
> > --
> > Ishfaq Malik
> > Software Developer
> > PackNet Ltd
> >
> > Office:   0161 660 3062
> >
> > ___
> > -- Bandwidth and Colocation Provided by
> > http://www.api-digital.com --
> >
> > AstriCon 2009 - October 13 - 15 Phoenix, Arizona
> > Register Now: http://www.astricon.net
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
> >
> > --
> > Best Regards
> > Shakeel Abbas
> >
> >
> >
> >
> > --
> > Best Regards
> > Shakeel Abbas
> >
> > 
> >
> > ___
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
> > AstriCon 2009 - October 13 - 15 Phoenix, Arizona
> > Register Now: http://www.astricon.net
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> Ishfaq Malik
> Software Developer
> PackNet Ltd
>
> Office:   0161 660 3062
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2009 - October 13 - 15 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
Best Regards
Shakeel Abbas
___
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Register Now: http://www.astricon.net

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Re: [asterisk-users] Invite after bye?

[Kevin P. Fleming]

Josip Djuricic wrote:

> so it's like this:
> 
> side A sends bye to asterisk, asterisk responds with 200 OK to side A,
> then it sends INVITE to side B, expects 200 OK from side B, and then
> sends ACK and BYE to side B

This occurs often when directmedia (canreinvite) is in use, and the
media path has been redirected to be between the endpoints; when
endpoint A issues a BYE, Asterisk knows that endpoint B can no longer
send media directly to endpoint A, so it reinvites the media path back
to Asterisk... only to then hangup the call. It's not optimal, but it
shouldn't cause any harm.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out at www.digium.com & www.asterisk.org

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[asterisk-users] Soft phone not registering

[Rakesh Sabharwal]

HI All,

I have installed Asterisk 1.4.26.2 on a centOS box on a public IP and trying to 
connect from softphone behind ADSL router.

The softphone is not able to register, we get some SIP messages on the server, 
which look like below.

Please advise where could be the issue.

Thnx
Rakesh

---
Retransmitting #3 (NAT) to x.x.x.x:38155:
OPTIONS sip:te...@192.168.1.7:5060;rinstance=5b19b87f10954011;transport=UDP 
SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK39432aec;rport
From: "asterisk" ;tag=as7d8aae9d
To: 
Contact: 
Call-ID: 3c92389c5e72d3e92fd8d20b70055...@x.x.x.x
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 16 Oct 2009 10:47:56 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


---
Retransmitting #4 (NAT) to x.x.x.x:38155:
OPTIONS sip:te...@192.168.1.7:5060;rinstance=5b19b87f10954011;transport=UDP 
SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK39432aec;rport
From: "asterisk" ;tag=as7d8aae9d
To: 
Contact: 
Call-ID: 3c92389c5e72d3e92fd8d20b70055...@x.x.x.x
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 16 Oct 2009 10:47:56 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0



sip.conf 

[general]
context = tutorial
bindport = 5060
bindaddr =0.0.0.0
domain = x.x.x.x
nat=yes
disallow = all
allow = alaw
keeprtpalive = yes
notifyringing = yes
canreinvite = no
type = peer
realm = asterisk
qualify = yes

[test2]
type = peer
host = dynamic
username = test2
context = tutorial
port = 5060
notifyringing = yes
nat = yes
type = friend
canreinvite = no
realm = asterisk
qualify = yes
mailbox=...@mb_tutorial

---


  

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Re: [asterisk-users] Check if a variable is set

[Ishfaq Malik]

I'm basically trying to make an argument optional in a macro, I'm 
starting to think it's not possible

If I do this in my macro
exten => s,2,ExecIf(EXISTS(${ARG3})=1 & ${ARG3}=1|

I see this in the console
Executing [...@macro-extcall:2] ExecIf("SIP/PACK501-08222428", "EXISTS()=1 
& =1|

As I didn't pass a third argument.

