[asterisk-users] PSTN Frequency parameters

[Gopalakrishnan N]

Hi Users,

Anybody knows about the frequency parameters and cadence values used in New
Caledonia. The PSTN service provider name is OPT Office of Posts and
Telecommunications.

Regards,
Gopal.
--
_
-- 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

[asterisk-users] File Convert

[Gopalakrishnan N]

Hi users,

I have Asterisk 1.6.2.20 in Ubuntu 10.04. I am trying to convert a gsm file
to G729 using file convert, but I am facing error as follows,

file convert /tmp/welcome.gsm /tmp/welcome.g729
Failed to convert /tmp/welcome.gsm to /tmp/welcome.g729!
Command 'file convert /tmp/welcome.gsm /tmp/welcome.g729' failed.
[Dec 20 17:24:18] WARNING[2221]: translate.c:256 ast_translator_build_path:
No translator path from g723 to alaw
[Dec 20 17:24:18] WARNING[2221]: file.c:184 ast_writestream: Unable to
translate to format g729, source format gsm

Even though I have the module format_g729.so. Do I need to have licensed
G729 codec for this? or codec_g729.so?

Kindly let me know how to convert the file.

Regards
--
_
-- 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

Re: [asterisk-users] File Convert

[Gopalakrishnan N]

Hi Tzafir,

Thanks for your reply.

Yes my actual format I have in wav, I need to convert that to G729, and
more over is there any open G729 license available for single license?

Regards,
Gopal

On Tue, Dec 20, 2011 at 7:15 PM, Tzafrir Cohen wrote:

> On Tue, Dec 20, 2011 at 05:34:46PM +0530, Gopalakrishnan N wrote:
> > Hi users,
> >
> > I have Asterisk 1.6.2.20 in Ubuntu 10.04. I am trying to convert a gsm
> file
> > to G729 using file convert, but I am facing error as follows,
> >
> > file convert /tmp/welcome.gsm /tmp/welcome.g729
> > Failed to convert /tmp/welcome.gsm to /tmp/welcome.g729!
> > Command 'file convert /tmp/welcome.gsm /tmp/welcome.g729' failed.
> > [Dec 20 17:24:18] WARNING[2221]: translate.c:256
> ast_translator_build_path:
> > No translator path from g723 to alaw
> > [Dec 20 17:24:18] WARNING[2221]: file.c:184 ast_writestream: Unable to
> > translate to format g729, source format gsm
> >
> > Even though I have the module format_g729.so. Do I need to have licensed
> > G729 codec for this? or codec_g729.so?
>
> Yes. The g729 codec module requires a per-codec-instance license. In
> your case you use a single codec for encoding the audio to G.729.
>
> BTW: if this is a file you recorded, why convert it from gsm and not
> from a higher-quality format? If this is from the stanard set of
> prompts: any chance it is already available as g729?
>
> --
>   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
>
> --
> _
> -- 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
>
--
_
-- 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

[asterisk-users] Installing Dahdi, libpri of different versions in one pc

[Gopalakrishnan N]

Hi,

I would like to install Dahdi, libpri and Asterisk of different versions in
one machine. Lets say, Asterisk 1.6.X, Asterisk 1.8.x and Asterisk 1.4.x to
be installed in one machine, this can be done using prefix while building
configure.

For dahdi, libpri can it be done in same way? Because I need to test
telephony cards (PRI, BRI, GSM & Transcoding) with different versions of
Asterisk, libpri and Dahdi, I can't remove and install again of each
versions since it is time consuming, sicne there are lot of versions
available.

Any comments would be appreciated.

Thanks.
--
_
-- 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

Re: [asterisk-users] Installing Dahdi, libpri of different versions in one pc

[Gopalakrishnan N]

Thanks for the comments, I hope you are the one developed the script.
Really great.

But still I am not clear with the script file, where I need to start. I
understand that I need to have live.conf file, this file do I need to
create? or it will create automatically also I am able to find libpri,
zaptel and dahdi installation directory specification. So I think first I
need to create live.conf file then to compile others rite?

Please correct me if I am wrong.

On Tue, Mar 13, 2012 at 2:45 PM, Tzafrir Cohen wrote:

> On Tue, Mar 13, 2012 at 01:07:55PM +0530, Gopalakrishnan N wrote:
> > Hi,
> >
> > I would like to install Dahdi, libpri and Asterisk of different versions
> in
> > one machine. Lets say, Asterisk 1.6.X, Asterisk 1.8.x and Asterisk 1.4.x
> to
> > be installed in one machine, this can be done using prefix while building
> > configure.
> >
> > For dahdi, libpri can it be done in same way? Because I need to test
> > telephony cards (PRI, BRI, GSM & Transcoding) with different versions of
> > Asterisk, libpri and Dahdi, I can't remove and install again of each
> > versions since it is time consuming, sicne there are lot of versions
> > available.
>
> Take a look at contrib/scripts/live_ast in the source tree of Asterisk.
> It's intended to run a "private" copy of Asterisk. There's also support
> there for using "private" copies of libpri and DAHDI if you actually
> need that.
>
> Alternatively, build every combination in its own chroot.
>
> In both cases, DAHDI kernel modules you actually load are system-global.
>
> --
>   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
>
> --
> _
> -- 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
>
--
_
-- 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

Re: [asterisk-users] Installing Dahdi, libpri of different versions in one pc

[Gopalakrishnan N]

Its because the card what I have only work with 1.4 and 1.6.

On Wed, Mar 14, 2012 at 4:05 AM, John Novack
wrote:

> **
> Why would you want to even bother testing EOL products, such as 1.4x and
> 1.6.x.x?
>
> Although I am a 1.4 Luddite, I really don't quite understand why you can't
> test with 1.8.x or 10, where you mihgt have a hope of getting something
> fixed if there is a problem, unless you already KNOW there is an issue with
> later versions.
>
> JMO
>
> John Novack
>
>
> Gopalakrishnan N wrote:
>
> Hi,
>
>  I would like to install Dahdi, libpri and Asterisk of different versions
> in one machine. Lets say, Asterisk 1.6.X, Asterisk 1.8.x and Asterisk 1.4.x
> to be installed in one machine, this can be done using prefix while
> building configure.
>
>  For dahdi, libpri can it be done in same way? Because I need to test
> telephony cards (PRI, BRI, GSM & Transcoding) with different versions of
> Asterisk, libpri and Dahdi, I can't remove and install again of each
> versions since it is time consuming, sicne there are lot of versions
> available.
>
>  Any comments would be appreciated.
>
>  Thanks.
>
>
> --
> _
> -- 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
>
>
> --
>
> Dog is my Co-pilot
>
>
--
_
-- 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

Re: [asterisk-users] Installing Dahdi, libpri of different versions in one pc

[Gopalakrishnan N]

I am not sure whether my PRI / BRI card would detect in virtual machine. I
have to check.

