Re: [asterisk-users] Is H323 supported when installing Asterisk from Digium Yum repository?

[Vladimir Mikhelson]

Bruce,

Forgot to mention.

ooh323.conf -- configuration file.

-Vladimir



On 3/10/2011 9:32 PM, Bruce B wrote:
> But even with *asterisk16-addons-ooh323.x86_64* I don't see any of the
> command for h323 in CLI to work. So, I am missing something still. 
>
> On Thu, Mar 10, 2011 at 10:23 PM, Bruce B  > wrote:
>
> I see this in the Digium repository:
>
> *asterisk16-addons-ooh323.x86_64*
>
> Wouldn't that just do it without having to re-install from the source?
>
> Thanks
>
>
> On Thu, Mar 10, 2011 at 10:17 PM, Bruce B  > wrote:
>
> Can you please provide link to the RPM?
>
> Thanks
>
>
> On Thu, Mar 10, 2011 at 9:58 PM, Vladimir Mikhelson
> mailto:v...@mikhelson.com>> wrote:
>
> Alternatively, you can use OOH323 which is available with yum.
>
> I am using it for couple years with no major problems.
>  Developer is
> very responsive.  Just finished 1.8 related adjustments to
> OOH323,
> should be available in 1.8.4.
>
> -Vladimir
>
>
>
> On 3/10/2011  2:29 PM, Jose P. Espinal
> wrote:
> > Bruce B wrote:
> >> Hi everyone,
> >>
> >> Installed asterisk from yum repository but I think
> H.323 is not
> >> supported as I tried commands like this and they don't
> work:
> >
> > [snip]
> >
> >>
> >>
> >> Of course I can't go to source since I am using the
> repository. How
> >> can I install H.323. Is that OH323 I should look for?
> >
> > a. About the first question:
> > As Danny N. said on his response, H.323 is not available
> on yum, you
> > would have to do it from source.
> >
> > b. About OH323 and H.323:
> > You could try OH323, but personally (from past
> experiences) I would go
> > with H.323.
> >
> > Several weeks ago I posted some SlackBuilds (Script for
> making
> > slackware binary packages) of Asterisk to the list. You
> could download
> > the scripts and pickup some tips in order to compile the
> latest
> > version of H323plus (new name for H.323 project).
> >
> > IMO, H323plus project team has made a great job working
> on H.323
> > implementation.
> >
> > The scripts are located at:
> > http://packages.eslackware.com/slackbuilds/asterisk/
> >
> > If you run into any difficulties about any detail on the
> scripts, you
> > are free to contact me off-list (of course, for free. No
> charge, not
> > fee, etc)
> >
> >
> >
>
> --
> 
> _
> -- Bandwidth and Colocation Provided by
> http://www.api-digital.com --
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> every Thurs:
>   http://www.asterisk.org/hello
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>
>
>
>
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Re: [asterisk-users] Is H323 supported when installing Asterisk from Digium Yum repository?

[Vladimir Mikhelson]

pbx*CLI> ooh323
reload  set show
pbx*CLI> ooh323

-Vladimir


On 3/10/2011 9:32 PM, Bruce B wrote:
> But even with *asterisk16-addons-ooh323.x86_64* I don't see any of the
> command for h323 in CLI to work. So, I am missing something still. 
>
> On Thu, Mar 10, 2011 at 10:23 PM, Bruce B  > wrote:
>
> I see this in the Digium repository:
>
> *asterisk16-addons-ooh323.x86_64*
>
> Wouldn't that just do it without having to re-install from the source?
>
> Thanks
>
>
> On Thu, Mar 10, 2011 at 10:17 PM, Bruce B  > wrote:
>
> Can you please provide link to the RPM?
>
> Thanks
>
>
> On Thu, Mar 10, 2011 at 9:58 PM, Vladimir Mikhelson
> mailto:v...@mikhelson.com>> wrote:
>
> Alternatively, you can use OOH323 which is available with yum.
>
> I am using it for couple years with no major problems.
>  Developer is
> very responsive.  Just finished 1.8 related adjustments to
> OOH323,
> should be available in 1.8.4.
>
> -Vladimir
>
>
>
> On 3/10/2011  2:29 PM, Jose P. Espinal
> wrote:
> > Bruce B wrote:
> >> Hi everyone,
> >>
> >> Installed asterisk from yum repository but I think
> H.323 is not
> >> supported as I tried commands like this and they don't
> work:
> >
> > [snip]
> >
> >>
> >>
> >> Of course I can't go to source since I am using the
> repository. How
> >> can I install H.323. Is that OH323 I should look for?
> >
> > a. About the first question:
> > As Danny N. said on his response, H.323 is not available
> on yum, you
> > would have to do it from source.
> >
> > b. About OH323 and H.323:
> > You could try OH323, but personally (from past
> experiences) I would go
> > with H.323.
> >
> > Several weeks ago I posted some SlackBuilds (Script for
> making
> > slackware binary packages) of Asterisk to the list. You
> could download
> > the scripts and pickup some tips in order to compile the
> latest
> > version of H323plus (new name for H.323 project).
> >
> > IMO, H323plus project team has made a great job working
> on H.323
> > implementation.
> >
> > The scripts are located at:
> > http://packages.eslackware.com/slackbuilds/asterisk/
> >
> > If you run into any difficulties about any detail on the
> scripts, you
> > are free to contact me off-list (of course, for free. No
> charge, not
> > fee, etc)
> >
> >
> >
>
> --
> 
> _
> -- Bandwidth and Colocation Provided by
> http://www.api-digital.com --
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> every Thurs:
>   http://www.asterisk.org/hello
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>
>
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Re: [asterisk-users] Is H323 supported when installing Asterisk from Digium Yum repository?

[Bruce B]

But even with *asterisk16-addons-ooh323.x86_64* I don't see any of the
command for h323 in CLI to work. So, I am missing something still.

On Thu, Mar 10, 2011 at 10:23 PM, Bruce B  wrote:

> I see this in the Digium repository:
>
> *asterisk16-addons-ooh323.x86_64*
>
> Wouldn't that just do it without having to re-install from the source?
>
> Thanks
>
>
> On Thu, Mar 10, 2011 at 10:17 PM, Bruce B  wrote:
>
>> Can you please provide link to the RPM?
>>
>> Thanks
>>
>>
>> On Thu, Mar 10, 2011 at 9:58 PM, Vladimir Mikhelson 
>> wrote:
>>
>>> Alternatively, you can use OOH323 which is available with yum.
>>>
>>> I am using it for couple years with no major problems.  Developer is
>>> very responsive.  Just finished 1.8 related adjustments to OOH323,
>>> should be available in 1.8.4.
>>>
>>> -Vladimir
>>>
>>>
>>>
>>> On <3%2F10%2F2011> <3%2F10%2F2011>3/10/2011 2:29 PM, Jose P. Espinal
>>> wrote:
>>> > Bruce B wrote:
>>> >> Hi everyone,
>>> >>
>>> >> Installed asterisk from yum repository but I think H.323 is not
>>> >> supported as I tried commands like this and they don't work:
>>> >
>>> > [snip]
>>> >
>>> >>
>>> >>
>>> >> Of course I can't go to source since I am using the repository. How
>>> >> can I install H.323. Is that OH323 I should look for?
>>> >
>>> > a. About the first question:
>>> > As Danny N. said on his response, H.323 is not available on yum, you
>>> > would have to do it from source.
>>> >
>>> > b. About OH323 and H.323:
>>> > You could try OH323, but personally (from past experiences) I would go
>>> > with H.323.
>>> >
>>> > Several weeks ago I posted some SlackBuilds (Script for making
>>> > slackware binary packages) of Asterisk to the list. You could download
>>> > the scripts and pickup some tips in order to compile the latest
>>> > version of H323plus (new name for H.323 project).
>>> >
>>> > IMO, H323plus project team has made a great job working on H.323
>>> > implementation.
>>> >
>>> > The scripts are located at:
>>> > http://packages.eslackware.com/slackbuilds/asterisk/
>>> >
>>> > If you run into any difficulties about any detail on the scripts, you
>>> > are free to contact me off-list (of course, for free. No charge, not
>>> > fee, etc)
>>> >
>>> >
>>> >
>>>
>>> --
>>> _
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>>   http://www.asterisk.org/hello
>>>
>>> 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] Is H323 supported when installing Asterisk from Digium Yum repository?

[Bruce B]

I see this in the Digium repository:

*asterisk16-addons-ooh323.x86_64*

Wouldn't that just do it without having to re-install from the source?

Thanks

On Thu, Mar 10, 2011 at 10:17 PM, Bruce B  wrote:

> Can you please provide link to the RPM?
>
> Thanks
>
>
> On Thu, Mar 10, 2011 at 9:58 PM, Vladimir Mikhelson wrote:
>
>> Alternatively, you can use OOH323 which is available with yum.
>>
>> I am using it for couple years with no major problems.  Developer is
>> very responsive.  Just finished 1.8 related adjustments to OOH323,
>> should be available in 1.8.4.
>>
>> -Vladimir
>>
>>
>>
>> On <3%2F10%2F2011>3/10/2011 2:29 PM, Jose P. Espinal wrote:
>> > Bruce B wrote:
>> >> Hi everyone,
>> >>
>> >> Installed asterisk from yum repository but I think H.323 is not
>> >> supported as I tried commands like this and they don't work:
>> >
>> > [snip]
>> >
>> >>
>> >>
>> >> Of course I can't go to source since I am using the repository. How
>> >> can I install H.323. Is that OH323 I should look for?
>> >
>> > a. About the first question:
>> > As Danny N. said on his response, H.323 is not available on yum, you
>> > would have to do it from source.
>> >
>> > b. About OH323 and H.323:
>> > You could try OH323, but personally (from past experiences) I would go
>> > with H.323.
>> >
>> > Several weeks ago I posted some SlackBuilds (Script for making
>> > slackware binary packages) of Asterisk to the list. You could download
>> > the scripts and pickup some tips in order to compile the latest
>> > version of H323plus (new name for H.323 project).
>> >
>> > IMO, H323plus project team has made a great job working on H.323
>> > implementation.
>> >
>> > The scripts are located at:
>> > http://packages.eslackware.com/slackbuilds/asterisk/
>> >
>> > If you run into any difficulties about any detail on the scripts, you
>> > are free to contact me off-list (of course, for free. No charge, not
>> > fee, etc)
>> >
>> >
>> >
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>   http://www.asterisk.org/hello
>>
>> 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] Is H323 supported when installing Asterisk from Digium Yum repository?

[Bruce B]

Can you please provide link to the RPM?

Thanks

On Thu, Mar 10, 2011 at 9:58 PM, Vladimir Mikhelson wrote:

> Alternatively, you can use OOH323 which is available with yum.
>
> I am using it for couple years with no major problems.  Developer is
> very responsive.  Just finished 1.8 related adjustments to OOH323,
> should be available in 1.8.4.
>
> -Vladimir
>
>
>
> On 3/10/2011 2:29 PM, Jose P. Espinal wrote:
> > Bruce B wrote:
> >> Hi everyone,
> >>
> >> Installed asterisk from yum repository but I think H.323 is not
> >> supported as I tried commands like this and they don't work:
> >
> > [snip]
> >
> >>
> >>
> >> Of course I can't go to source since I am using the repository. How
> >> can I install H.323. Is that OH323 I should look for?
> >
> > a. About the first question:
> > As Danny N. said on his response, H.323 is not available on yum, you
> > would have to do it from source.
> >
> > b. About OH323 and H.323:
> > You could try OH323, but personally (from past experiences) I would go
> > with H.323.
> >
> > Several weeks ago I posted some SlackBuilds (Script for making
> > slackware binary packages) of Asterisk to the list. You could download
> > the scripts and pickup some tips in order to compile the latest
> > version of H323plus (new name for H.323 project).
> >
> > IMO, H323plus project team has made a great job working on H.323
> > implementation.
> >
> > The scripts are located at:
> > http://packages.eslackware.com/slackbuilds/asterisk/
> >
> > If you run into any difficulties about any detail on the scripts, you
> > are free to contact me off-list (of course, for free. No charge, not
> > fee, etc)
> >
> >
> >
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] Is H323 supported when installing Asterisk from Digium Yum repository?