Essentially, what I'm trying to do in php terms would be this
if(isset($var) && $var==1)

Ish

ABBAS SHAKEEL wrote:
> Sorry its macro I called it a function.
>
> This link will be helpful to you
> http://www.voip-info.org/wiki/index.php?page=Asterisk+variables
>
>
> On Fri, Oct 16, 2009 at 3:13 PM, ABBAS SHAKEEL 
> mailto:shakeel.abbas@gmail.com>> wrote:
>
> If you want to check in Console then NOOP can be used .
> if in case of function call you can check its length if there
> exists some thing
>
>
> On Fri, Oct 16, 2009 at 3:04 PM, Ishfaq Malik  > wrote:
>
> Hi
>
> Is there any way to check if a variable is set in asterisk?
> I've had a
> look around and can't find a purpose built function for it.
>
> I'm going to be using it to see if an argument has been passed
> with a
> macro or not (e.g. see if ${ARG3} is set or not)
>
> Thanks in advance
>
> Ish
> --
> Ishfaq Malik
> Software Developer
> PackNet Ltd
>
> Office:   0161 660 3062
>
> ___
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>
>
>
>
> -- 
> Best Regards
> Shakeel Abbas
>
>
>
>
> -- 
> Best Regards
> Shakeel Abbas
>
> 
>
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-- 
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Software Developer
PackNet Ltd

Office:   0161 660 3062

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Re: [asterisk-users] Check if a variable is set

[ABBAS SHAKEEL]

Sorry its macro I called it a function.
This link will be helpful to you
http://www.voip-info.org/wiki/index.php?page=Asterisk+variables


On Fri, Oct 16, 2009 at 3:13 PM, ABBAS SHAKEEL
wrote:

> If you want to check in Console then NOOP can be used .if in case of
> function call you can check its length if there exists some thing
>
>
> On Fri, Oct 16, 2009 at 3:04 PM, Ishfaq Malik  wrote:
>
>> Hi
>>
>> Is there any way to check if a variable is set in asterisk? I've had a
>> look around and can't find a purpose built function for it.
>>
>> I'm going to be using it to see if an argument has been passed with a
>> macro or not (e.g. see if ${ARG3} is set or not)
>>
>> Thanks in advance
>>
>> Ish
>> --
>> Ishfaq Malik
>> Software Developer
>> PackNet Ltd
>>
>> Office:   0161 660 3062
>>
>> ___
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> AstriCon 2009 - October 13 - 15 Phoenix, Arizona
>> Register Now: http://www.astricon.net
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
> Best Regards
> Shakeel Abbas
>
>


-- 
Best Regards
Shakeel Abbas
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Re: [asterisk-users] Check if a variable is set

[ABBAS SHAKEEL]

If you want to check in Console then NOOP can be used .if in case of
function call you can check its length if there exists some thing

On Fri, Oct 16, 2009 at 3:04 PM, Ishfaq Malik  wrote:

> Hi
>
> Is there any way to check if a variable is set in asterisk? I've had a
> look around and can't find a purpose built function for it.
>
> I'm going to be using it to see if an argument has been passed with a
> macro or not (e.g. see if ${ARG3} is set or not)
>
> Thanks in advance
>
> Ish
> --
> Ishfaq Malik
> Software Developer
> PackNet Ltd
>
> Office:   0161 660 3062
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2009 - October 13 - 15 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
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Shakeel Abbas
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[asterisk-users] Invite after bye?

[Josip Djuricic]

Hi there 

 

noticed a strange thing in asterisk 1.6.2x 1.6.1x

after one of the clients sends bye 

asterisk first sends invite to other side 

then after 200 ok it sends bye 

I am not sure but that could be some missconfiguration issue or a bug? 

 

so it's like this:

side A sends bye to asterisk, asterisk responds with 200 OK to side A, then
it sends INVITE to side B, expects 200 OK from side B, and then sends ACK
and BYE to side B

 

Thanks,

 

Josip

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[asterisk-users] Check if a variable is set

[Ishfaq Malik]

Hi

Is there any way to check if a variable is set in asterisk? I've had a 
look around and can't find a purpose built function for it.