On Sun, Mar 18, 2012 at 8:14 AM, virendra bhati  wrote:

> you may installed different version at different virtual machines...
> it will be easy and not time consuming as well.
>
> On Wed, Mar 14, 2012 at 11:22 AM, Gopalakrishnan N <
> gopalakrishnan...@gmail.com> wrote:
>
>> Its because the card what I have only work with 1.4 and 1.6.
>>
>>
>> On Wed, Mar 14, 2012 at 4:05 AM, John Novack <
>> jnov...@stromberg-carlson.org> wrote:
>>
>>> **
>>> Why would you want to even bother testing EOL products, such as 1.4x and
>>> 1.6.x.x?
>>>
>>> Although I am a 1.4 Luddite, I really don't quite understand why you
>>> can't test with 1.8.x or 10, where you mihgt have a hope of getting
>>> something fixed if there is a problem, unless you already KNOW there is an
>>> issue with later versions.
>>>
>>> JMO
>>>
>>> John Novack
>>>
>>>
>>> Gopalakrishnan N wrote:
>>>
>>> Hi,
>>>
>>>  I would like to install Dahdi, libpri and Asterisk of different
>>> versions in one machine. Lets say, Asterisk 1.6.X, Asterisk 1.8.x and
>>> Asterisk 1.4.x to be installed in one machine, this can be done using
>>> prefix while building configure.
>>>
>>>  For dahdi, libpri can it be done in same way? Because I need to test
>>> telephony cards (PRI, BRI, GSM & Transcoding) with different versions of
>>> Asterisk, libpri and Dahdi, I can't remove and install again of each
>>> versions since it is time consuming, sicne there are lot of versions
>>> available.
>>>
>>>  Any comments would be appreciated.
>>>
>>>  Thanks.
>>>
>>>
>>> --
>>> _
>>> -- 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
>>>
>>>
>>> --
>>>
>>> Dog is my Co-pilot
>>>
>>>
>>
>> --
>> _
>> -- 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
>>
>
>
>
> --
>
> Thanks and regards
>
>  Virendra Bhati
> +91-8885268942
> Software Engineer
> E-mail-: virbh...@gmail.com
> Skype id:- virbhati2
> Hyderabad(India)
>
>
> --
> _
> -- 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
>
--
_
-- 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

[asterisk-users] sip pregi net account registration

[Gopalakrishnan N]

Hi guys,

I am trying to configure sip.pregi.net account with my Asterisk 1.4.X,
since its a free account, its not getting registered, even my machine IP is
allowed in firewall. In the same machine if i register openser account
which is in public i am able to register. while checking the sip debug the
register request is keep on sending but there is no response.

what i did is i registered the same account in my softphone installed in my
laptop, there it got registered. only with Asterisk its not registering, I
tried allowing externip as my routers IP, even then its not getting
registered.

Does anyone used sip.pregi.net account with Asterisk? If so let me know the
settings.


Thank you.
--
_
-- 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

[asterisk-users] Nicaragua PSTN Frequency Parameters

[Gopalakrishnan N]

Hi,

Does anybody knows about the PSTN frequency parameter with on/off hook
times for the city Nicaragua. This is part of USA below to Mexico.

Regards.
--
_
-- 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

Re: [asterisk-users] Vitelity Setup

[Gopalakrishnan N]

Hi Alejandro,

I removed the registration and tried as like yours, even inbound calls are
not landing, anyways let me check with vitelity support.

Hi Stephan,
I am not using any SBC. As i said let me check with their support.

Thanks for all the views & comments.

Regards,


On Wed, May 23, 2012 at 10:48 PM, Jeff LaCoursiere  wrote:

>
> On Wed, 2012-05-23 at 13:02 -0400, Alejandro Imass wrote:
> > On Wed, May 23, 2012 at 12:32 PM, Jeff LaCoursiere 
> wrote:
> > > On Wed, 2012-05-23 at 11:00 -0400, Alejandro Imass wrote:
> > >> On Wed, May 23, 2012 at 10:40 AM, Jeff LaCoursiere 
> wrote:
> > >> >
> >
> > [...]
> >
> > > Just wanted to point out that after experiences with dozens of
> > > termination providers, I rate Vitelity pretty low.  We still use them
> > > for US termination, which seems fine and relatively low cost.
> > >
> >
> > Thanks for the detailed input. How do you rate Gafachi? It took us a
> > bit to understand the line model but we plan to use them massively...
> > do you have any experience with Gafachi?
> >
>
> I don't, but looks interesting.  We should probably move this thread to
> the -biz list :)
>
> j
>
>
>
> --
> _
> -- 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
>
--
_
-- 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

Re: [asterisk-users] Vitelity Setup

[Gopalakrishnan N]

yes I did that, even then i am not able to make outbound and inbound as
well.

On Thu, May 24, 2012 at 12:42 PM, Alejandro Imass  wrote:

> On Thu, May 24, 2012 at 2:01 AM, Gopalakrishnan N
>  wrote:
> > Hi Alejandro,
> >
> > I removed the registration and tried as like yours, even inbound calls
> are
> > not landing, anyways let me check with vitelity support.
> >
>
> In the Vitel web app you ust set the routing method to the IP of your
> pbx, maybe that's what's happening I'm pretty sure they check that
> the outbound calls use the same IP.
>
> > Hi Stephan,
> > I am not using any SBC. As i said let me check with their support.
> >
> > Thanks for all the views & comments.
> >
> > Regards,
> >
> >
> > On Wed, May 23, 2012 at 10:48 PM, Jeff LaCoursiere 
> wrote:
> >>
> >>
> >> On Wed, 2012-05-23 at 13:02 -0400, Alejandro Imass wrote:
> >> > On Wed, May 23, 2012 at 12:32 PM, Jeff LaCoursiere 
> >> > wrote:
> >> > > On Wed, 2012-05-23 at 11:00 -0400, Alejandro Imass wrote:
> >> > >> On Wed, May 23, 2012 at 10:40 AM, Jeff LaCoursiere <
> j...@sunfone.com>
> >> > >> wrote:
> >> > >> >
> >> >
> >> > [...]
> >> >
> >> > > Just wanted to point out that after experiences with dozens of
> >> > > termination providers, I rate Vitelity pretty low.  We still use
> them
> >> > > for US termination, which seems fine and relatively low cost.
> >> > >
> >> >
> >> > Thanks for the detailed input. How do you rate Gafachi? It took us a
> >> > bit to understand the line model but we plan to use them massively...
> >> > do you have any experience with Gafachi?
> >> >
> >>
> >> I don't, but looks interesting.  We should probably move this thread to
> >> the -biz list :)
> >>
> >> j
> >>
> >>
> >>
> >> --
> >> _
> >> -- 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
> >
> >
>
--
_
-- 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

Re: [asterisk-users] Vitelity Setup

[Gopalakrishnan N]

Actually I understood that register line is not required, also since my PBX
is behind the pfsense firewall, now what i am going to do is putting the
PBX directly in public network (i.e. without firewall) and will check whats
going to happen.

Hope things would sort out.

Regards.