[Vladimir Mikhelson]

Alternatively, you can use OOH323 which is available with yum.

I am using it for couple years with no major problems.  Developer is
very responsive.  Just finished 1.8 related adjustments to OOH323,
should be available in 1.8.4.

-Vladimir



On 3/10/2011 2:29 PM, Jose P. Espinal wrote:
> Bruce B wrote:
>> Hi everyone,
>>
>> Installed asterisk from yum repository but I think H.323 is not
>> supported as I tried commands like this and they don't work:
>
> [snip]
>
>>
>>
>> Of course I can't go to source since I am using the repository. How
>> can I install H.323. Is that OH323 I should look for?
>
> a. About the first question:
> As Danny N. said on his response, H.323 is not available on yum, you
> would have to do it from source.
>
> b. About OH323 and H.323:
> You could try OH323, but personally (from past experiences) I would go
> with H.323.
>
> Several weeks ago I posted some SlackBuilds (Script for making
> slackware binary packages) of Asterisk to the list. You could download
> the scripts and pickup some tips in order to compile the latest
> version of H323plus (new name for H.323 project).
>
> IMO, H323plus project team has made a great job working on H.323
> implementation.
>
> The scripts are located at:
> http://packages.eslackware.com/slackbuilds/asterisk/
>
> If you run into any difficulties about any detail on the scripts, you
> are free to contact me off-list (of course, for free. No charge, not
> fee, etc)
>
>
>

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Re: [asterisk-users] Is H323 supported when installing Asterisk from Digium Yum repository?

[Bruce B]

Thanks for the comments guys. So, I am bit lost. I went over the manual info
from Danny and that requires install of Asterisk again which I don't have
the luxury to because I use the Yum repository and everything is tied in
together.

For the slack build scripts I am kinda lost as well. I was hoping to get a
link to h.323 or oh323 rpm maybe

But if I have at the end of the day going either an rpm or not MUST I
re-install Asterisk? if so, I may as well start doing that now.

Thanks again guys.

On Thu, Mar 10, 2011 at 3:29 PM, Jose P. Espinal wrote:

> Bruce B wrote:
>
>> Hi everyone,
>>
>> Installed asterisk from yum repository but I think H.323 is not supported
>> as I tried commands like this and they don't work:
>>
>
> [snip]
>
>
>
>>
>> Of course I can't go to source since I am using the repository. How can I
>> install H.323. Is that OH323 I should look for?
>>
>
> a. About the first question:
> As Danny N. said on his response, H.323 is not available on yum, you would
> have to do it from source.
>
> b. About OH323 and H.323:
> You could try OH323, but personally (from past experiences) I would go with
> H.323.
>
> Several weeks ago I posted some SlackBuilds (Script for making slackware
> binary packages) of Asterisk to the list. You could download the scripts and
> pickup some tips in order to compile the latest version of H323plus (new
> name for H.323 project).
>
> IMO, H323plus project team has made a great job working on H.323
> implementation.
>
> The scripts are located at:
> http://packages.eslackware.com/slackbuilds/asterisk/
>
> If you run into any difficulties about any detail on the scripts, you are
> free to contact me off-list (of course, for free. No charge, not fee, etc)
>
>
>
> --
> Jose P. Espinal
> http://www.eSlackware.com
> IRC: Khratos @ #asterisk / -doc / -bugs
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>  http://www.asterisk.org/hello
>
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> To UNSUBSCRIBE or update options visit:
>  http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] 1.8.3 - IAX - echo - jitterbuffer

[sean darcy]

On Mon, Mar 7, 2011 at 6:53 PM, Dave Platt  wrote:
>> I'm using iaxagent on a Droid X to connect by IAX to 1.8.3 at the
>> office. 1.8.3 has sip phones. The audio is fine on the Droid X side. On
>> the office side, they hear an echo of _their_ speech, not mine.
>>
>> The office uses sip-providers generally without any echo problem.
>>
>> Where do I start to figure this out? How do I narrow it down? Can I
>> figure out if it is an iaxagent problem? Could using jitterbuffer cause
>> this?
>
> One thing you must consider, is that this echo they're hearing
> may be entirely an acoustic one, within (or around) the Droid
> itself.
>
> It's very possible for the microphone in a handset to
> pick up sound being emitted by the handset's speaker.  This
> acoustic feedback can occur within the handset itself (sound
> from the speaker "leaks" through the chassis of the handset and
> reaches the microphone from behind), via mechanical coupling
> through the handset body, or by the mic picking up the sound
> from the outside (after it has come out of the speaker
> into the air).
>
> The best way to determine whether this is the case, is
> probably to shut down the speaker and isolate the mic...
> plug in an earphone which has a separate mic on its cord,
> and see if the callers still report the echo.  If they do,
> it's due to electronic or digital goofs somewhere, but if they
> do not, it's due to acoustic feedback at the handset.
>
> There are (in principle) three ways to reduce or eliminate
> the echo:
>
> -  Damp it out physically - block the acoustic feedback
>   pathways.  In a small USB phone handset I have, I found
>   it necessary to "stuff" the open spaces inside the handset
>   with cotton and foam, to block the back-wave from the speaker
>   before it reached the microphone.
>
> -  Use software which has some sort of VOX (voice-operated
>   switch) detection or squelching... so that when the sound
>   level at the microphone is less than you'd get by speaking
>   into the mic, the handset "cuts off" the mic audio pathway
>   entirely, and sends only silence (or sends nothing at all,
>   although Asterisk doesn't always deal gracefully with this).
>
> -  Use a better handset.
>

I get no acoustic echo from the Droid X when I make a standard cell
call, even at full volume. Nor is the blogosphere full of complaints
about echos on Droid X. It's also physically one of the largest, if
not the largest, cell phone.

This may mean that the call function in android has really good echo cancelling.

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Re: [asterisk-users] Asterisk queues : command to run when a call isbeing bridged

[Mike]

Thank you, just what I needed to know!

 

Mike

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby
Sent: Thursday, March 10, 2011 5:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk queues : command to run when a call
isbeing bridged

 

On Thu, Mar 10, 2011 at 4:28 PM, Warren Selby  wrote:

 

In your AGI you should be able to read the custid variable, and I'm pretty
sure there is also a QUEUEMEMBER variable that's set with the agent
extension (not sure of the variable name, that's just off the top of my
head). 

 


Just had a look, and I think the variable is actually MEMBERINTERFACE.  You
will probably have to set "setinterfacevar=yes" in the specific queue
definition.

-- 
Thanks,
--Warren Selby, dCAP
http://www.selbytech.com

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Re: [asterisk-users] Connecting Asterisk to Siemens Hipath 3750

[Edwin Lam]

On 3/10/11 6:43 AM, Bobola Oke wrote:


The telco has a DB9 terminated interface straight to the PBX and I cannot make
sense out of the interface for the PBX. What kind of interface is this? How do I
connect the RJ48 of the PRI cards to make this whole setting work.


searching through this list's archive and found this:
http://lists.digium.com/pipermail/asterisk-users/2006-December/174258.html


--
Edwin Lam 
Systems Engineer, OfficeWyze, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/lookup?op=get&search=0xD6506D20


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Re: [asterisk-users] background music during call

[Danny Nicholas]

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Albert
Sent: Thursday, March 10, 2011 4:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] background music during call

 

Hi all,

I am looking for application or solution how i can play a sound during call.

The whole idea is to make a fun type of service, where user can call number,
than from ivr menu choose background music, f.e. traffic jam sounds, crying
baby,  and call friend to make him a prank :)

Does any one of you know such function or solution how to implement this ?


Thanks and regads,
Robert



This is a reasonably simple task.  Set up directories in your
/var/lib/asterisk/moh to hold the sound files for your "MOH".  In your IVR
dialplan set up something like this:

[select-sound]

Exten => s,1,playback(menu)

Exten => s,n,waitexten(5)

Exten => 1,1,goto(play1,s,1)

Exten => 2,1,goto(play2,s,1)

[play1]

Exten => s,1,SetMusicOnHold(one)

Exten => s,n,WaitMusicOnHold(20)

Exten => s,n,playback(vm-goodbye)

Exten => s,n,hangup

[play2]

Exten => s,1,SetMusicOnHold(two)

Exten => s,n,WaitMusicOnHold(20)

Exten => s,n,playback(vm-goodbye)

Exten => s,n,hangup

 

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[asterisk-users] background music during call

[Albert]

Hi all,

I am looking for application or solution how i can play a sound during call.

The whole idea is to make a fun type of service, where user can call
number, than from ivr menu choose background music, f.e. traffic jam
sounds, crying baby,  and call friend to make him a prank :)

Does any one of you know such function or solution how to implement this ?


Thanks and regads,
Albert


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[asterisk-users] background music during call

[Albert]

Hi all,

I am looking for application or solution how i can play a sound during call.

The whole idea is to make a fun type of service, where user can call
number, than from ivr menu choose background music, f.e. traffic jam
sounds, crying baby,  and call friend to make him a prank :)

Does any one of you know such function or solution how to implement this ?


Thanks and regads,
Robert


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Re: [asterisk-users] Asterisk queues : command to run when a call isbeing bridged

[Warren Selby]

On Thu, Mar 10, 2011 at 4:28 PM, Warren Selby  wrote:

>
> In your AGI you should be able to read the custid variable, and I'm pretty
> sure there is also a QUEUEMEMBER variable that's set with the agent
> extension (not sure of the variable name, that's just off the top of my
> head).
>
>
Just had a look, and I think the variable is actually MEMBERINTERFACE.  You
will probably have to set "setinterfacevar=yes" in the specific queue
definition.

-- 
Thanks,
--Warren Selby, dCAP
http://www.selbytech.com
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Re: [asterisk-users] Asterisk queues : command to run when a call isbeing bridged

[Warren Selby]

Just create an AGI and add it to the Queue() call:

[sales_queue]
; Sales Queue
exten => queuein,1,Verbose(Entering Sales Queue)
exten => queuein,n,Read(_custid,pls-entr-cust-id,4)
exten => queuein,n,Queue(sales,,,300,connect_script)
exten => queuein,n,Hangup()


In your AGI you should be able to read the custid variable, and I'm pretty sure 
there is also a QUEUEMEMBER variable that's set with the agent extension (not 
sure of the variable name, that's just off the top of my head). 



Thanks,
--Warren Selby, dCAP

On Mar 10, 2011, at 3:37 PM, "Danny Nicholas"  wrote:

> 
> From: asterisk-users-boun...@lists.digium.com 
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
> Sent: Thursday, March 10, 2011 3:30 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: [asterisk-users] Asterisk queues : command to run when a call 
> isbeing bridged
>  
> Hi,
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Re: [asterisk-users] Asked to transmit frame type slin, while native formats is 0x8 (alaw) read/write = 0x4 (ulaw)/0x8 (alaw)

[Matt Riddell]

On 11/03/11 7:52 AM, Nick Ustinov wrote:

These are the same for sip users and trunks

disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=g729

Who is asking to transmit frame type slin ?


Maybe transcodeviaslin or something with a "Local" channel?

--
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Matt Riddell
___

http://www.venturevoip.com/news.php (Daily Asterisk News)
http://www.venturevoip.com/exchange.php (Full ITSP Solution)
http://www.venturevoip.com/cc.php (Call Centre Solutions)

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Re: [asterisk-users] [1.4.21.2] Read() disconnects half-way through?

[Gilles]

On Thu, 10 Mar 2011 11:20:47 -0600, "Danny Nicholas"
 wrote:
>Just a guess - the problem may be with Originate instead of Read.   If you
>make a an extension  that does this:
>Exten => ,1,Goto(test,s,1)
>
>Does the behavior manifest itself as well?

Bingo! Works fine if I move this section and call the extension from
XLite instead. Thanks much for the tip.


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Re: [asterisk-users] Multiple SIP endpoint registrations

[Benny Amorsen]

"--[ UxBoD ]--"  writes:

> Hi,
> With Asterisk 1.8 is it now possible to register the same SIP account at
> multiple endpoints and for both to ring when the associated extension is
> dialed ?

No. Our solution is to give each phone an account and make a
Local/234@somecontext which dials SIP/234-foo&SIP/234-bar.