I'm going to be using it to see if an argument has been passed with a 
macro or not (e.g. see if ${ARG3} is set or not)

Thanks in advance

Ish
-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062

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Re: [asterisk-users] Queues with unavailable members

[Benny Amorsen]

"C. Chad Wallace"  writes:

> OK, I decided to write it up in AEL.  It's incomplete and untested, but
> it probably gets the idea across a little better.
>
> context agentcalls {
>   _2XX => {
> Set(AGENT=${EXTEN});  // Assuming agent ID is extension.
> 
> if (${EPOCH}>${DB(AgentPaused/${AGENT})}) {
>   // Let the call through to the cell phone
>   Dial(...);
>
>   if () {
> // Flag agent as paused for the next 30 seconds.
> Set(DB(AgentPaused/${AGENT})=$[${EPOCH}+30]);
>   };
> }
> else {
>   // Agent still paused.
> };
>   };
> };

I was going in the same direction at the end of my first mail, but I
hadn't written any code. There is a problem though: The Queue
application will keep sending calls to the Local channel, which have to
be rejected, over and over.

Would it perhaps work to simply Wait(30) if the call is rejected by the
phone? If the Queue assumes that the phone is busy for those 30 seconds,
I have accomplished my goal. It's worth a shot.


/Benny


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Re: [asterisk-users] multiple call

[Dovid Bender]

- Original Message - 
From: "Matt Riddell" 
To: 
Sent: Thursday, October 15, 2009 09:50
Subject: Re: [asterisk-users] multiple call


> On 15/10/09 4:42 AM, Faheem wrote:
>> Through Asterisk AMI, you can not dial multiple number at the same time.
>> If you are going to implement a concurrent call scenario, then AMI would
>> not be a valid choice. Multiple calls can be implemented with callfile.
>
> Totally incorrect.
>
> We do hundreds of simultaneous calls at the same time using the Asterisk
> Manager.
>
> -- 
> Cheers,
>
> Matt Riddell
> Director
> ___
>
I would agree with Mat. We had more problems with call files then with the 
AMI. The only issue I had with the AMI (1.4.X) is that after multiple 
connections our script would connect to the AMI but the calls would not go 
out. We used astmanproxy and that resolved the issue. 


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Re: [asterisk-users] DTMF failing in some calls

[Dovid Bender]

Asterisk had some major issues with rfc2833 in 1.2.X. You should consider 
upgrading.
  - Original Message - 
  From: abdelkader 
  To: asterisk-users@lists.digium.com 
  Cc: mettichi ; jetcomm2...@yahoo.fr 
  Sent: Wednesday, October 14, 2009 11:36
  Subject: [asterisk-users] DTMF failing in some calls


  Hello,

  I am using Asterisk 1.2.33 under Debian ETCH linux.

  I have the following problem with DTMF:

  In my callback system, I calls an access DID. My system calls me back to my 
phone. It asks me for a password to let me dial an international number. If the 
authentication succeeds, I can dial a number in the system.

  Sometimes Asterisk catches only some of the digits I have entered. Sometimes, 
it duplicates some other digits. Sometimes, the system does not catch anything. 
And finally, sometimes it works properly.

  I am using rfc2833 as dtmf mode. 

  Please help.

  Thanks.


--


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Re: [asterisk-users] sporadic one-way audio

[Ishfaq Malik]

Hi

For the sporadic one way audio, check that the codec list in the snom 
phones is the same as set by the server. The codec list is in the RTP 
tab of the identities.