On Sat, May 26, 2012 at 2:48 AM, Stephen J Alexander
wrote:

> If your server says it is registered, that could be part of the problem.
> Vitelity doesn't use trunk registration, only IP authentication. You should
> not be using a registration string in your trunk definition. I don't know
> if it will hurt but it won't help.
>
> It sounds like you might have only 1 trunk defined, but you need 2; one
> for inbound and one for outbound. Their servers for incoming calls and for
> outgoing calls are separate. If fixing that doesn't do the job, make sure
> that incoming traffic from Vitelity is correctly routed to your PBX (and
> that they have the correct IP to send SIP traffic to).
>
> Regards,
>
> Stephen J Alexander
> MPBX, LLC
> http://mpbx.com
> 832-713-6729
>
>
> On Fri, May 25, 2012 at 4:12 PM, Ralph Green  wrote:
>
>> Howdy,
>>  Since the subject is Viteiy Setup, I don't think this is off topic.
>> My big problem with Vitelity is getting my server to register for
>> incoming calls.  I can make outgoing calls just fine.  My server says
>> it is registered with Vitelity, but no calls come in.  Every attempt
>> to call the number generates an email saying there was a failed call.
>> I am using IAX, not SIP, and that is probably part of the problem.
>> IAX should work better in several ways, but few enough people use it.
>> Vitelity support has been unhelpful so far.  My suspicion is that
>> there is a setting they need to make in their server so that calls go
>> to the registered IAX server, instead of looking for a SIP
>> registration, which is not there.  Has anyone here worked past such a
>> problem?  Was there some special thing I need to ask Vitelity?
>> Thanks,
>> Ralph
>>
>>
>> On 5/24/12, Stephen J Alexander  wrote:
>> > If I were troubleshooting this, the next thing I would do is verify
>> > connectivity on the relevant ports – more plainly, make sure that
>> there's
>> > not a firewall rule with unintended consequences somewhere between your
>> > asterisk and your ISP. Otherwise, as Alejandro suggests – check with
>> > Vitelity support.
>> >
>> > Regards,
>> >
>> > Stephen J Alexander
>> > MPBX, LLC
>> > http://mpbx.com
>> > 832-713-6729
>> >
>> >
>> > On Thu, May 24, 2012 at 9:24 AM, Alejandro Imass  wrote:
>> >
>> >> On Thu, May 24, 2012 at 4:07 AM, Gopalakrishnan N
>> >>  wrote:
>> >> > yes I did that, even then i am not able to make outbound and inbound
>> as
>> >> > well.
>> >> >
>> >> >
>> >>
>> >>
>> >> That's weird. Guess you're gonna have to place a detailed ticket to
>> >> them. It sounds like a network problem to me but without any detailed
>> >> info it's hard to say. Maybe you can try sip set debug in the console
>> >> for the IP and see if you can get an idea of what is happening at the
>> >> packet level.
>> >>
>> >> We use Vitel, Skype SIP (we recently eliminated this one), and now
>> >> Gafachi and they all seem to work per there set-up instructions right
>> >> away.
>> >>
>> >> --
>> >> Alejandro
>> >>
>> >> --
>> >> _
>> >> -- 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
>> >>
>> >
>>
>> --
>> _
>> -- 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
>>
>
>
> --
> _
> -- 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
>
--
_
-- 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

Re: [asterisk-users] Vitelity Setup

[Gopalakrishnan N]

Finally I got it working by removing the pfsense firewall. Something to do
with pfsense firewall.

Regards

On Mon, May 28, 2012 at 2:36 PM, Gopalakrishnan N <
gopalakrishnan...@gmail.com> wrote:

> Actually I understood that register line is not required, also since my
> PBX is behind the pfsense firewall, now what i am going to do is putting
> the PBX directly in public network (i.e. without firewall) and will check
> whats going to happen.
>
> Hope things would sort out.
>
> Regards.
>
>
> On Sat, May 26, 2012 at 2:48 AM, Stephen J Alexander  > wrote:
>
>> If your server says it is registered, that could be part of the problem.
>> Vitelity doesn't use trunk registration, only IP authentication. You should
>> not be using a registration string in your trunk definition. I don't know
>> if it will hurt but it won't help.
>>
>> It sounds like you might have only 1 trunk defined, but you need 2; one
>> for inbound and one for outbound. Their servers for incoming calls and for
>> outgoing calls are separate. If fixing that doesn't do the job, make sure
>> that incoming traffic from Vitelity is correctly routed to your PBX (and
>> that they have the correct IP to send SIP traffic to).
>>
>> Regards,
>>
>> Stephen J Alexander
>> MPBX, LLC
>> http://mpbx.com
>> 832-713-6729
>>
>>
>> On Fri, May 25, 2012 at 4:12 PM, Ralph Green  wrote:
>>
>>> Howdy,
>>>  Since the subject is Viteiy Setup, I don't think this is off topic.
>>> My big problem with Vitelity is getting my server to register for
>>> incoming calls.  I can make outgoing calls just fine.  My server says
>>> it is registered with Vitelity, but no calls come in.  Every attempt
>>> to call the number generates an email saying there was a failed call.
>>> I am using IAX, not SIP, and that is probably part of the problem.
>>> IAX should work better in several ways, but few enough people use it.
>>> Vitelity support has been unhelpful so far.  My suspicion is that
>>> there is a setting they need to make in their server so that calls go
>>> to the registered IAX server, instead of looking for a SIP
>>> registration, which is not there.  Has anyone here worked past such a
>>> problem?  Was there some special thing I need to ask Vitelity?
>>> Thanks,
>>> Ralph
>>>
>>>
>>> On 5/24/12, Stephen J Alexander  wrote:
>>> > If I were troubleshooting this, the next thing I would do is verify
>>> > connectivity on the relevant ports – more plainly, make sure that
>>> there's
>>> > not a firewall rule with unintended consequences somewhere between your
>>> > asterisk and your ISP. Otherwise, as Alejandro suggests – check with
>>> > Vitelity support.
>>> >
>>> > Regards,
>>> >
>>> > Stephen J Alexander
>>> > MPBX, LLC
>>> > http://mpbx.com
>>> > 832-713-6729
>>> >
>>> >
>>> > On Thu, May 24, 2012 at 9:24 AM, Alejandro Imass  wrote:
>>> >
>>> >> On Thu, May 24, 2012 at 4:07 AM, Gopalakrishnan N
>>> >>  wrote:
>>> >> > yes I did that, even then i am not able to make outbound and
>>> inbound as
>>> >> > well.
>>> >> >
>>> >> >
>>> >>
>>> >>
>>> >> That's weird. Guess you're gonna have to place a detailed ticket to
>>> >> them. It sounds like a network problem to me but without any detailed
>>> >> info it's hard to say. Maybe you can try sip set debug in the console
>>> >> for the IP and see if you can get an idea of what is happening at the
>>> >> packet level.
>>> >>
>>> >> We use Vitel, Skype SIP (we recently eliminated this one), and now
>>> >> Gafachi and they all seem to work per there set-up instructions right
>>> >> away.
>>> >>
>>> >> --
>>> >> Alejandro
>>> >>
>>> >> --
>>> >> _
>>> >> -- 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:
>>&g

[asterisk-users] can't get libpri/PRI to work

[Gopalakrishnan N]

Hi,

I am using Red Hat Enterprise Linux 5.5 (32bit) and Asterisk 1.8.12, Dahdi
2.4.1.2 and libpri 1.4.2. The installations is fine. But in the Asterisk
CLI prompt the pri commands are missing, only the pri intense debug span is
populated. Even if i execute that command it results me to "pri set debug 2
span 1 is  not a valid command"

Any assistance would be appreciated.

Regards,
Gopal.
--
_
-- 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

Re: [asterisk-users] can't get libpri/PRI to work

[Gopalakrishnan N]

Finally we made it work by enabling chan_dahdi.so file in asterisk make
menuselect. But the confusion what I have is without doing this, when I
unload and load chan_dahdi.so file I didn'd faced any error. Anyways now it
is working.

Regards,
Gopal.