There are some challenges with Local channels, and we are working with
Olle E. Johansson to get some of them resolved. One of them is that in
some cases Asterisk cannot do reinvite if Local is done without /n.


/Benny


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Re: [asterisk-users] Asterisk queues : command to run when a call isbeing bridged

[Danny Nicholas]

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Thursday, March 10, 2011 3:30 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Asterisk queues : command to run when a call
isbeing bridged

 

Hi,

 

Is there any way to run a command (AGI script, whatever else) at that moment
when the call that was in the queue is being bridged to a specific agent?

 

An examples of what I would want to do with this is, for example, have
Asterisk ask the caller for his 4 digit customer ID before being put in the
Queue.  Once I know who the caller is being connected to (which agent) I'd
run a script that has the agent and the customer ID as parameters.

 

So the script needs to run only when we know who`s taking the call.

 

Where do I start, what function/option do I use?

 

Mike

 

Looks like a simple modification (famous last words) to your queue command:

core show application queue

 

  -= Info about application 'Queue' =-

 

[Synopsis]

Queue a call for a call queue

 

[Description]

  Queue(queuename[|options[|URL][|announceoverride][|timeout][|AGI]):

Queues an incoming call in a particular call queue as defined in
queues.conf.

This application will return to the dialplan if the queue does not exist, or

any of the join options cause the caller to not enter the queue.

The option string may contain zero or more of the following characters:

  'd' -- data-quality (modem) call (minimum delay).

  'h' -- allow callee to hang up by hitting '*', or whatver disconnect
sequence

 defined in the featuremap section in features.conf.

  'H' -- allow caller to hang up by hitting '*', or whatever disconnect
sequence

 defined in the featuremap section in features.conf.

  'n' -- no retries on the timeout; will exit this application and

 go to the next step.

  'i' -- ignore call forward requests from queue members and do nothing

 when they are requested.

  'r' -- ring instead of playing MOH

  't' -- allow the called user transfer the calling user by pressing '#'
or

 whatever blindxfer sequence defined in the featuremap section
in

 features.conf

  'T' -- to allow the calling user to transfer the call by pressing '#'
or

 whatever blindxfer sequence defined in the featuremap section
in

 features.conf

  'w' -- allow the called user to write the conversation to disk via
Monitor

 by pressing the automon sequence defined in the featuremap
section in

 features.conf

  'W' -- allow the calling user to write the conversation to disk via
Monitor

 by pressing the automon sequence defined in the featuremap
section in

 features.conf

  In addition to transferring the call, a call may be parked and then picked

up by another user, by transferring to the parking lot extension. See
features.conf.

  The optional URL will be sent to the called party if the channel supports

it.

  The optional AGI parameter will setup an AGI script to be executed on the

calling party's channel once they are connected to a queue member.

  The timeout will cause the queue to fail out after a specified number of

seconds, checked between each queues.conf 'timeout' and 'retry' cycle.

  This application sets the following channel variable upon completion:

  QUEUESTATUSThe status of the call as a text string, one of

 TIMEOUT | FULL | JOINEMPTY | LEAVEEMPTY | JOINUNAVAIL |
LEAVEUNAVAIL

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[asterisk-users] Asterisk queues : command to run when a call is being bridged

[Mike]

Hi,

 

Is there any way to run a command (AGI script, whatever else) at that moment
when the call that was in the queue is being bridged to a specific agent?

 

An examples of what I would want to do with this is, for example, have
Asterisk ask the caller for his 4 digit customer ID before being put in the
Queue.  Once I know who the caller is being connected to (which agent) I'd
run a script that has the agent and the customer ID as parameters.

 

So the script needs to run only when we know who`s taking the call.

 

Where do I start, what function/option do I use?

 

Mike

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Re: [asterisk-users] Dialplan: funcionality testing

[Luiz Gustavo Chiaretto]

Hi Paul, 

I've tried to run script runtests.py and a problem appeared ... 

root@voip01:/testsuite# ./runtests.py 
Traceback (most recent call last): 
File "./runtests.py", line 21, in  
from asterisk.version import AsteriskVersion 
ImportError: No module named version 

I am trying to resolve this dependency for hours. Do you know what python 
module do i need install to resolve this problem? 



Luiz Gustavo Chiaretto 



- Original Message -
From: "Paul Belanger"  
To: asterisk-users@lists.digium.com 
Sent: Thursday, March 10, 2011 3:27:04 PM 
Subject: Re: [asterisk-users] Dialplan: funcionality testing 

On 11-03-10 01:14 PM, Luiz Gustavo Chiaretto wrote: 
> I thought the Testsuite was a tool used for asterisk developers and not used 
> for functionality testing. 
> 
It is used to test all aspects of Asterisk. So, if you have something 
specific you want to test, write a test for it, submit it upstream 
(issue tracker), and we cab merge it into the testsuite. Then every 
each time svn commit is made, http://bamboo.asterisk.org will run the 
testsuite against it. 

It is beneficial to both you and the community :) 

-- 
Paul Belanger 
Digium, Inc. | Software Developer 
twitter: pabelanger | IRC: pabelanger (Freenode) 
Check us out at: http://digium.com & http://asterisk.org 

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Re: [asterisk-users] Is H323 supported when installing Asterisk from Digium Yum repository?

[Jose P. Espinal]

Bruce B wrote:

Hi everyone,

Installed asterisk from yum repository but I think H.323 is not 
supported as I tried commands like this and they don't work:


[snip]




Of course I can't go to source since I am using the repository. How can 
I install H.323. Is that OH323 I should look for?


a. About the first question:
As Danny N. said on his response, H.323 is not available on yum, you 
would have to do it from source.


b. About OH323 and H.323:
You could try OH323, but personally (from past experiences) I would go 
with H.323.


Several weeks ago I posted some SlackBuilds (Script for making slackware 
binary packages) of Asterisk to the list. You could download the scripts 
and pickup some tips in order to compile the latest version of H323plus 
(new name for H.323 project).


IMO, H323plus project team has made a great job working on H.323 
implementation.


The scripts are located at:
http://packages.eslackware.com/slackbuilds/asterisk/

If you run into any difficulties about any detail on the scripts, you 
are free to contact me off-list (of course, for free. No charge, not 
fee, etc)




--
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http://www.eSlackware.com
IRC: Khratos @ #asterisk / -doc / -bugs

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Re: [asterisk-users] Is H323 supported when installing Asterisk fromDigium Yum repository?

[Danny Nicholas]

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce B
Sent: Thursday, March 10, 2011 1:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Is H323 supported when installing Asterisk
fromDigium Yum repository?

 

Hi everyone,

 

Installed asterisk from yum repository but I think H.323 is not supported as
I tried commands like this and they don't work:

 

*   h.323 debug: Enable chan_h323 debug
*   h.323 gk cycle: Manually re-register with the Gatekeper
*   h.323 hangup: Manually try to hang up a call
*   h.323 no debug: Disable chan_h323 debug
*   h.323 no trace: Disable H.323 Stack Tracing
*   h.323 show codecs: Show enabled codecs
*   h.323 show tokens: Manually try to hang up a call
*   h.323 trace: Enable H.323 Stack Tracing

 

Of course I can't go to source since I am using the repository. How can I
install H.323. Is that OH323 I should look for?

 

Thanks

 

Having just jumped through the H323 hoop I can say with reasonable certainty
that H323 is NOT a direct YUM option.   This link will come in handy if you
decide to go the "manual route".

http://astrecipes.net/index.php?n=286

 

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[asterisk-users] Is H323 supported when installing Asterisk from Digium Yum repository?

[Bruce B]

Hi everyone,

Installed asterisk from yum repository but I think H.323 is not supported as
I tried commands like this and they don't work:


   - *h.323 debug*: Enable chan_h323 debug
   - *h.323 gk cycle*: Manually re-register with the Gatekeper
   - *h.323 hangup*: Manually try to hang up a call
   - *h.323 no debug*: Disable chan_h323 debug
   - *h.323 no trace*: Disable H.323 Stack Tracing
   - *h.323 show codecs*: Show enabled codecs
   - *h.323 show tokens*: Manually try to hang up a call
   - *h.323 trace*: Enable H.323 Stack Tracing


Of course I can't go to source since I am using the repository. How can I
install H.323. Is that OH323 I should look for?

Thanks
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Re: [asterisk-users] Asked to transmit frame type slin, while native formats is 0x8 (alaw) read/write = 0x4 (ulaw)/0x8 (alaw)

[Nick Ustinov]

These are the same for sip users and trunks

disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=g729

Who is asking to transmit frame type slin ?

Nick

On Thu, Mar 10, 2011 at 1:02 AM, Paul Belanger  wrote:
> On 11-03-09 02:26 PM, Nick Ustinov wrote:
>> Using asterisk 1.8.4-rc2
>>
>> What could be the cause?
>>
> Your allow / disallow settings for codecs.
>
> --
> Paul Belanger
> Digium, Inc. | Software Developer
> twitter: pabelanger | IRC: pabelanger (Freenode)
> Check us out at: http://digium.com & http://asterisk.org
>
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Re: [asterisk-users] Dialplan: funcionality testing

[Paul Belanger]

On 11-03-10 01:14 PM, Luiz Gustavo Chiaretto wrote:
> I thought the Testsuite was a tool used for asterisk developers and not used 
> for functionality testing. 
> 
It is used to test all aspects of Asterisk.  So, if you have something
specific you want to test, write a test for it, submit it upstream
(issue tracker), and we cab merge it into the testsuite.  Then every
each time svn commit is made, http://bamboo.asterisk.org will run the
testsuite against it.

It is beneficial to both you and the community :)

-- 
Paul Belanger
Digium, Inc. | Software Developer
twitter: pabelanger | IRC: pabelanger (Freenode)
Check us out at: http://digium.com & http://asterisk.org

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Re: [asterisk-users] Dialplan: funcionality testing

[Luiz Gustavo Chiaretto]

Paul, 

Thanks for your answer, 

I thought the Testsuite was a tool used for asterisk developers and not used 
for functionality testing. 

I'll try to use it in my tests. 

Best Regards, 



Luiz Gustavo Chiaretto 



- Original Message -
From: "Paul Belanger"  
To: asterisk-users@lists.digium.com 
Sent: Thursday, March 10, 2011 3:01:25 PM 
Subject: Re: [asterisk-users] Dialplan: funcionality testing 

On 11-03-10 12:52 PM, Luiz Gustavo Chiaretto wrote: 
> Hello, 
> 
> I've tried to use adhearsion, but i think it's used for stress testing, not 
> for funcionality testing. 
> 
> Somebody knows somehow that i can test my dialplan? 
> 
Write a test for the testsuite[1]. 

[1] http://svn.asterisk.org/svn/testsuite/asterisk/trunk/ 

-- 
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Digium, Inc. | Software Developer 
twitter: pabelanger | IRC: pabelanger (Freenode) 
Check us out at: http://digium.com & http://asterisk.org 

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Re: [asterisk-users] Dialplan: funcionality testing

[Paul Belanger]

On 11-03-10 12:52 PM, Luiz Gustavo Chiaretto wrote:
> Hello, 
> 
> I've tried to use adhearsion, but i think it's used for stress testing, not 
> for funcionality testing. 
> 
> Somebody knows somehow that i can test my dialplan? 
> 
Write a test for the testsuite[1].

[1] http://svn.asterisk.org/svn/testsuite/asterisk/trunk/

-- 
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Digium, Inc. | Software Developer
twitter: pabelanger | IRC: pabelanger (Freenode)
Check us out at: http://digium.com & http://asterisk.org

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[asterisk-users] Dialplan: funcionality testing

[Luiz Gustavo Chiaretto]

Hello, 

I've tried to use adhearsion, but i think it's used for stress testing, not for 
funcionality testing. 

Somebody knows somehow that i can test my dialplan? 

Best Regards, 



Luiz Gustavo Chiaretto 



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Re: [asterisk-users] Metaswitch to Asterisk problems

[C F]

Make sure you get some DNIS, on the meta you might have to play around
to get it, it might come in in the form of some sip headers or as
asterisk expects it as an extension. In any event, once asterisk knows
which extension (DID) it belongs to just send it to VM.