Hope that helps

Ish

Brent Davidson wrote:
> We have several offices running Asterisk version 1.4.20.1, and OSLEC  
> with Rhino R4FXO-EC and one running a Digium TDM800P card for interface 
> to analog lines.  All offices are running Snom 300 phones.  Phones all 
> have static addresses and are on the same physical network as the server.
>
> The problem we are having is that every so often we get someone calling 
> in where we can hear their voice, but they can't hear us.  If we 
> immediately call them back everything is fine.  The problem affects all 
> offices and also happens when making sip to sip calls from one snom 300 
> to another. 
>
> In addition we periodically have calls that drop off in the middle of a 
> conversation like the connection was lost.  I haven't been able to 
> replicate any of these problems and the people that are having them 
> can't seem to keep track of when they occur so I can go back and look in 
> the logs.
>
> I suspect that both problems may be related though.  Possibly a 
> registration issue?  Any ideas are welcome.
>
> Thanks,
> Brent Davidson
>
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>   

-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062

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Re: [asterisk-users] How to do a 3 party Warm Transfer in Asteriks 1.4

[Dovid Bender]

What you have described is an attended transfer so the docs that you found 
should help you.
  - Original Message - 
  From: Miguel Molina 
  To: Asterisk Users Mailing List - Non-Commercial Discussion 
  Sent: Tuesday, October 13, 2009 16:22
  Subject: Re: [asterisk-users] How to do a 3 party Warm Transfer in Asteriks 
1.4




We are running Asterisk 1.4 and need some help to determine how (if)  * 
supports 3 party warm transfers.  I've searched quite a bit  and all I can find 
is information on "attended transfers".  What we are looking for is: (1) 
external inbound call A comes to * extension B, caller A is placed on hold and 
extension B calls external third party C.  After explaining caller A issue to 
Party C, Ext B brings Caller A onto the call and introduces A to C.  After the 
into, ext B then drops off the call while A & C continue the call.  Any help 
would be appreciated.
Thanks Much, 
Jeff Johnson 
This email and any attached files are confidential and intended solely for the 
intended recipient(s). If you are not the named recipient you should not read, 
distribute, copy or alter this email. Any views or opinions expressed in this 
email are those of the author and do not represent those of NeturallySpeaking, 
LLC. 

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  Lee, John (Sydney) escribió: 
I don't think this can be done.
In your scenario, B is effectively the host and if B drops the line, both A and 
C will be dropped off as well.


From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff Johnson
Sent: Monday, 12 October 2009 2:25 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] How to do a 3 party Warm Transfer in Asteriks 1.4The 
built-in attended transfer asterisk has, works OK and won't drop the entire 
call when B hangs up to complete the attended transfer (with the * key). On the 
asterisk attended transfer, B calls C to explain the A issue (while A waits 
with MoH) and then hangs up (with the * key) to complete the attended transfer 
leaving A and C connected, but the part where you want all three talking do to 
the "warm" introduction cannot be done, unless you use a Meetme conference to 
put all of them on the same conversation.

  There may be some applications that do this without the need of a Meetme 
conference, maybe someone else can enlighten us.

  Cheers,

-- 
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center


--


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Re: [asterisk-users] The City of Amsterdam has been deploying asterisk throughout the city!

[Ivan Stepaniuk]

Ron Arts wrote:
> If you're interested, here is the press release:
> http://www.neonova.nl/nl/content/press/?tid=129735
>   

This list is for non-commercial discussion.

-- 
Iván Stepaniuk
Alba Fotónica S. L.
www.albafotonica.com


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Re: [asterisk-users] Linux/Asterisk on game consoles?

[Stelios Koroneos]

On Fri, 2009-10-16 at 08:16 +0200, Vincent wrote:
> Hello
> 
> I don't know much about game consoles, and I was wondering if someone
> had successfully ported Linux and Asterisk to the current hardware,
> ie. Nintendo Wii, Sony PS3, or Microsoft XBox360?
> 
> Thank you.
> 

I did it with PS3 and Asterisk 1.2 about a year ago
With Yellow Dog linux running on PS3
Was not using any of co-processors though, just the main cpu.

-- 
Stelios S. Koroneos

Digital OPSiS - Embedded Intelligence

Tel +30 210 9858296 Ext 100
Fax +30 210 9858298
http://www.digital-opsis.com


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