On Thu, Aug 2, 2012 at 5:58 PM, Gopalakrishnan N <
gopalakrishnan...@gmail.com> wrote:

> Hi,
>
> I am using Red Hat Enterprise Linux 5.5 (32bit) and Asterisk 1.8.12, Dahdi
> 2.4.1.2 and libpri 1.4.2. The installations is fine. But in the Asterisk
> CLI prompt the pri commands are missing, only the pri intense debug span is
> populated. Even if i execute that command it results me to "pri set debug 2
> span 1 is  not a valid command"
>
> Any assistance would be appreciated.
>
> Regards,
> Gopal.
>
--
_
-- 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

[asterisk-users] Asterisk hangs while starting in OpenSuse 12.2

[Gopalakrishnan N]

Hi,

I am using OpenSuse 12.2 64bit OS which uses Kernel 3.3.x version and
downloaded Asterisk 1.8 current version, after installing Asterisk, while
starting Asterisk it hangs at the stage of loading modules.conf, I checked
the forum https://issues.asterisk.org/jira/browse/ASTERISK-19245 but still
after updating through yast also i am facing the issue.

Have anybody faced this type of issue with OpenSuse 12.2, its really wired
working with OpenSuse 12.2, even i tried with OpenSuse 12.1 as well which
results to same failure.


Regards,
Gopal.
--
_
-- 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

Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2

[Gopalakrishnan N]

Hi,

Thanks for your comments. Even I tried with 12.1 also, its the same issue,
I am not sure whether it may be hardware related. Logs below,

OS details - uname -a
Linux laptop-prasad 3.3.0-2-desktop #1 SMP PREEMPT Sat Mar 24 00:11:53 UTC
2012 (7e9dd21) x86_64 x86_64 x86_64 GNU/Linux

while executing asterisk -c from the root prompt, its stuck as below
and the CPU usage is fully utilized,

  == Manager registered action DBPut
  == Manager registered action DBDel
  == Manager registered action DBDelTree
  == Parsing '/etc/asterisk/enum.conf':   == Found
  == Registered application 'CallCompletionRequest'
  == Registered application 'CallCompletionCancel'
  == Parsing '/etc/asterisk/ccss.conf':   == Found
 Asterisk Dynamic Loader Starting:
  == Parsing '/etc/asterisk/modules.conf':   == Found
[Aug 14 10:48:36] NOTICE[3805]: loader.c:1133 load_modules: 184 modules
will be loaded.


Any advice would be much appreciated.

Regards,
Gopal.


On Tue, Aug 14, 2012 at 3:37 AM, Bryant Zimmerman wrote:

> I am running OpenSuse 12.1 with no issues. I have not tried 12.2 beta yet.
>
> Thanks
>
> Bryant Zimmerman (ZK Tech Inc.)
> 616-855-1030 Ext. 2003
>
>
> --
> *From*: "Gopalakrishnan N" 
> *Sent*: Monday, August 13, 2012 8:19 AM
> *To*: "Asterisk Users Mailing List - Non-Commercial Discussion" <
> asterisk-users@lists.digium.com>
> *Subject*: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2
>
>
> Hi,
>
>  I am using OpenSuse 12.2 64bit OS which uses Kernel 3.3.x version and
> downloaded Asterisk 1.8 current version, after installing Asterisk, while
> starting Asterisk it hangs at the stage of loading modules.conf, I checked
> the forum https://issues.asterisk.org/jira/browse/ASTERISK-19245 but
> still after updating through yast also i am facing the issue.
>
>  Have anybody faced this type of issue with OpenSuse 12.2, its really
> wired working with OpenSuse 12.2, even i tried with OpenSuse 12.1 as well
> which results to same failure.
>
>
>  Regards,
> Gopal.
>
>
> --
> _
> -- 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
>
--
_
-- 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

Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2

[Gopalakrishnan N]

If I change autoload=no then asterisk is starting, but post to that loading
modules even chan_sip.so asterisk hangs. Its strange, only in OpenSuse I am
facing this. In CentOS, Ubuntu its working fine, same Asterisk version with
same hardware.

Regards.


On Tue, Aug 14, 2012 at 11:05 AM, Steve Edwards
wrote:

> On Tue, 14 Aug 2012, Gopalakrishnan N wrote:
>
>  while executing asterisk -c from the root prompt, its stuck as below
>> and the CPU usage is fully utilized,
>>
>
> [snip]
>
>
>== Parsing '/etc/asterisk/modules.conf':   == Found
>> [Aug 14 10:48:36] NOTICE[3805]: loader.c:1133 load_modules: 184 modules
>> will be loaded.
>>
>
> I'm just a 1.2 Luddite, but I'll take a stab...
>
> I'm guessing you're autoloading everything. (My personal preference is to
> turn autoloading off and explicitly load just what I need.)
>
> Mung a directory listing of your modules so each module name is prefixed
> with 'noload'
>
> Paste this into you modules.conf.
>
> Comment out the first half of the 'noloads.' If Asterisk still hangs, the
> problem is somewhere in the second half. If not, un-comment the ones you
> just commented and comment out the second half.
>
> Continue this process (bi-section search) until you identify the errant
> module.
>
> You'll have to fiddle a bit as you discover module inter-dependencies.
>
> You could probably make some educated guesses and start with modules that
> touch hardware like dahdi or any timing cruft, but the above process will
> work -- even after a couple of beers.
>
> --
> Thanks in advance,
> --**--**
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
> Newline  Fax: +1-760-731-3000
> --
> _
> -- 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
>
--
_
-- 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

Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2

[Gopalakrishnan N]

Hi Patrick,

Thanks for your suggestion, even though I added my hostname in the
/etc/hosts, still the problem persists. Also I tried to install in OpenSuse
12.2 (32bit) in virtualbox (like vmware) even there I faced problem like
hanging at modules while starting Asterisk.

Regards,
Gopal.


> Please do not top post and properly trim your replies.
>
> Have you made sure that on the OpenSuse box your DNS is configured
> properly? You should be able to lookup your IP address/FQDN both ways. So
> for example 192.168.1.1 (replace with your IP adres) should resolve in
> your.box.com (replace with your FQDN) and vice versa your.box.com should
> resolve into 192.168.1.1. See man dig or man nslookup for commands that can
> do DNS lookups.
>
> Regards,
> Patrick
>
>
>
>
> --
> __**__**_
> -- 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
>
--
_
-- 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

Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2

[Gopalakrishnan N]

Its really weird working with OpenSuse. I am not sure how others are using
with OpenSuse. Through Yast also I tried to install Asterisk package, it
didn't find.

Now I am clueless to work with OpenSuse.



Regards.


On Fri, Aug 17, 2012 at 2:38 PM, Gopalakrishnan N <
gopalakrishnan...@gmail.com> wrote:

> Hi Patrick,
>
> Thanks for your suggestion, even though I added my hostname in the
> /etc/hosts, still the problem persists. Also I tried to install in OpenSuse
> 12.2 (32bit) in virtualbox (like vmware) even there I faced problem like
> hanging at modules while starting Asterisk.
>
> Regards,
> Gopal.
>
>
>
>> Please do not top post and properly trim your replies.
>>
>> Have you made sure that on the OpenSuse box your DNS is configured
>> properly? You should be able to lookup your IP address/FQDN both ways. So
>> for example 192.168.1.1 (replace with your IP adres) should resolve in
>> your.box.com (replace with your FQDN) and vice versa your.box.com should
>> resolve into 192.168.1.1. See man dig or man nslookup for commands that can
>> do DNS lookups.
>>
>> Regards,
>> Patrick
>>
>>
>>
>>
>> --
>> __**__**_
>> -- 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<http://lists.digium.com/mailman/listinfo/asterisk-users>
>>
>
>
--
_
-- 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

Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2

[Gopalakrishnan N]

>From the forum I understand OpenSuse 12.2 is pre-relase and better to use
OpenSuse 12.1. Lets check with OpenSuse 12.1.