On Thu, Mar 10, 2011 at 10:02 AM, Chris Ledford
 wrote:
> I am setting up VM off Metaswitch due to a problem with Metaswitch VM. I
> have a couple days to prove this works and I need a little assist please.
>
>
>
> I am using TRIXBOX 2.6.2.5 and have the Meta SIP trunk up. I have extensions
> built that can talk to each other. I took a trace on the TRIXBOX that shows
> when I dial my test phone on Metaswitch it goes to VM after a couple rings
> and the call goes to my TRIXBOX and asterisk plays a RTP message saying that
> the number cannot be contacted.
>
>
>
> I don’t understand how the TN on Metaswitch translates to the TRIXBOX VM
> account.
>
>
>
> Can I please get a response on or off forum for some assistance in what I am
> doing wrong or if there is a good post or website with a guide on this
> config..
>
>
>
> I have looked at several online over that last couple of weeks and they are
> dated and not running TRIXBOX 2.6.2.5, so trying to conform what I am
> reading to my config is confusing.
>
>
>
> Thanks in advance.
>
>
>
> V/r
>
>
>
> Chris Ledford
>
> CCNA/CCSP/CCNP Voice
>
> Comptia A+/Net+/Linux+/Sec+
>
> EWC/CTTC(sw) USN
> T3 Engineer
> http://navy.togetherweserved.com/profile/13552
>
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> _
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>               http://www.asterisk.org/hello
>
> 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] [1.4.21.2] Read() disconnects half-way through?

[Danny Nicholas]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles
Sent: Thursday, March 10, 2011 10:53 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] [1.4.21.2] Read() disconnects half-way through?

Hello

I'm using the Read() function to play a message prompting for the
user to type a number followed by the # key to validate, with a 30s
time-out and 2 tries:
==
[test]
exten => s,1,Wait(2)
exten => s,n,Answer

;typed DTMF: prompt for number to dial: 2 tries, 30s time-tout
exten => s,n(nbr2call),Read(NBR2CALL,please-type-number,,,2,30)

exten => s,n,GotoIf($[${LEN(${NBR2CALL})} != 10]?nbr2call)
exten => s,n,Playback(phone:${NBR2CALL},say)

exten => s,n(end),Wait(2)
exten => s,n,Hangup()
==

I notice that it sometimes works fine, but sometimes, Asterisk hangs
up while I'm still typing:
==
CLI> originate Zap/1/5551234 extension s@test

Executing [s@test:3] Read("Zap/1-1",
"NBR2CALL|please-type-number|||2|30") in new stack
--  Playing 'please-type-number' (language 'fr')
-- User disconnected
 == Spawn extension (test, s, 3) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
==

Has someone also experienced this? Is Read() unreliable and if that's
the case, should I use another way to let a user type a phone number?

Thank you.


Just a guess - the problem may be with Originate instead of Read.   If you
make a an extension  that does this:
Exten => ,1,Goto(test,s,1)

Does the behavior manifest itself as well?



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[asterisk-users] [1.4.21.2] Read() disconnects half-way through?

[Gilles]

Hello

I'm using the Read() function to play a message prompting for the
user to type a number followed by the # key to validate, with a 30s
time-out and 2 tries:
==
[test]
exten => s,1,Wait(2)
exten => s,n,Answer

;typed DTMF: prompt for number to dial: 2 tries, 30s time-tout
exten => s,n(nbr2call),Read(NBR2CALL,please-type-number,,,2,30)

exten => s,n,GotoIf($[${LEN(${NBR2CALL})} != 10]?nbr2call)
exten => s,n,Playback(phone:${NBR2CALL},say)

exten => s,n(end),Wait(2)
exten => s,n,Hangup()
==

I notice that it sometimes works fine, but sometimes, Asterisk hangs
up while I'm still typing:
==
CLI> originate Zap/1/5551234 extension s@test

Executing [s@test:3] Read("Zap/1-1",
"NBR2CALL|please-type-number|||2|30") in new stack
--  Playing 'please-type-number' (language 'fr')
-- User disconnected
 == Spawn extension (test, s, 3) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
==

Has someone also experienced this? Is Read() unreliable and if that's
the case, should I use another way to let a user type a phone number?

Thank you.


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Re: [asterisk-users] One Way Audio

[Tim King]

Still not working now that audio is restored jitter makes it inaudible?  I
am ready to move this to commercial if someone can tell me how I need to pay
for support,

Thanks

Tim

On Thu, Mar 10, 2011 at 10:19 AM, Tim King  wrote:

> It looks like the issue was my provider enforcing a codec translation that
> was not working.
>
>
> On Thu, Mar 10, 2011 at 9:21 AM, Satish Patel wrote:
>
>> Also it could be the routing issue as well.
>>
>> --
>> Sent from my iPhone
>>
>> On Mar 9, 2011, at 7:43 PM, Duncan Turnbull 
>> wrote:
>>
>> So that suggests audio is flowing as follows in a unidirectional manner
>>
>> 199.173.66.22.53102 > 74.204.4.5.11732  IP 74.204.4.5.11732 >
>> 209.216.2.203.60362
>>
>>
>> Somewhere near the end this pops up which is slightly different, I am
>> guessing 74.204.4.5 is your asterisk box
>>
>> 19:18:36.389548 IP 74.204.4.5.11732 > 174.133.195.194.18364: UDP, length
>> 172
>>
>> I am not sure why this is happening or if its still part of the same
>> conversation
>>
>> Overall it looks a bit like the asterisk box thinks it needs to send rtp
>> to a different location than perhaps its meant to i.e. its asymmetric - you
>> can check the sdp in the sip invite to see where media is expected to be
>> sent to
>>
>> There is no rtp coming back from 209.216.2.203 so possibly this is device
>> that isn't meant to be part of the conversation and either doesn't exist or
>> is not expecting anything and thus not responding
>>
>> What are the addresses of the devices in this conversation? so that you
>> can match the traffic to device
>>
>> Cheers Duncan
>>
>> On 10/03/2011, at 1:20 PM, Tim King wrote:
>>
>> It looks like this:
>> 19:18:34.782016 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:34.789527 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:34.802064 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:34.809757 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:34.821855 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:34.829598 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:34.842015 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:34.849764 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:34.861902 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:34.869568 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:34.881882 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:34.889739 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:34.901882 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:34.909612 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:34.921984 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:34.929664 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:34.941855 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:34.949589 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:34.962003 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:34.969592 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:34.981851 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:34.989543 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:35.002006 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:35.009973 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:35.022008 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:35.029539 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:35.042071 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:35.049561 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:35.062008 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:35.069612 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:35.081986 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:35.089519 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:35.101918 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:35.109722 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:35.122021 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:35.129590 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:35.141878 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:35.149709 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:35.161886 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:35.169561 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:35.181879 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:35.189710 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:35.201965 IP 199.173.66.22.53103 > 74.204.4.5.11733:

Re: [asterisk-users] [1.4] Reading phone number the French way?

[Gilles]

On Thu, 10 Mar 2011 15:10:19 + (UTC), t...@softins.co.uk (Tony
Mountifield) wrote:
>> -- Executing [@internal:6] Playback("SIP/xlite-02a56004",
>> "phone:0892123456}|say") in new stack
...

>You have a spurious } after 0892123456 which is preventing it
>from matching the pattern in say.conf

Some eyesight... Thanks a bunch!


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Re: [asterisk-users] One Way Audio

[Tim King]

It looks like the issue was my provider enforcing a codec translation that
was not working.

On Thu, Mar 10, 2011 at 9:21 AM, Satish Patel  wrote:

> Also it could be the routing issue as well.
>
> --
> Sent from my iPhone
>
> On Mar 9, 2011, at 7:43 PM, Duncan Turnbull  wrote:
>
> So that suggests audio is flowing as follows in a unidirectional manner
>
> 199.173.66.22.53102 > 74.204.4.5.11732  IP 74.204.4.5.11732 >
> 209.216.2.203.60362
>
>
> Somewhere near the end this pops up which is slightly different, I am
> guessing 74.204.4.5 is your asterisk box
>
> 19:18:36.389548 IP 74.204.4.5.11732 > 174.133.195.194.18364: UDP, length
> 172
>
> I am not sure why this is happening or if its still part of the same
> conversation
>
> Overall it looks a bit like the asterisk box thinks it needs to send rtp to
> a different location than perhaps its meant to i.e. its asymmetric - you can
> check the sdp in the sip invite to see where media is expected to be sent to
>
> There is no rtp coming back from 209.216.2.203 so possibly this is device
> that isn't meant to be part of the conversation and either doesn't exist or
> is not expecting anything and thus not responding
>
> What are the addresses of the devices in this conversation? so that you can
> match the traffic to device
>
> Cheers Duncan
>
> On 10/03/2011, at 1:20 PM, Tim King wrote:
>
> It looks like this:
> 19:18:34.782016 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:34.789527 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:34.802064 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:34.809757 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:34.821855 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:34.829598 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:34.842015 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:34.849764 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:34.861902 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:34.869568 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:34.881882 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:34.889739 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:34.901882 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:34.909612 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:34.921984 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:34.929664 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:34.941855 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:34.949589 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:34.962003 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:34.969592 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:34.981851 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:34.989543 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.002006 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.009973 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.022008 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.029539 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.042071 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.049561 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.062008 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.069612 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.081986 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.089519 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.101918 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.109722 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.122021 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.129590 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.141878 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.149709 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.161886 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.169561 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.181879 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.189710 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.201965 IP 199.173.66.22.53103 > 74.204.4.5.11733: UDP, length 60
> 19:18:35.201974 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.209552 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.221898 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.229625 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 1

Re: [asterisk-users] [1.4] Reading phone number the French way?

[Tony Mountifield]

In article ,
Gilles  wrote:
> On Thu, 10 Mar 2011 14:37:45 +0100, Gilles 
> wrote:
> >I figured out how extensions.conf and say.conf work and posted my
> >results in the reply to Dave.
> 
> Noticed something strange, though: 0800123456 is played OK (ie.
> 0.800.12.34.56) , but 092123456 is played digit by digit (0.8.9.2,
> etc.):
> 
> == say.conf
> 
> ;1-9
> _[n]um:X => digits/${SAY}
> 
> ;10-99
> _[n]um:1X => digits/${SAY}
> _[n]um:[2-9]0 =>  digits/${SAY}
> _[n]um:[2-6]1 => digits/${SAY:0:1}0, vm-and, digits/${SAY:1}
> _[n]um:71 => digits/60, vm-and, num:1${SAY:1}
> _[n]um:7X => digits/60, num:1${SAY:1}
> _[n]um:9X => digits/80, num:1${SAY:1}
> _[n]um:[2-9][1-9] =>  digits/${SAY:0:1}0, num:${SAY:1}
> 
> ;100-999
> _[n]um:100 => digits/hundred
> _[n]um:1XX => digits/hundred, num:${SAY:1}
> _[n]um:[2-9]00 => num:${SAY:0:1}, digits/hundred
> _[n]um:[2-9]XX => num:${SAY:0:1}, digits/hundred, num:${SAY:1}
> 
> ;0800XX -> 0899XX
> ;_pho[n]e:08 => num:${SAY:0:1}, num:${SAY:1:3},
> num:${SAY:4:2}, num:${SAY:6:2}, num:${SAY:8:2}
> 
> == CLI
> 
> -- Executing [@internal:4] Playback("SIP/xlite-02a56004",
> "phone:0810009032|say") in new stack
> --  Playing 'digits/0' (language 'fr')
> --  Playing 'digits/8' (language 'fr')
> --  Playing 'digits/hundred' (language 'fr')
> --  Playing 'digits/10' (language 'fr')
> --  Playing 'digits/0' (language 'fr')
> --  Playing 'digits/0' (language 'fr')
> --  Playing 'digits/90' (language 'fr')
> --  Playing 'digits/30' (language 'fr')
> --  Playing 'digits/2' (language 'fr')
> 
> -- Executing [@internal:6] Playback("SIP/xlite-02a56004",
> "phone:0892123456}|say") in new stack
^
> --  Playing 'digits/0' (language 'fr')
> --  Playing 'digits/8' (language 'fr')
> --  Playing 'digits/9' (language 'fr')
> --  Playing 'digits/2' (language 'fr')
> --  Playing 'digits/1' (language 'fr')
> --  Playing 'digits/2' (language 'fr')
> --  Playing 'digits/3' (language 'fr')
> --  Playing 'digits/4' (language 'fr')
> --  Playing 'digits/5' (language 'fr')
> -- Executing [@internal:7] Hangup("SIP/xlite-02a56004", "") in new
> stack

You have a spurious } after 0892123456 which is preventing it
from matching the pattern in say.conf

Cheers
Tony
-- 
Tony Mountifield
Work: t...@softins.co.uk - http://www.softins.co.uk
Play: t...@mountifield.org - http://tony.mountifield.org

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[asterisk-users] Metaswitch to Asterisk problems

[Chris Ledford]

I am setting up VM off Metaswitch due to a problem with Metaswitch VM. I have a 
couple days to prove this works and I need a little assist please.