Regards.


On Mon, Aug 20, 2012 at 5:34 PM, Gopalakrishnan N <
gopalakrishnan...@gmail.com> wrote:

> Its really weird working with OpenSuse. I am not sure how others are using
> with OpenSuse. Through Yast also I tried to install Asterisk package, it
> didn't find.
>
> Now I am clueless to work with OpenSuse.
>
>
>
> Regards.
>
>
> On Fri, Aug 17, 2012 at 2:38 PM, Gopalakrishnan N <
> gopalakrishnan...@gmail.com> wrote:
>
>> Hi Patrick,
>>
>> Thanks for your suggestion, even though I added my hostname in the
>> /etc/hosts, still the problem persists. Also I tried to install in OpenSuse
>> 12.2 (32bit) in virtualbox (like vmware) even there I faced problem like
>> hanging at modules while starting Asterisk.
>>
>> Regards,
>> Gopal.
>>
>>
>>
>>> Please do not top post and properly trim your replies.
>>>
>>> Have you made sure that on the OpenSuse box your DNS is configured
>>> properly? You should be able to lookup your IP address/FQDN both ways. So
>>> for example 192.168.1.1 (replace with your IP adres) should resolve in
>>> your.box.com (replace with your FQDN) and vice versa your.box.comshould 
>>> resolve into 192.168.1.1. See man dig or man nslookup for commands
>>> that can do DNS lookups.
>>>
>>> Regards,
>>> Patrick
>>>
>>>
>>>
>>>
>>> --
>>> __**__**
>>> _
>>> -- 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<http://lists.digium.com/mailman/listinfo/asterisk-users>
>>>
>>
>>
>
--
_
-- 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

Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2

[Gopalakrishnan N]

Ok Thanks Bryant, let me try with OpenSuse 12.1.

Regards.

On Mon, Aug 20, 2012 at 7:46 PM, Bryant Zimmerman wrote:

> I have the current version of 8.x and 10.x on systems. I am using OpenSuse
> 12.1, We are working on getting a 12.2 boxs up just running out of time.
> Asterisk on all of our boxes are complied from source.
>
> Thanks
>
> Bryant Zimmerman (ZK Tech Inc.)
> 616-855-1030 Ext. 2003
>
>
> ------
> *From*: "Gopalakrishnan N" 
> *Sent*: Monday, August 20, 2012 10:11 AM
> *To*: "Bryant Zimmerman" 
> *Subject*: Re: [asterisk-users] Asterisk hangs while starting in OpenSuse
> 12.2
>
>
> It's really glad that asterisk is installed at your machine in open suse.
> Can you let me know which version you are using and the architecture.
>
> Regards.
> On Aug 20, 2012 6:22 PM, "Bryant Zimmerman"  wrote:
>
>> I compile from source..
>>
>> Sent from my Verizon Wireless Phone
>>
>> - Reply message -
>> From: "Gopalakrishnan N" 
>> Date: Mon, Aug 20, 2012 8:15 am
>> Subject: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2
>> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <
>> asterisk-users@lists.digium.com>
>>
>>  From the forum I understand OpenSuse 12.2 is pre-relase and better to
>> use OpenSuse 12.1. Lets check with OpenSuse 12.1.
>>
>>  Regards.
>>
>>
>> On Mon, Aug 20, 2012 at 5:34 PM, Gopalakrishnan N <
>> gopalakrishnan...@gmail.com> wrote:
>>
>>> Its really weird working with OpenSuse. I am not sure how others are
>>> using with OpenSuse. Through Yast also I tried to install Asterisk package,
>>> it didn't find.
>>>
>>>  Now I am clueless to work with OpenSuse.
>>>
>>>
>>>
>>>  Regards.
>>>
>>>
>>> On Fri, Aug 17, 2012 at 2:38 PM, Gopalakrishnan N <
>>> gopalakrishnan...@gmail.com> wrote:
>>>
>>>> Hi Patrick,
>>>>
>>>>  Thanks for your suggestion, even though I added my hostname in the
>>>> /etc/hosts, still the problem persists. Also I tried to install in OpenSuse
>>>> 12.2 (32bit) in virtualbox (like vmware) even there I faced problem like
>>>> hanging at modules while starting Asterisk.
>>>>
>>>>  Regards,
>>>> Gopal.
>>>>
>>>>
>>>>
>>>>>  Please do not top post and properly trim your replies.
>>>>>
>>>>> Have you made sure that on the OpenSuse box your DNS is configured
>>>>> properly? You should be able to lookup your IP address/FQDN both ways. So
>>>>> for example 192.168.1.1 (replace with your IP adres) should resolve in
>>>>> your.box.com (replace with your FQDN) and vice versa your.box.comshould 
>>>>> resolve into 192.168.1.1. See man dig or man nslookup for commands
>>>>> that can do DNS lookups.
>>>>>
>>>>> Regards,
>>>>> Patrick
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> _
>>>>> -- 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
>>>>>
>>>>
>>>>
>>>
>>
>
--
_
-- 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

Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2

[Gopalakrishnan N]

Hi,

Again I stuck up with OpenSuse 12.1 also, Installed OpenSuse 12.1 (32bit)
version in virtualbox. Downloaded Asterisk 1.8.15. Installed, installation
went fine.

While starting Asterisk, it hangs here,
*Asterisk Dynamic Loader Starting:*
*  == Parsing '/etc/asterisk/modules.conf':   == Found*
*[Aug 23 14:56:14] NOTICE[19340]: loader.c:1133 load_modules: 186 modules
will be loaded.*

any my linux machine uname -a output is below,
*Linux linux-w6le.site 3.1.0-1.2-default #1 SMP Thu Nov 3 14:45:45 UTC 2011
(187dde0) i686 i686 i386 GNU/Linux*
*
*
Any suggestion would be much appreciated.

Regards,
Gopal.

On Tue, Aug 21, 2012 at 11:24 AM, Gopalakrishnan N <
gopalakrishnan...@gmail.com> wrote:

> Ok Thanks Bryant, let me try with OpenSuse 12.1.
>
> Regards.
>
>
> On Mon, Aug 20, 2012 at 7:46 PM, Bryant Zimmerman wrote:
>
>> I have the current version of 8.x and 10.x on systems. I am using
>> OpenSuse 12.1, We are working on getting a 12.2 boxs up just running out of
>> time. Asterisk on all of our boxes are complied from source.
>>
>> Thanks
>>
>> Bryant Zimmerman (ZK Tech Inc.)
>> 616-855-1030 Ext. 2003
>>
>>
>> --
>> *From*: "Gopalakrishnan N" 
>> *Sent*: Monday, August 20, 2012 10:11 AM
>> *To*: "Bryant Zimmerman" 
>> *Subject*: Re: [asterisk-users] Asterisk hangs while starting in
>> OpenSuse 12.2
>>
>>
>> It's really glad that asterisk is installed at your machine in open suse.
>> Can you let me know which version you are using and the architecture.
>>
>> Regards.
>> On Aug 20, 2012 6:22 PM, "Bryant Zimmerman"  wrote:
>>
>>> I compile from source..
>>>
>>> Sent from my Verizon Wireless Phone
>>>
>>> - Reply message -
>>> From: "Gopalakrishnan N" 
>>> Date: Mon, Aug 20, 2012 8:15 am
>>> Subject: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2
>>> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <
>>> asterisk-users@lists.digium.com>
>>>
>>>  From the forum I understand OpenSuse 12.2 is pre-relase and better to
>>> use OpenSuse 12.1. Lets check with OpenSuse 12.1.
>>>
>>>  Regards.
>>>
>>>
>>> On Mon, Aug 20, 2012 at 5:34 PM, Gopalakrishnan N <
>>> gopalakrishnan...@gmail.com> wrote:
>>>
>>>> Its really weird working with OpenSuse. I am not sure how others are
>>>> using with OpenSuse. Through Yast also I tried to install Asterisk package,
>>>> it didn't find.
>>>>
>>>>  Now I am clueless to work with OpenSuse.
>>>>
>>>>
>>>>
>>>>  Regards.
>>>>
>>>>
>>>> On Fri, Aug 17, 2012 at 2:38 PM, Gopalakrishnan N <
>>>> gopalakrishnan...@gmail.com> wrote:
>>>>
>>>>> Hi Patrick,
>>>>>
>>>>>  Thanks for your suggestion, even though I added my hostname in the
>>>>> /etc/hosts, still the problem persists. Also I tried to install in 
>>>>> OpenSuse
>>>>> 12.2 (32bit) in virtualbox (like vmware) even there I faced problem like
>>>>> hanging at modules while starting Asterisk.
>>>>>
>>>>>  Regards,
>>>>> Gopal.
>>>>>
>>>>>
>>>>>
>>>>>>  Please do not top post and properly trim your replies.
>>>>>>
>>>>>> Have you made sure that on the OpenSuse box your DNS is configured
>>>>>> properly? You should be able to lookup your IP address/FQDN both ways. So
>>>>>> for example 192.168.1.1 (replace with your IP adres) should resolve in
>>>>>> your.box.com (replace with your FQDN) and vice versa your.box.comshould 
>>>>>> resolve into 192.168.1.1. See man dig or man nslookup for commands
>>>>>> that can do DNS lookups.
>>>>>>
>>>>>> Regards,
>>>>>> Patrick
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> _
>>>>>> -- 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
>>>>>>
>>>>>
>>>>>
>>>>
>>>
>>
>
--
_
-- 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

Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2

[Gopalakrishnan N]

Hi Bryant,

As you said, I dont have Hyper-V, I avoided virtualbox and tested in normal
host directly, even then it hangs while loading modules.
 *Asterisk Dynamic Loader Starting:*
*  == Parsing '/etc/asterisk/modules.conf':   == Found*
*[Aug 27 11:52:21] NOTICE[22886]: loader.c:1133 load_modules: 186 modules
will be loaded.*

This is really tuff working with OpenSuse. I am clueless how to sort our
this.

Regards.

On Fri, Aug 24, 2012 at 3:55 AM, Hans Witvliet  wrote:

> On Thu, 2012-08-23 at 15:01 +0530, Gopalakrishnan N wrote:
> > Hi,
> >
> >
> > Again I stuck up with OpenSuse 12.1 also, Installed OpenSuse 12.1
> > (32bit) version in virtualbox. Downloaded Asterisk 1.8.15. Installed,
> > installation went fine.
> >
> >
>
> Have you tried the versions from the OBS?
>
> Or perhaps a virtualbox issue? Its notorious for vapourizing
> cpu-cycles...
>
> hw
>
>
>
> --
> _
> -- 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
>
--
_
-- 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

Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2

[Gopalakrishnan N]

Hi Patrick,

With other OS it works like charm. Only with OpenSuse, I am facing this
issue, since I have a situation to stick with OpenSuse, I am struggling in
this.

Regards.

On Mon, Aug 27, 2012 at 4:24 PM, Patrick Lists <
asterisk-l...@puzzled.xs4all.nl> wrote:

> On 27-08-12 08:25, Gopalakrishnan N wrote:
>
>> This is really tuff working with OpenSuse. I am clueless how to sort our
>> this.
>>
>
> Maybe switch to a different distribution? I have used CentOS and RHEL for
> years without any problems and as far as I know both debian and ubuntu
> should work without problems too.
>
> Regards,
> Patrick
>
>
>
>
> --
> __**__**_
> -- 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<http://lists.digium.com/mailman/listinfo/asterisk-users>
>
--
_
-- 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

Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2

[Gopalakrishnan N]

Hi danny,

Are you talking about modules or sip extensions and dahdi extensions
because its a fresh installation also it doesn't have dahdi interface, I am
just trying to have only ip side.

Regards
On Aug 27, 2012 7:27 PM, "Danny Nicholas"  wrote:

> I use OpenSuse here (our production boxes are Suse Enterprise 11 SP1 and
> 10 SP2).  My advice would be to try to start the box with as few SIP/DAHDI
> channels as possible to begin with and add as you get things stable.
>
> ** **
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gopalakrishnan N
> *Sent:* Monday, August 27, 2012 8:52 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Asterisk hangs while starting in OpenSuse
> 12.2
>
> ** **
>
> Hi Patrick,
>
> ** **
>
> With other OS it works like charm. Only with OpenSuse, I am facing this
> issue, since I have a situation to stick with OpenSuse, I am struggling in
> this. 
>
> ** **
>
> Regards. 
>
> On Mon, Aug 27, 2012 at 4:24 PM, Patrick Lists <
> asterisk-l...@puzzled.xs4all.nl> wrote:
>
> On 27-08-12 08:25, Gopalakrishnan N wrote:
>
> This is really tuff working with OpenSuse. I am clueless how to sort our
> this.
>
> ** **
>
> Maybe switch to a different distribution? I have used CentOS and RHEL for
> years without any problems and as far as I know both debian and ubuntu
> should work without problems too.
>
> Regards,
> Patrick
>
>
>
>
>
> --
> _
> -- 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
>
> ** **
>
> --
> _
> -- 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
>
--
_
-- 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

Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2

[Gopalakrishnan N]

I tried that too, what happens is asterisk is loading but after that if I
try to start any one module for example chan_sip.so, asterisk hangs.

Regards.
On Aug 28, 2012 6:44 PM, "Danny Nicholas"  wrote:

> Extensions/trunks.  Another thought is that you might make your
> modules.conf not load anything to start with so you can eliminate a rogue
> module as the problem.  Just change autoload=yes to autoload=no.
>
> ** **
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gopalakrishnan N
> *Sent:* Monday, August 27, 2012 11:47 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Asterisk hangs while starting in OpenSuse
> 12.2
>
> ** **
>
> Hi danny,
>
> Are you talking about modules or sip extensions and dahdi extensions
> because its a fresh installation also it doesn't have dahdi interface, I am
> just trying to have only ip side. 
>
> Regards
>
> On Aug 27, 2012 7:27 PM, "Danny Nicholas"  wrote:
>
> I use OpenSuse here (our production boxes are Suse Enterprise 11 SP1 and
> 10 SP2).  My advice would be to try to start the box with as few SIP/DAHDI
> channels as possible to begin with and add as you get things stable.
>
>  
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gopalakrishnan N
> *Sent:* Monday, August 27, 2012 8:52 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Asterisk hangs while starting in OpenSuse
> 12.2
>
>  
>
> Hi Patrick,
>
>  
>
> With other OS it works like charm. Only with OpenSuse, I am facing this
> issue, since I have a situation to stick with OpenSuse, I am struggling in
> this. ****
>
>  
>
> Regards. 
>
> On Mon, Aug 27, 2012 at 4:24 PM, Patrick Lists <
> asterisk-l...@puzzled.xs4all.nl> wrote:
>
> On 27-08-12 08:25, Gopalakrishnan N wrote:
>
> This is really tuff working with OpenSuse. I am clueless how to sort our
> this.
>
>  
>
> Maybe switch to a different distribution? I have used CentOS and RHEL for
> years without any problems and as far as I know both debian and ubuntu
> should work without problems too.
>
> Regards,
> Patrick
>
>
>
>
>
> --
> _
> -- 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
>
>  
>
>
> --
> _
> -- 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
>
> --
> _
> -- 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
>
--
_
-- 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

Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2

[Gopalakrishnan N]

If I don't need to install dahdi hardware, is it really I need to have
libpri installed?