I am using TRIXBOX 2.6.2.5 and have the Meta SIP trunk up. I have extensions 
built that can talk to each other. I took a trace on the TRIXBOX that shows 
when I dial my test phone on Metaswitch it goes to VM after a couple rings and 
the call goes to my TRIXBOX and asterisk plays a RTP message saying that the 
number cannot be contacted.

I don't understand how the TN on Metaswitch translates to the TRIXBOX VM 
account.

Can I please get a response on or off forum for some assistance in what I am 
doing wrong or if there is a good post or website with a guide on this config..

I have looked at several online over that last couple of weeks and they are 
dated and not running TRIXBOX 2.6.2.5, so trying to conform what I am reading 
to my config is confusing.

Thanks in advance.

V/r

Chris Ledford
CCNA/CCSP/CCNP Voice
Comptia A+/Net+/Linux+/Sec+
EWC/CTTC(sw) USN
T3 Engineer
http://navy.togetherweserved.com/profile/13552
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Re: [asterisk-users] [1.4] Reading phone number the French way?

[Gilles]

On Thu, 10 Mar 2011 08:49:43 -0600, "Danny Nicholas"
 wrote:
>"sip set debug peer" is probably going to be
>your best bet for say.conf debugging.  If you do "sip set debug peer "
>as opposed to "core set debug 10", you are getting "nodal debugging" as
>opposed to "general debugging".

Thanks for the tip. It just adds SIP messages to the existing display.
Looks like Asterisk doesn't provide a way to check how it uses
patterns in say.conf.


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Re: [asterisk-users] TDM410P & dahdi driver == no lights?

[Brian Henning]

Hi Shaun,

Thanks so much for your response!

Unfortunately this is a production server now, so I'm a little wary of
testing out non-release builds (if it were not production, I would
definitely test).

Seeing consistent call operation plus your info below quells my concerns.
I'll just be happy without lights.

Cheers,
~Brian

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shaun Ruffell
Sent: Tuesday, March 08, 2011 1:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] TDM410P & dahdi driver == no lights?

On 03/07/2011 08:22 PM, Brian Henning wrote:
> 
> I have just installed an Asterisk server with a Digium TDM410P card with 3
> FXO modules (no module in the 4th slot).
> 
> It's lived on two different machines (a test machine, which had Linux
kernel
> 2.6.28, and a new dedicated machine which has Linux kernel 2.6.32).
> 
> On the test machine (2.6.28), I used the Zaptel drivers.  Once the kernel
> modules were loaded, the lights on the TDM410P came on green for the
> installed FXO modules.
> 
> On the new server, the Zaptel drivers wouldn't build so I switched over to
> dahdi.  Everything seems to be working, EXCEPT there are no lights on the
> TDM410P!  I guess I can ignore that the lights aren't lit up, because it
> seems to be functioning as expected (I can dial out and receive incoming
> calls)...but it's disconcerting that the lights aren't on.  Yes, the Molex
> power connector is connected (although I think that's only needed by FXS
> modules).
> 
> I've tried google searches but haven't found anything mentioning this odd
> behavior.  Is this expected?
> 

Brian,

I haven't run it, but looking through the code this appears to be a
regression I added in 2.4.1 / current trunk (introduced in r9720 [1]).
I've opened issue 18939 and attached some patches if you want to try
them.  That way I can add your reported by / tested-by information if
you would like.

[1] http://svn.asterisk.org/view/dahdi?view=revision&revision=9720
[2] https://issues.asterisk.org/view.php?id=18939

Thanks,
-- 
Shaun Ruffell
Digium, Inc. | Linux Kernel Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] [1.4] Reading phone number the French way?

[Danny Nicholas]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles
Sent: Thursday, March 10, 2011 8:39 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] [1.4] Reading phone number the French way?

On Thu, 10 Mar 2011 08:34:51 -0600, "Danny Nicholas"
 wrote:
>== say.conf
>
>;1-9
>_[n]um:X => digits/${SAY}
>
>;10-99
>_[n]um:1X => digits/${SAY}
>_[n]um:[2-9]0 =>  digits/${SAY}
>_[n]um:[2-6]1 => digits/${SAY:0:1}0, vm-and, digits/${SAY:1}
>_[n]um:71 => digits/60, vm-and, num:1${SAY:1}
>_[n]um:7X => digits/60, num:1${SAY:1}
>_[n]um:9X => digits/80, num:1${SAY:1}
>_[n]um:[2-9][1-9] =>  digits/${SAY:0:1}0, num:${SAY:1}
>
>;100-999
>_[n]um:100 => digits/hundred
>_[n]um:1XX => digits/hundred, num:${SAY:1}
>_[n]um:[2-9]00 => num:${SAY:0:1}, digits/hundred
>_[n]um:[2-9]XX => num:${SAY:0:1}, digits/hundred, num:${SAY:1}
>
>;0800XX -> 0899XX
>;_pho[n]e:08 => num:${SAY:0:1}, num:${SAY:1:3},
>num:${SAY:4:2}, num:${SAY:6:2}, num:${SAY:8:2}
>
>== CLI
>
>-- Executing [@internal:4] Playback("SIP/xlite-02a56004",
>"phone:0810009032|say") in new stack
>--  Playing 'digits/0' (language 'fr')
>--  Playing 'digits/8' (language 'fr')
>--  Playing 'digits/hundred' (language 'fr')
>--  Playing 'digits/10' (language 'fr')
>--  Playing 'digits/0' (language 'fr')
>--  Playing 'digits/0' (language 'fr')
>--  Playing 'digits/90' (language 'fr')
>--  Playing 'digits/30' (language 'fr')
>--  Playing 'digits/2' (language 'fr')
>
>-- Executing [@internal:6] Playback("SIP/xlite-02a56004",
>"phone:0892123456}|say") in new stack
>--  Playing 'digits/0' (language 'fr')
>--  Playing 'digits/8' (language 'fr')
>--  Playing 'digits/9' (language 'fr')
>--  Playing 'digits/2' (language 'fr')
>--  Playing 'digits/1' (language 'fr')
>--  Playing 'digits/2' (language 'fr')
>--  Playing 'digits/3' (language 'fr')
>--  Playing 'digits/4' (language 'fr')
>--  Playing 'digits/5' (language 'fr')
>-- Executing [@internal:7] Hangup("SIP/xlite-02a56004", "") in new
>stack
>== 

>This one is easy.  You have a specific pattern to match 0800.  092 has no
>pattern and therefore defaults back to an "Asterisk Standard Playback".

Sorry, it was a typo: I did mean numbers starting with "0800" or
"0892". Those are the FR equivalent of 1.800 numbers in the US.

Going through the patterns above, it should use the same patterns for
either numbers, but for some reason, fails for 0892 :-/

BTW, is there a way to enable say.conf debugging in the CLI, so I
could check which patterns it goes through to analyze a number?

Thank you.

Thought that was too easy...  "sip set debug peer" is probably going to be
your best bet for say.conf debugging.  If you do "sip set debug peer "
as opposed to "core set debug 10", you are getting "nodal debugging" as
opposed to "general debugging".


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Re: [asterisk-users] Connecting Asterisk to Siemens Hipath 3750

[Bobola Oke]

Hi Josue,

Thanks for your reply.

I have similar settings on my server with CRC4 enabled.

I have a Digium TE210 card on the asterisk server.

The present topology of the network:

Telco<---E1>Siemens HiPath 3750<->Analog
Lines

I want this new scenario

Telco<---E1-->Asterisk<--E1>Siemens<--->Analog
Lines
^
|
|
  VoIP phones

The telco has a DB9 terminated interface straight to the PBX and I cannot
make sense out of the interface for the PBX. What kind of interface is this?
How do I connect the RJ48 of the PRI cards to make this whole setting work.


Thanks alot.


2011/3/10 Josué Conti 

> Hello Bobola, I'm using a Sangoma card with Siemens HiPath 3750 with ISDN
> Protocol with configurations below:
>
> Asterisk:
> #zaptel.conf
> span=1,1,0,ccs,hdb3,crc4
> bchan=1-15,17-31
> dchan=16
> loadzone=us
> defaultzone=us
>
> #zapata.conf
> [trunkgroups]
>
> [channels]
> language=pt_BR
> context=default
> switchtype=euroisdn
> pridialplan=unknown
> prilocaldialplan=unknown
> facilityenable = yes
> signalling=pri_cpe
> ;rxwink=300
> usecallerid=yes
> hidecallerid=no
> callwaiting=yes
> usecallingpres=yes
> restrictcid=no
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> canpark=yes
> cancallforward=yes
> callreturn=yes
> echocancel=yes
> echocancelwhenbridged=yes
> rxgain=0.0
> txgain=0.0
> group=1
> callgroup=1
> immediate=no
> callerid=asreceived
> musiconhold=default
> group=1
> channel=>1-15
> channel=>17-31
>
> In HiPath 3750 verify the fields:
> Line/Network ---> Trunks ---> Parameters (inside the slot/trunk) --->ISDN
> Flags ---> Protocol Description ---> T1/S2M: Euro-Amt PP (with CRC4) in your
> case, but I don't use CRC4 so my configuration is T1/S2M: Euro-Amt PP
> without CRC, OK?
> I hope this information help you.
>
> Best Regards
>
> Josue
>
> 2011/3/10 Bobola Oke 
>
>> Hello all,
>>
>> I am trying to connect asterisk to a Siemens Hipath 3750 PBX system.
>>
>> I have a physical connection issue. I know that I should use a crossover
>> RJ48 cable to link the two systems. The problem however is that the physical
>> interface of the Siemens system is very unfamiliar. From my digging around,
>> I think that this is an S2M interface.
>> http://www.mail-archive.com/asterisk-users@lists.digium.com/msg86298.html
>>
>>
>> Please any suggestions on how to go about this?
>>
>> Thanks
>>
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>>
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>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
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Re: [asterisk-users] [1.4] Reading phone number the French way?

[Gilles]

On Thu, 10 Mar 2011 08:34:51 -0600, "Danny Nicholas"
 wrote:
>== say.conf
>
>;1-9
>_[n]um:X => digits/${SAY}
>
>;10-99
>_[n]um:1X => digits/${SAY}
>_[n]um:[2-9]0 =>  digits/${SAY}
>_[n]um:[2-6]1 => digits/${SAY:0:1}0, vm-and, digits/${SAY:1}
>_[n]um:71 => digits/60, vm-and, num:1${SAY:1}
>_[n]um:7X => digits/60, num:1${SAY:1}
>_[n]um:9X => digits/80, num:1${SAY:1}
>_[n]um:[2-9][1-9] =>  digits/${SAY:0:1}0, num:${SAY:1}
>
>;100-999
>_[n]um:100 => digits/hundred
>_[n]um:1XX => digits/hundred, num:${SAY:1}
>_[n]um:[2-9]00 => num:${SAY:0:1}, digits/hundred
>_[n]um:[2-9]XX => num:${SAY:0:1}, digits/hundred, num:${SAY:1}
>
>;0800XX -> 0899XX
>;_pho[n]e:08 => num:${SAY:0:1}, num:${SAY:1:3},
>num:${SAY:4:2}, num:${SAY:6:2}, num:${SAY:8:2}
>
>== CLI
>
>-- Executing [@internal:4] Playback("SIP/xlite-02a56004",
>"phone:0810009032|say") in new stack
>--  Playing 'digits/0' (language 'fr')
>--  Playing 'digits/8' (language 'fr')
>--  Playing 'digits/hundred' (language 'fr')
>--  Playing 'digits/10' (language 'fr')
>--  Playing 'digits/0' (language 'fr')
>--  Playing 'digits/0' (language 'fr')
>--  Playing 'digits/90' (language 'fr')
>--  Playing 'digits/30' (language 'fr')
>--  Playing 'digits/2' (language 'fr')
>
>-- Executing [@internal:6] Playback("SIP/xlite-02a56004",
>"phone:0892123456}|say") in new stack
>--  Playing 'digits/0' (language 'fr')
>--  Playing 'digits/8' (language 'fr')
>--  Playing 'digits/9' (language 'fr')
>--  Playing 'digits/2' (language 'fr')
>--  Playing 'digits/1' (language 'fr')
>--  Playing 'digits/2' (language 'fr')
>--  Playing 'digits/3' (language 'fr')
>--  Playing 'digits/4' (language 'fr')
>--  Playing 'digits/5' (language 'fr')
>-- Executing [@internal:7] Hangup("SIP/xlite-02a56004", "") in new
>stack
>== 

>This one is easy.  You have a specific pattern to match 0800.  092 has no
>pattern and therefore defaults back to an "Asterisk Standard Playback".