Regards.
 On Aug 28, 2012 10:26 PM, "Danny Nicholas"  wrote:

> Check Jason Parker’s post from today and see if you skipped any of the
> preliminary build steps.  It is possible that something like libpri is
> biting you.
>
> ** **
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gopalakrishnan N
> *Sent:* Tuesday, August 28, 2012 11:52 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Asterisk hangs while starting in OpenSuse
> 12.2
>
> ** **
>
> I tried that too, what happens is asterisk is loading but after that if I
> try to start any one module for example chan_sip.so, asterisk hangs.
>
> Regards.
>
> On Aug 28, 2012 6:44 PM, "Danny Nicholas"  wrote:
>
> Extensions/trunks.  Another thought is that you might make your
> modules.conf not load anything to start with so you can eliminate a rogue
> module as the problem.  Just change autoload=yes to autoload=no.
>
>  
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gopalakrishnan N
> *Sent:* Monday, August 27, 2012 11:47 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Asterisk hangs while starting in OpenSuse
> 12.2
>
>  
>
> Hi danny,
>
> Are you talking about modules or sip extensions and dahdi extensions
> because its a fresh installation also it doesn't have dahdi interface, I am
> just trying to have only ip side. 
>
> Regards
>
> On Aug 27, 2012 7:27 PM, "Danny Nicholas"  wrote:
>
> I use OpenSuse here (our production boxes are Suse Enterprise 11 SP1 and
> 10 SP2).  My advice would be to try to start the box with as few SIP/DAHDI
> channels as possible to begin with and add as you get things stable.
>
>  
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gopalakrishnan N
> *Sent:* Monday, August 27, 2012 8:52 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Asterisk hangs while starting in OpenSuse
> 12.2
>
>  
>
> Hi Patrick,
>
>  
>
> With other OS it works like charm. Only with OpenSuse, I am facing this
> issue, since I have a situation to stick with OpenSuse, I am struggling in
> this. 
>
>  
>
> Regards. 
>
> On Mon, Aug 27, 2012 at 4:24 PM, Patrick Lists <
> asterisk-l...@puzzled.xs4all.nl> wrote:
>
> On 27-08-12 08:25, Gopalakrishnan N wrote:
>
> This is really tuff working with OpenSuse. I am clueless how to sort our
> this.
>
>  
>
> Maybe switch to a different distribution? I have used CentOS and RHEL for
> years without any problems and as far as I know both debian and ubuntu
> should work without problems too.
>
> Regards,
> Patrick
>
>
>
>
>
> --
> _
> -- 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
>
>  
>
>
> --
> _
> -- 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
>
>
> --
> _
> -- 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
>
> --
> _
> -- 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-user

Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2

[Gopalakrishnan N]

Hi,

I have installed libpri, Dahdi 2.6 and Asterisk 1.8.15, in normal host, I
am not using any virtualbox, still i struck in loading the modules.

Regards.


On Tue, Aug 28, 2012 at 10:47 PM, Bryant Zimmerman wrote:

> I would install both dahdi and libpri. I brought up a 12.2 RC-2 VM on
> hyper-v Windows 8 and followed our standard asterisk build and have no
> issues yet but we have not run full testing to confirm.  Also a point of
> not 12.2 is RC for the next 8 days or so.
>
>
> Thanks
>
> Bryant Zimmerman (ZK Tech Inc.)
> 616-855-1030 Ext. 2003
>
>
> ----------
> *From*: "Gopalakrishnan N" 
> *Sent*: Tuesday, August 28, 2012 1:13 PM
> *To*: "Asterisk Users Mailing List - Non-Commercial Discussion" <
> asterisk-users@lists.digium.com>
>
> *Subject*: Re: [asterisk-users] Asterisk hangs while starting in OpenSuse
> 12.2
>
>
> If I don't need to install dahdi hardware, is it really I need to have
> libpri installed?
>
> Regards.
> On Aug 28, 2012 10:26 PM, "Danny Nicholas"  wrote:
>
>>  Check Jason Parker’s post from today and see if you skipped any of the
>> preliminary build steps.  It is possible that something like libpri is
>> biting you.
>>
>>
>>
>> *From:* asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gopalakrishnan N
>> *Sent:* Tuesday, August 28, 2012 11:52 AM
>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>> *Subject:* Re: [asterisk-users] Asterisk hangs while starting in
>> OpenSuse 12.2
>>
>>
>>
>> I tried that too, what happens is asterisk is loading but after that if I
>> try to start any one module for example chan_sip.so, asterisk hangs.
>>
>> Regards.
>>
>> On Aug 28, 2012 6:44 PM, "Danny Nicholas"  wrote:
>>
>> Extensions/trunks.  Another thought is that you might make your
>> modules.conf not load anything to start with so you can eliminate a rogue
>> module as the problem.  Just change autoload=yes to autoload=no.
>>
>>
>>
>> *From:* asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gopalakrishnan N
>> *Sent:* Monday, August 27, 2012 11:47 PM
>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>> *Subject:* Re: [asterisk-users] Asterisk hangs while starting in
>> OpenSuse 12.2
>>
>>
>>
>> Hi danny,
>>
>> Are you talking about modules or sip extensions and dahdi extensions
>> because its a fresh installation also it doesn't have dahdi interface, I am
>> just trying to have only ip side.
>>
>> Regards
>>
>> On Aug 27, 2012 7:27 PM, "Danny Nicholas"  wrote:
>>
>> I use OpenSuse here (our production boxes are Suse Enterprise 11 SP1 and
>> 10 SP2).  My advice would be to try to start the box with as few SIP/DAHDI
>> channels as possible to begin with and add as you get things stable.
>>
>>
>>
>> *From:* asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gopalakrishnan N
>> *Sent:* Monday, August 27, 2012 8:52 AM
>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>> *Subject:* Re: [asterisk-users] Asterisk hangs while starting in
>> OpenSuse 12.2
>>
>>
>>
>> Hi Patrick,
>>
>>
>>
>> With other OS it works like charm. Only with OpenSuse, I am facing this
>> issue, since I have a situation to stick with OpenSuse, I am struggling in
>> this.
>>
>>
>>
>> Regards.
>>
>> On Mon, Aug 27, 2012 at 4:24 PM, Patrick Lists <
>> asterisk-l...@puzzled.xs4all.nl> wrote:
>>
>> On 27-08-12 08:25, Gopalakrishnan N wrote:
>>
>> This is really tuff working with OpenSuse. I am clueless how to sort our
>> this.
>>
>>
>>
>> Maybe switch to a different distribution? I have used CentOS and RHEL for
>> years without any problems and as far as I know both debian and ubuntu
>> should work without problems too.
>>
>> Regards,
>> Patrick
>>
>>
>>
>>
>> --
>> _
>> -- 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.digiu

Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2

[Gopalakrishnan N]

Hi,

I have started asterisk using strace, and the log is listed in pastebin
http://pastebin.com/ry2Q1e6x

Moreover, for some peoples Asterisk is properly installed in OpenSuse 12.1
(i586), can you please correct me with the installation steps what I did,
my steps as follows,

   1. OpenSuse fresh installation
   2. Login to root in terminal (sudo -i)
   3. Download libpri, dahdi and Asterisk
   4. Install libpri and dahdi (even though I am not using any dahdi
   hardware) - make and make install
   5. Installation of Asterisk (./configure, make menuconfig, make, make
   install and make samples)
   6. Start Asterisk (asterisk -c) - here hangs while loading modules.

any other packages has to be installed or the installation is fine! please
advice!