Sorry, it was a typo: I did mean numbers starting with "0800" or
"0892". Those are the FR equivalent of 1.800 numbers in the US.

Going through the patterns above, it should use the same patterns for
either numbers, but for some reason, fails for 0892 :-/

BTW, is there a way to enable say.conf debugging in the CLI, so I
could check which patterns it goes through to analyze a number?

Thank you.


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Re: [asterisk-users] [1.4] Reading phone number the French way?

[Danny Nicholas]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles
Sent: Thursday, March 10, 2011 8:32 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] [1.4] Reading phone number the French way?

On Thu, 10 Mar 2011 14:37:45 +0100, Gilles 
wrote:
>I figured out how extensions.conf and say.conf work and posted my
>results in the reply to Dave.

Noticed something strange, though: 0800123456 is played OK (ie.
0.800.12.34.56) , but 092123456 is played digit by digit (0.8.9.2,
etc.):

== say.conf

;1-9
_[n]um:X => digits/${SAY}

;10-99
_[n]um:1X => digits/${SAY}
_[n]um:[2-9]0 =>  digits/${SAY}
_[n]um:[2-6]1 => digits/${SAY:0:1}0, vm-and, digits/${SAY:1}
_[n]um:71 => digits/60, vm-and, num:1${SAY:1}
_[n]um:7X => digits/60, num:1${SAY:1}
_[n]um:9X => digits/80, num:1${SAY:1}
_[n]um:[2-9][1-9] =>  digits/${SAY:0:1}0, num:${SAY:1}

;100-999
_[n]um:100 => digits/hundred
_[n]um:1XX => digits/hundred, num:${SAY:1}
_[n]um:[2-9]00 => num:${SAY:0:1}, digits/hundred
_[n]um:[2-9]XX => num:${SAY:0:1}, digits/hundred, num:${SAY:1}

;0800XX -> 0899XX
;_pho[n]e:08 => num:${SAY:0:1}, num:${SAY:1:3},
num:${SAY:4:2}, num:${SAY:6:2}, num:${SAY:8:2}

== CLI

-- Executing [@internal:4] Playback("SIP/xlite-02a56004",
"phone:0810009032|say") in new stack
--  Playing 'digits/0' (language 'fr')
--  Playing 'digits/8' (language 'fr')
--  Playing 'digits/hundred' (language 'fr')
--  Playing 'digits/10' (language 'fr')
--  Playing 'digits/0' (language 'fr')
--  Playing 'digits/0' (language 'fr')
--  Playing 'digits/90' (language 'fr')
--  Playing 'digits/30' (language 'fr')
--  Playing 'digits/2' (language 'fr')

-- Executing [@internal:6] Playback("SIP/xlite-02a56004",
"phone:0892123456}|say") in new stack
--  Playing 'digits/0' (language 'fr')
--  Playing 'digits/8' (language 'fr')
--  Playing 'digits/9' (language 'fr')
--  Playing 'digits/2' (language 'fr')
--  Playing 'digits/1' (language 'fr')
--  Playing 'digits/2' (language 'fr')
--  Playing 'digits/3' (language 'fr')
--  Playing 'digits/4' (language 'fr')
--  Playing 'digits/5' (language 'fr')
-- Executing [@internal:7] Hangup("SIP/xlite-02a56004", "") in new
stack
== 

Can't figure out why it doesn't use the same pattern to play 0800 and
092 numbers. Any idea?

Thank you.

This one is easy.  You have a specific pattern to match 0800.  092 has no
pattern and therefore defaults back to an "Asterisk Standard Playback".


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Re: [asterisk-users] [1.4] Reading phone number the French way?

[Gilles]

On Thu, 10 Mar 2011 14:37:45 +0100, Gilles 
wrote:
>I figured out how extensions.conf and say.conf work and posted my
>results in the reply to Dave.

Noticed something strange, though: 0800123456 is played OK (ie.
0.800.12.34.56) , but 092123456 is played digit by digit (0.8.9.2,
etc.):

== say.conf

;1-9
_[n]um:X => digits/${SAY}

;10-99
_[n]um:1X => digits/${SAY}
_[n]um:[2-9]0 =>  digits/${SAY}
_[n]um:[2-6]1 => digits/${SAY:0:1}0, vm-and, digits/${SAY:1}
_[n]um:71 => digits/60, vm-and, num:1${SAY:1}
_[n]um:7X => digits/60, num:1${SAY:1}
_[n]um:9X => digits/80, num:1${SAY:1}
_[n]um:[2-9][1-9] =>  digits/${SAY:0:1}0, num:${SAY:1}

;100-999
_[n]um:100 => digits/hundred
_[n]um:1XX => digits/hundred, num:${SAY:1}
_[n]um:[2-9]00 => num:${SAY:0:1}, digits/hundred
_[n]um:[2-9]XX => num:${SAY:0:1}, digits/hundred, num:${SAY:1}

;0800XX -> 0899XX
;_pho[n]e:08 => num:${SAY:0:1}, num:${SAY:1:3},
num:${SAY:4:2}, num:${SAY:6:2}, num:${SAY:8:2}

== CLI

-- Executing [@internal:4] Playback("SIP/xlite-02a56004",
"phone:0810009032|say") in new stack
--  Playing 'digits/0' (language 'fr')
--  Playing 'digits/8' (language 'fr')
--  Playing 'digits/hundred' (language 'fr')
--  Playing 'digits/10' (language 'fr')
--  Playing 'digits/0' (language 'fr')
--  Playing 'digits/0' (language 'fr')
--  Playing 'digits/90' (language 'fr')
--  Playing 'digits/30' (language 'fr')
--  Playing 'digits/2' (language 'fr')

-- Executing [@internal:6] Playback("SIP/xlite-02a56004",
"phone:0892123456}|say") in new stack
--  Playing 'digits/0' (language 'fr')
--  Playing 'digits/8' (language 'fr')
--  Playing 'digits/9' (language 'fr')
--  Playing 'digits/2' (language 'fr')
--  Playing 'digits/1' (language 'fr')
--  Playing 'digits/2' (language 'fr')
--  Playing 'digits/3' (language 'fr')
--  Playing 'digits/4' (language 'fr')
--  Playing 'digits/5' (language 'fr')
-- Executing [@internal:7] Hangup("SIP/xlite-02a56004", "") in new
stack
== 

Can't figure out why it doesn't use the same pattern to play 0800 and
092 numbers. Any idea?

Thank you.


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Re: [asterisk-users] One Way Audio

[Satish Patel]

Also it could be the routing issue as well.

--
Sent from my iPhone

On Mar 9, 2011, at 7:43 PM, Duncan Turnbull   
wrote:


So that suggests audio is flowing as follows in a unidirectional  
manner



199.173.66.22.53102 > 74.204.4.5.11732  IP 74.204.4.5.11732 > 
209.216.2.203.60362


Somewhere near the end this pops up which is slightly different, I  
am guessing 74.204.4.5 is your asterisk box


19:18:36.389548 IP 74.204.4.5.11732 > 174.133.195.194.18364: UDP,  
length 172


I am not sure why this is happening or if its still part of the same  
conversation


Overall it looks a bit like the asterisk box thinks it needs to send  
rtp to a different location than perhaps its meant to i.e. its  
asymmetric - you can check the sdp in the sip invite to see where  
media is expected to be sent to


There is no rtp coming back from 209.216.2.203 so possibly this is  
device that isn't meant to be part of the conversation and either  
doesn't exist or is not expecting anything and thus not responding


What are the addresses of the devices in this conversation? so that  
you can match the traffic to device


Cheers Duncan

On 10/03/2011, at 1:20 PM, Tim King wrote:


It looks like this:
19:18:34.782016 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP,  
length 172
19:18:34.789527 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP,  
length 172
19:18:34.802064 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP,  
length 172
19:18:34.809757 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP,  
length 172
19:18:34.821855 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP,  
length 172
19:18:34.829598 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP,  
length 172
19:18:34.842015 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP,  
length 172
19:18:34.849764 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP,  
length 172
19:18:34.861902 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP,  
length 172
19:18:34.869568 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP,  
length 172
19:18:34.881882 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP,  
length 172
19:18:34.889739 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP,  
length 172
19:18:34.901882 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP,  
length 172
19:18:34.909612 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP,  
length 172
19:18:34.921984 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP,  
length 172
19:18:34.929664 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP,  
length 172
19:18:34.941855 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP,  
length 172
19:18:34.949589 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP,  
length 172
19:18:34.962003 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP,  
length 172
19:18:34.969592 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP,  
length 172
19:18:34.981851 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP,  
length 172
19:18:34.989543 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP,  
length 172
19:18:35.002006 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP,  
length 172
19:18:35.009973 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP,  
length 172
19:18:35.022008 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP,  
length 172
19:18:35.029539 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP,  
length 172
19:18:35.042071 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP,  
length 172
19:18:35.049561 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP,  
length 172
19:18:35.062008 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP,  
length 172
19:18:35.069612 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP,  
length 172
19:18:35.081986 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP,  
length 172
19:18:35.089519 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP,  
length 172
19:18:35.101918 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP,  
length 172
19:18:35.109722 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP,  
length 172
19:18:35.122021 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP,  
length 172
19:18:35.129590 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP,  
length 172
19:18:35.141878 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP,  
length 172
19:18:35.149709 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP,  
length 172
19:18:35.161886 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP,  
length 172
19:18:35.169561 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP,  
length 172
19:18:35.181879 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP,  
length 172
19:18:35.189710 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP,  
length 172
19:18:35.201965 IP 199.173.66.22.53103 > 74.204.4.5.11733: UDP,  
length 60
19:18:35.201974 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP,  
length 172
19:18:35.209552 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP,  
length 172
19:18:35.221898 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP,  
length 172
19:18:35.229625 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP,  
length 172
19:18:35.241894 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP,  
length 172
19:18:35.249566 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP,  
length 172
19:18:35.261999 IP 199.17

Re: [asterisk-users] Connecting Asterisk to Siemens Hipath 3750

[Josué Conti]

Hello Bobola, I'm using a Sangoma card with Siemens HiPath 3750 with ISDN
Protocol with configurations below:

Asterisk:
#zaptel.conf
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
loadzone=us
defaultzone=us

#zapata.conf
[trunkgroups]

[channels]
language=pt_BR
context=default
switchtype=euroisdn
pridialplan=unknown
prilocaldialplan=unknown
facilityenable = yes
signalling=pri_cpe
;rxwink=300
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
restrictcid=no
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
immediate=no
callerid=asreceived
musiconhold=default
group=1
channel=>1-15
channel=>17-31

In HiPath 3750 verify the fields:
Line/Network ---> Trunks ---> Parameters (inside the slot/trunk) --->ISDN
Flags ---> Protocol Description ---> T1/S2M: Euro-Amt PP (with CRC4) in your
case, but I don't use CRC4 so my configuration is T1/S2M: Euro-Amt PP
without CRC, OK?
I hope this information help you.

Best Regards

Josue

2011/3/10 Bobola Oke 

> Hello all,
>
> I am trying to connect asterisk to a Siemens Hipath 3750 PBX system.
>
> I have a physical connection issue. I know that I should use a crossover
> RJ48 cable to link the two systems. The problem however is that the physical
> interface of the Siemens system is very unfamiliar. From my digging around,
> I think that this is an S2M interface.
> http://www.mail-archive.com/asterisk-users@lists.digium.com/msg86298.html
>
>
> Please any suggestions on how to go about this?
>
> Thanks
>
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Re: [asterisk-users] [1.4] Reading phone number the French way?

[Gilles]

On Thu, 10 Mar 2011 15:30:51 +0200, Tzafrir Cohen
 wrote:
>I think you're missing SayDigits().
>
>say.conf does use the syntax of the extensions.conf, but it's not a
>dialplan.

Thanks for the input, but SayDigit() isn't right for what I want to
do, since it simply reads a phone number digit-by-digit, which is not
the way phone numbers are read in France.

SayNumber() doesn't work either in this particular case, but is OK for
countries where phone numbers are read digit-by-digit.

I figured out how extensions.conf and say.conf work and posted my
results in the reply to Dave.

Thank you.


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Re: [asterisk-users] [1.4] Reading phone number the French way?

[Tzafrir Cohen]

On Thu, Mar 10, 2011 at 12:55:18PM +0100, Gilles wrote:
> On Tue, 08 Mar 2011 13:22:18 +0100, Gilles 
> wrote:
> >I need to write a script which prompts the callee to type a number,
> >and then read it back to them as confirmation:
> 
> Apparently, the right way to read a phone number back to the user is
> not to use SayNumber() (which might be OK for US-style reading) but
> rather Playback(:,say), which will then rely on
> say.conf

I think you're missing SayDigits().

say.conf does use the syntax of the extensions.conf, but it's not a
dialplan.

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] [1.8] Unable to Register: Registration denied because of contact ACL

[Jeff LaCoursiere]

Thanks. But Like I said,  that's all done. Here's the Endpoint config:

[authentication]
[basic-options](!)                ; a template
        dtmfmode=rfc2833
        context=Phones
        type=friend
        contactdeny=0.0.0.0/0.0.0.0
        contactpermit=172.16.16.0/255.255.255.0
        deny=0.0.0.0/0.0.0.0
        permit=172.16.16.0/24
        host=dynamic
        qualify=no
        insecure=port,invite

[natted-phone](!,basic-options)   ; another template inheriting basic-options
        nat=yes
        directmedia=no

[555](natted-phone)
secret=$$ecret$$
disallow=all
allow=ulaw
allow=gsm

no deal! The irony is that we have a similar configuration at another place, 
but we didn't need to put
anything there and the phones register regardless!

Is this broken



Perhaps the contactdeny is taking precedence in 1.8.  Try it without the 
contactdeny - maybe the existence of a contactpermit will imply a 
contactdeny of "everything else".


Cheers,

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Re: [asterisk-users] [1.8] Unable to Register: Registration denied because of contact ACL

[Jeff LaCoursiere]

On Thu, 10 Mar 2011, Vladimir Mikhelson wrote:


Pay attention, you have permit=172.16.16.0/24 whereas suggestion was 
permit=0.0.0.0/0.0.0.0



Pay attention?  Maybe you should.  He is clearly trying to restrict access 
to the local network, not open it up to the world.


j



On 3/10/2011 1:48 AM, RR wrote:
  On Thu, Mar 10, 2011 at 2:12 AM, Faisal Hanif  wrote:

You can add following line to your peers configuration

 

permit=0.0.0.0/0.0.0.0

 

It will allow to use that peer’s account from any IP



Thanks. But Like I said,  that's all done. Here's the Endpoint config:

[authentication]
[basic-options](!)                ; a template
        dtmfmode=rfc2833
        context=Phones
        type=friend
        contactdeny=0.0.0.0/0.0.0.0
        contactpermit=172.16.16.0/255.255.255.0
        deny=0.0.0.0/0.0.0.0
        permit=172.16.16.0/24
        host=dynamic
        qualify=no
        insecure=port,invite

[natted-phone](!,basic-options)   ; another template inheriting basic-options
        nat=yes
        directmedia=no

[555](natted-phone)
secret=$$ecret$$
disallow=all
allow=ulaw
allow=gsm

no deal! The irony is that we have a similar configuration at another place, 
but we didn't need to put
anything there and the phones register regardless!

Is this broken


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Re: [asterisk-users] One Way Audio

[Tim King]

My message with the configuration attached is awaiting moderator approval. I
will try to paste relevant data here.

*sip.conf*
[general]
context=inbound ;
allowoverlap=no
udpbindaddr=0.0.0.0
tcpenable=no
tcpbindaddr=0.0.0.0
srvlookup=yes
disallow=all
allow=ulaw
allow=alaw
dtmfmode = rfc2833
directmedia=no

[castlewire]
type=user
host=74.204.4.206
context=outb2
dtmfmode=rfc2833
username=castlewire
secret=1234
quallify=yes
canreinvite=no

[equity]
type=friend
host=dynamic
context=outb2
dtmfmode=rfc2833
username=equity
secret=1234
quallify=yes
canreinvite=no

[3000]
type=friend
host=dynamic
nat=yes
context=inbound
dtmfmode=rfc2833
username=3000
secret=1234
quallify=yes
canreinvite=no

[6168182996]
type=friend
host=dynamic
nat=yes
context=outb2
dtmfmode=rfc2833
username=6168182996
secret=1234
quallify=yes
canreinvite=no

[VITELITY]
type=friend
host=64.2.142.93
port=5060
dtmfmode=auto
context=inbound

[QWEST_OUT]
type=friend
host=67.135.79.80
port=5060
dtmfmode=inband

[QWEST8XX_IN]
type=friend
host=67.135.79.199
port=5060
context=qwest800

[DIDX1]
type=peer
host=67.15.128.14
context=inbound
canreinvite=no

[DIDX2]
type=peer
host=67.15.128.18
context=inbound
canreinvite=no

[DIDX3]
type=peer
host=208.44.220.237
context=inbound
canreinvite=no

[DIDX4]
type=peer
host=208.44.220.234
context=inbound
canreinvite=no

[DIDX5]
type=peer
host=209.62.66.242
context=inbound
canreinvite=no

[DIDX6]
type=peer
host=64.246.22.119
context=inbound
canreinvite=no

[DIDX7]
type=peer
host=70.84.58.18
context=inbound
canreinvite=no

[DIDX8]
type=peer
host=174.133.195.194
context=inbound
canreinvite=no

*iax.conf*

[general]
bandwidth=low
disallow=all
allow=ulaw
allow=alaw
jitterbuffer=no
forcejitterbuffer=no
autokill=yes

register=equity_out:1234@74.204.4.166
;register => IAX2/castlewire_trix:1234@74.204.4.206

[CASTLEWIRE]
type=friend
;host=74.204.4.206
host=dynamic
trunk=yes
auth=md5,plaintext,rsa
secret=1234
username=CASTLEWIRE
qualify=yes
context=outb2

[castlewire_trix]
type=friend
;host=74.204.4.206
host=dynamic
trunk=yes
auth=md5,plaintext,rsa
secret=1234
username=castlewire_trix
qualify=yes
context=outb2
requirecalltoken=no

[equity]
type=friend
host=dynamic
context=equity-fix
secret=1234
username=default
channels=10
trunk=yes
timezone=America/Detroit
qualify=yes
requirecalltoken=no

[equity_out]
type=friend
host=dynamic
context=outb2
secret=1234
username=equity_out
channels=10
trunk=yes
timezone=America/Detroit
qualify=yes
requirecalltoken=no

*extensions.conf*

[inbound]
;Equity Logistics
;exten => 6168182400,1,Dial(IAX2/equity/${EXTEN})
;exten => 6168182400,n,Hangup()
;exten => 8182400,1,Dial(IAX2/equity/${EXTEN})
;exten => 8182400,n,Hangup()

exten => 6168182400,1,Dial(SIP/equity/${EXTEN})
exten => 6168182400,n,Hangup()

exten => 6168182996,1,Dial(SIP/${EXTEN})
exten => 6168182996,n,Hangup()
;exten => 6168182996,1,Answer()
;exten => 6168182996,n,Milliwatt()

exten => 3000,1,Dial(SIP/${EXTEN})
exten => 3000,n,Hangup()

;CASTELWIRE NUMBERS
exten => 6168182000,1,Dial(IAX2/castlewire_trix/${EXTEN})
exten => 6168182000,n,Hangup()

;exten => 6168182000,1,Dial(SIP/4403712250@12.194.10.18)
;exten => 6168182000,n,Hangup()


exten => 6168182999,1,Set(portnum=${CALLERID(rdnis)})
exten => 6168182999,n,Set(cutNum=${CUT(portnum|\-|6)})
exten => 6168182999,n,Dial(SIP/${cutNum})
exten => 6168182999,n,Hangup()
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[asterisk-users] console.conf.sample in 1.8.3

[Jerry Geis]

In console.conf.sample it says run the command "console list available" 
CLI command.


It does not seem to be present:
console list available
No such command 'console list available'

These are the only console commands I see:

   console answer Answer an incoming console call
   console autoanswer Sets/displays autoanswer
 console dial Dial an extension on the console
   console hangup Hangup a call on the console
console send text Send text to the remote device
console {mute|unmute} [toggle] Disable/Enable mic input

Is chan_console just a super set of chan_alsa?

In 1.4 Console/Dsp was ALSA, is Console/DSP now chan_console?
In 1.8 what do I use for ALSA?

Thanks,

Jerry



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[asterisk-users] Connecting Asterisk to Siemens Hipath 3750

[Bobola Oke]

Hello all,

I am trying to connect asterisk to a Siemens Hipath 3750 PBX system.

I have a physical connection issue. I know that I should use a crossover
RJ48 cable to link the two systems. The problem however is that the physical
interface of the Siemens system is very unfamiliar. From my digging around,
I think that this is an S2M interface.
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg86298.html


Please any suggestions on how to go about this?

Thanks
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Re: [asterisk-users] One Way Audio

[Steve Davies]

On 10 March 2011 11:17, Ishfaq Malik  wrote:
> Just fixed our problem with
>
> directmedia=no
>
> but this only applies if your extensions are behind a nat
>
> Ish
>

There are several reasons why "directmedia=no" might be the correct
configuration.

1) NAT - probably the most common reason
2) Routing - Sometimes devices cannot route to each other directly
3) ITSP calls. Many SIP providers will not accept a redirect

and I am sure there are many more...

Cheers,
Steve

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Re: [asterisk-users] [1.4] Reading phone number the French way?

[Gilles]

On Thu, 10 Mar 2011 13:18:41 +0100, Dave Cotton
 wrote:
>Look at the GotoIf statement for example

Thanks Dave for the tip, but I found that I needed to change a pattern
that was already in say.conf:

===
[fr](date-base,digit-base) ;BAD _[n]um:0. => num:${SAY:1}
_[n]um:0X => num:${SAY:0:1}, num:${SAY:1:1}
...

;regular phone numbers : landlines and cellphones
;_pho[n]e:0[1-9] => num:${SAY:0:1}, num:${SAY:1:1},
num:${SAY:2:2}, num:${SAY:4:2}, num:${SAY:6:2}, num:${SAY:8:2}
===

If I got it right, the way say.conf works, is that it reads the whole
say.conf to make a list of the different patterns. Then, when reading
a prefix+number, it reads the patterns on the right side and tries to
find if it furthers matches another pattern.

In the example above, "_[n]um:0X" will match "num:${SAY:6:2}", which
will read the two digits as expected, ie. without ignoring a leading
zero.

Thank you.


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Re: [asterisk-users] [1.4] Reading phone number the French way?

[Dave Cotton]

On 10/03/11 12:55, Gilles wrote:

On Tue, 08 Mar 2011 13:22:18 +0100, Gilles
wrote:

I need to write a script which prompts the callee to type a number,
and then read it back to them as confirmation:


Apparently, the right way to read a phone number back to the user is
not to use SayNumber() (which might be OK for US-style reading) but
rather Playback(:,say), which will then rely on
say.conf

For instance:
=== extensions.conf
exten =>  ,1,Set(NBR2CALL=0142928100) ;exten =>
,n,SayNumber(${NBR2CALL}) exten =>
,n,Playback(phone:${NBR2CALL},say)
===

Using this almost works:
=== say.conf
_pho[n]e:0[1-9] =>  num:${SAY:0:1}, num:${SAY:1:1},
num:${SAY:2:2}, num:${SAY:4:2}, num:${SAY:6:2}, num:${SAY:8:2}
===

The remaining problem is when a couple starts with a zero, eg. 01
(should be read "zero one"): In this case, Asterisk ignores the
leading zero and simply pronounces the second digit ("one")

Does someone know of a trick so that the pattern handles couples that
have a leading zero?



Look at the GotoIf statement for example

exten => _9X.,1,GotoIf($["${EXTEN:1:2}" = "00"]?ft)
exten => _9X.,n,GotoIf($["${EXTEN:1:2}" = "08"]?ft)
exten => _9X.,n,GotoIf($["${EXTEN:1:2}" = "06"]?ft)
exten => _9X.,n,GotoIf($["${EXTEN:1:1}" = "3"]?ft)
exten => _9X.,n,GotoIf($["${EXTEN:1:1}" = "1"]?ft)
exten => _9X.,n,GotoIf($["${TMP:0:2}" != "OK"]?ft)
exten => _9X.,n, ...
exten => _9X.,n(ft),  ...

DC

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Re: [asterisk-users] [1.4] Reading phone number the French way?

[Gilles]

On Tue, 08 Mar 2011 13:22:18 +0100, Gilles 
wrote:
>I need to write a script which prompts the callee to type a number,
>and then read it back to them as confirmation:

Apparently, the right way to read a phone number back to the user is
not to use SayNumber() (which might be OK for US-style reading) but
rather Playback(:,say), which will then rely on
say.conf

For instance:
=== extensions.conf
exten => ,1,Set(NBR2CALL=0142928100) ;exten =>
,n,SayNumber(${NBR2CALL}) exten =>
,n,Playback(phone:${NBR2CALL},say)
=== 

Using this almost works:
=== say.conf
_pho[n]e:0[1-9] => num:${SAY:0:1}, num:${SAY:1:1},
num:${SAY:2:2}, num:${SAY:4:2}, num:${SAY:6:2}, num:${SAY:8:2}
=== 

The remaining problem is when a couple starts with a zero, eg. 01
(should be read "zero one"): In this case, Asterisk ignores the
leading zero and simply pronounces the second digit ("one")

Does someone know of a trick so that the pattern handles couples that
have a leading zero?

Thank you.


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Re: [asterisk-users] One Way Audio

[Ishfaq Malik]

Just fixed our problem with

directmedia=no

but this only applies if your extensions are behind a nat

Ish

On Thu, 2011-03-10 at 09:40 +, Ishfaq Malik wrote:
> I've been having a similar (well exactly the same) problem this last
> week and have been bashing my head trying to fix it.
> 
> Just one question, are you using RealTime?
> 
> Ish
> 
> On Wed, 2011-03-09 at 17:40 -0500, Tim King wrote:
> > I am having trouble with no return audio on inbound calls. I have been
> > working on this for 18 hours and even built a fresh server and moved
> > everything over and I am getting the same results. I need someone that
> > can help get this resolved tonight. I can provide access to all
> > machines involved.
> > 
> > Please email me at tim.compnetw...@gmail.com if you can help.
> > --
> > _
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> > New to Asterisk? Join us for a live introductory webinar every Thurs:
> >http://www.asterisk.org/hello
> > 
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> >http://lists.digium.com/mailman/listinfo/asterisk-users
> 

-- 
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Software Developer
PackNet Ltd

Office:   0161 660 3062


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Re: [asterisk-users] Is this true for Asterisk as SBC?

[Faisal Hanif]

Asterisk doesn't have all features of SBC like relay and forward request on
packet level but all depends on your scenario what you need.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL
INDRODIYA
Sent: Thursday, March 10, 2011 3:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Is this true for Asterisk as SBC?

 

Hi All,

I have starting to reading About SBC and found one artical reagding SBC and
they gives a solutions like this.

i want to know is this true in realtime sceanario while we think of an big
implementation and is it possible with cloud computing.

i have found from 
http://www.smartvox.co.uk/products_gateways_explained.htm

Asterisk as a Session Border Controller
Equip the Asterisk server with two ethernet ports, connect one to the
Internet and the other to your internal network; set up the firewall,
configure the dial plans and you've got everything you need for a fully
functional Session Border Controller. 

*   IP phones can register with the SBC either from the internal network
or from the Internet.
*   Use your SBC as an Inbound and/or Outbound proxy to have complete
control over incoming and outbound calls
*   Use it to control access to your IPBX and to overcome the usual
problems associated with interfacing VoIP between your private network and
the Internet
*   Solve one-way audio and other notoriously difficult and annoying NAT
traversal problems while, at the same time, improving your systems security

regards
dhaval

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[asterisk-users] Experience with Phones, Asterisk and other pbx, cloud services, etc

[randulo]

Hi,

There's been a wave of questions on Asterisk lists about phones that
work well with Asterisk, services, etc. This week's VUC is all about
sharing your experience with various equipment and service providers.

The call begins on Friday at around 12 noon EST (9AM PST, 5PM GMT).
Info, recordings: http://voipusersconference.org

IRC: #vuc Freenode.net (or http://vuc.me/irc) - Local times: http://vuc.me/next

SIP:200...@login.zipdx.com using g722 or g711
Skype:vuc.me

Looking forward to your questions and answers...

/r

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[asterisk-users] Is this true for Asterisk as SBC?

[DHAVAL INDRODIYA]

*Hi All,

I have starting to reading About SBC and found one artical reagding SBC and
they gives a solutions like this.

i want to know is this true in realtime sceanario while we think of an big
implementation and is it possible with cloud computing.

i have found from
http://www.smartvox.co.uk/products_gateways_explained.htm

Asterisk as a Session Border Controller*
Equip the Asterisk server with two ethernet ports, connect one to the
Internet and the other to your internal network; set up the firewall,
configure the dial plans and you've got everything you need for a fully
functional Session Border Controller.

   - IP phones can register with the SBC either from the internal network or
   from the Internet.
   - Use your SBC as an Inbound and/or Outbound proxy to have complete
   control over incoming and outbound calls
   - Use it to control access to your IPBX and to overcome the usual
   problems associated with interfacing VoIP between your private network and
   the Internet
   - Solve one-way audio and other notoriously difficult and annoying NAT
   traversal problems while, at the same time, improving your systems security

regards
dhaval
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Re: [asterisk-users] problem with crashing Asterisk 1.8

[Peter den Hartog]

1.8.0 :-), Nothing fancy just simple dialing/trunking.

On Thu, Mar 10, 2011 at 11:31 AM, --[ UxBoD ]--  wrote:

>
> --
>
> My Asterisk 1.8 (with Dahdi/Wanrouter) is crashing every minute or 2. It
> just keeps restarting.
> Any pointers on log files to watch? I tried to debug it but i couldn't find
> a good reason for the crashes.
> Maby the box is just overloaded or something like that but there should be
> a log file telling me that, right?
>
> Thanks,
> Peter
>
> --
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>
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> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
> Which version of 1.8 are you using ? If you are using call pickup that can
> generate a segfault and crash Asterisk in version 1.8.3. Am hoping 1.8.4
> will be out soon.
> --
> Thanks, Phil
>
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Re: [asterisk-users] problem with crashing Asterisk 1.8

[--[ UxBoD ]--]

- Original Message -

> My Asterisk 1.8 (with Dahdi/Wanrouter) is crashing every minute or 2.
> It just keeps restarting.
> Any pointers on log files to watch? I tried to debug it but i
> couldn't find a good reason for the crashes.
> Maby the box is just overloaded or something like that but there
> should be a log file telling me that, right?

> Thanks,
> Peter
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello

> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
Which version of 1.8 are you using ? If you are using call pickup that can 
generate a segfault and crash Asterisk in version 1.8.3. Am hoping 1.8.4 will 
be out soon. 
-- 
Thanks, Phil --
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[asterisk-users] problem with crashing Asterisk 1.8

[Peter den Hartog]

My Asterisk 1.8 (with Dahdi/Wanrouter) is crashing every minute or 2. It
just keeps restarting.
Any pointers on log files to watch? I tried to debug it but i couldn't find
a good reason for the crashes.
Maby the box is just overloaded or something like that but there should be a
log file telling me that, right?

Thanks,
Peter
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Re: [asterisk-users] [1.8] Unable to Register: Registration denied because of contact ACL

[Faisal Hanif]

One more thing check if your SBC is configured in relay mode or forward
mode.  If it is in relay mode you will have original SIP-UA IP in all
requests coming on asterisk and only SBC IP in via but if it is forward mode
you may can have SBC IP all the way in all requests.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vladimir
Mikhelson
Sent: Thursday, March 10, 2011 1:42 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] [1.8] Unable to Register: Registration denied
because of contact ACL

 

Pay attention, you have permit=172.16.16.0/24 whereas suggestion was
permit=0.0.0.0/0.0.0.0



On 3/10/2011 1:48 AM, RR wrote: 

On Thu, Mar 10, 2011 at 2:12 AM, Faisal Hanif  wrote:

You can add following line to your peers configuration

 

permit=0.0.0.0/0.0.0.0

 

It will allow to use that peer's account from any IP

 

 

Thanks. But Like I said,  that's all done. Here's the Endpoint config:

 

[authentication]

[basic-options](!); a template

dtmfmode=rfc2833

context=Phones

type=friend

contactdeny=0.0.0.0/0.0.0.0

contactpermit=172.16.16.0/255.255.255.0

deny=0.0.0.0/0.0.0.0

permit=172.16.16.0/24

host=dynamic

qualify=no

insecure=port,invite

 

[natted-phone](!,basic-options)   ; another template inheriting
basic-options

nat=yes

directmedia=no

 

[555](natted-phone)

secret=$$ecret$$

disallow=all

allow=ulaw

allow=gsm

 

no deal! The irony is that we have a similar configuration at another place,
but we didn't need to put anything there and the phones register regardless!

 

Is this broken

 

 
 
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Re: [asterisk-users] One Way Audio

[Ishfaq Malik]

I've been having a similar (well exactly the same) problem this last
week and have been bashing my head trying to fix it.

Just one question, are you using RealTime?

Ish

On Wed, 2011-03-09 at 17:40 -0500, Tim King wrote:
> I am having trouble with no return audio on inbound calls. I have been
> working on this for 18 hours and even built a fresh server and moved
> everything over and I am getting the same results. I need someone that
> can help get this resolved tonight. I can provide access to all
> machines involved.
> 
> Please email me at tim.compnetw...@gmail.com if you can help.
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
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Software Developer
PackNet Ltd

Office:   0161 660 3062


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Re: [asterisk-users] [1.8] Unable to Register: Registration denied because of contact ACL

[Vladimir Mikhelson]

Pay attention, you have permit=172.16.16.0/24 
whereas suggestion was permit=0.0.0.0/0.0.0.0 



On 3/10/2011 1:48 AM, RR wrote:
> On Thu, Mar 10, 2011 at 2:12 AM, Faisal Hanif  > wrote:
>
> You can add following line to your peers configuration
>
>  
>
> permit=0.0.0.0/0.0.0.0 
>
>  
>
> It will allow to use that peer’s account from any IP
>
>
>
> Thanks. But Like I said,  that's all done. Here's the Endpoint config:
>
> [authentication]
> [basic-options](!); a template
> dtmfmode=rfc2833
> context=Phones
> type=friend
> contactdeny=0.0.0.0/0.0.0.0 
> contactpermit=172.16.16.0/255.255.255.0
> 
> deny=0.0.0.0/0.0.0.0 
> permit=172.16.16.0/24 
> host=dynamic
> qualify=no
> insecure=port,invite
>
> [natted-phone](!,basic-options)   ; another template inheriting
> basic-options
> nat=yes
> directmedia=no
>
> [555](natted-phone)
> secret=$$ecret$$
> disallow=all
> allow=ulaw
> allow=gsm
>
> no deal! The irony is that we have a similar configuration at another
> place, but we didn't need to put anything there and the phones
> register regardless!
>
> Is this broken
>
>
> --
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