Regards.


On Thu, Aug 30, 2012 at 7:03 PM, Tzafrir Cohen wrote:

> On Thu, Aug 30, 2012 at 01:42:06PM +0200, Patrick Lists wrote:
> > On 08/30/2012 09:45 AM, Gopalakrishnan N wrote:
> > >Hi,
> > >
> > >I have installed libpri, Dahdi 2.6 and Asterisk 1.8.15, in normal host,
> > >I am not using any virtualbox, still i struck in loading the modules.
> >
> > Please do not top post.
> >
> > Install strace and then start asterisk with the command:
> > # strace asterisk
>
> Asterisk will fork into the background and the process you trace will
> exit.
>
>   strace -f asterisk #?
>   strace asterisk -f #?
>
> Just in case you wonder, 'asterisk -f strace' will not work as you might
> have expected from the above examples. Nither will '-f strace asterisk'.
>
> '-U asterisk ' may also come in handy.
>
> --
>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
>
> --
> _
> -- 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
>
--
_
-- 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

[asterisk-users] SIP Retransmitting REGISTER message

[Gopalakrishnan N]

Hi,

I was trying to register a VoIP trunk in Asterisk , where its keep on
sending Register message to the server, where I am not getting any response
from server.

But whereas if i register in Xlite softphone the account is getting
registered.

I suspect it could be network related issue, but since in softphone it is
getting registered from the same network.

Any ideas to isolate things would be appreciated.

Regards,
Gopal.
--
_
-- 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

Re: [asterisk-users] SIP Retransmitting REGISTER message

[Gopalakrishnan N]

there is no firewall, its just the router gave by the service provider. May
be the SIP port issue?

Regards.

On Wed, Sep 26, 2012 at 6:17 PM, Danny Nicholas  wrote:

> The Asterisk server and softphone are hitting the firewall from two
> different points.  Start there.
>
> ** **
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gopalakrishnan N
> *Sent:* Wednesday, September 26, 2012 7:45 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] SIP Retransmitting REGISTER message
>
> ** **
>
> Hi,
>
> ** **
>
> I was trying to register a VoIP trunk in Asterisk , where its keep on
> sending Register message to the server, where I am not getting any response
> from server. 
>
> ** **
>
> But whereas if i register in Xlite softphone the account is getting
> registered. 
>
> ** **
>
> I suspect it could be network related issue, but since in softphone it is
> getting registered from the same network. 
>
> ** **
>
> Any ideas to isolate things would be appreciated. 
>
> ** **
>
> Regards,
>
> Gopal. 
>
> --
> _
> -- 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
>
--
_
-- 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

Re: [asterisk-users] SIP Retransmitting REGISTER message

[Gopalakrishnan N]

But even then all the IP go via router, so when it goes to service provider
it will go as the same IP address, since its coming from the same network.

Because the softphone and asterisk machine are local network which is
commonly connected to a router.

Regards.

On Wed, Sep 26, 2012 at 6:31 PM, Gopalakrishnan N <
gopalakrishnan...@gmail.com> wrote:

> ahh... ! OK.. I though of this...
>
>
>
> On Wed, Sep 26, 2012 at 6:24 PM, Danny Nicholas  wrote:
>
>> Another possibility – you registered from the softphone first and the
>> provider took the IP address from your PC and “locked out” the IP address
>> of your Asterisk server.
>>
>> ** **
>>
>> *From:* asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gopalakrishnan N
>> *Sent:* Wednesday, September 26, 2012 7:51 AM
>>
>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>> *Subject:* Re: [asterisk-users] SIP Retransmitting REGISTER message
>>
>> ** **
>>
>> there is no firewall, its just the router gave by the service provider.
>> May be the SIP port issue?
>>
>> ** **
>>
>> Regards.
>>
>> ** **
>>
>> On Wed, Sep 26, 2012 at 6:17 PM, Danny Nicholas 
>> wrote:
>>
>> The Asterisk server and softphone are hitting the firewall from two
>> different points.  Start there.
>>
>>  
>>
>> *From:* asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gopalakrishnan N
>> *Sent:* Wednesday, September 26, 2012 7:45 AM
>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>> *Subject:* [asterisk-users] SIP Retransmitting REGISTER message
>>
>>  
>>
>> Hi,
>>
>>  
>>
>> I was trying to register a VoIP trunk in Asterisk , where its keep on
>> sending Register message to the server, where I am not getting any response
>> from server. 
>>
>>  
>>
>> But whereas if i register in Xlite softphone the account is getting
>> registered. 
>>
>>  
>>
>> I suspect it could be network related issue, but since in softphone it is
>> getting registered from the same network. 
>>
>>  
>>
>> Any ideas to isolate things would be appreciated. 
>>
>>  
>>
>> Regards,
>>
>> Gopal. 
>>
>>
>> --
>> _
>> -- 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
>>
>> ** **
>>
>> --
>> _
>> -- 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
>>
>
>
--
_
-- 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

Re: [asterisk-users] SIP Retransmitting REGISTER message

[Gopalakrishnan N]

I have registered in sip.conf and in my network i am not using any port
forwarding kind of stuff (NAT), Asterisk server is directly connected to
Internet and the Internet router doesn't have any firewall.

And attached is asterisk log, that SIP REGISTER messages keep on sending
and no response from the server.

I am sure that this is some network issue, because the same account i
tested in different network (Network B) in some other place and it got
registered, even i am able to make call.

One thing which i don't understand is in same network (Network A) in xlite
phone the account is getting registered and not in Asterisk server.

I just want to isolate things why I am not getting any response, or
somewhere the response is getting lost! :(


Regards,
Gopal.

On Wed, Sep 26, 2012 at 6:32 PM, SamyGo  wrote:

> Hi,
> How are you connected to server ? How have you configured your asterisk
> server to register to other side ? What about any NAT involved in your
> scenario ?Turn on sip debug and share your registrations.
>
> BR
> Sammy
> On Sep 26, 2012 5:54 PM, "Danny Nicholas"  wrote:
>
>> Another possibility – you registered from the softphone first and the
>> provider took the IP address from your PC and “locked out” the IP address
>> of your Asterisk server.
>>
>> ** **
>>
>> *From:* asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gopalakrishnan N
>> *Sent:* Wednesday, September 26, 2012 7:51 AM
>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>> *Subject:* Re: [asterisk-users] SIP Retransmitting REGISTER message
>>
>> ** **
>>
>> there is no firewall, its just the router gave by the service provider.
>> May be the SIP port issue?
>>
>> ** **
>>
>> Regards.
>>
>> ** **
>>
>> On Wed, Sep 26, 2012 at 6:17 PM, Danny Nicholas 
>> wrote:
>>
>> The Asterisk server and softphone are hitting the firewall from two
>> different points.  Start there.
>>
>>  
>>
>> *From:* asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gopalakrishnan N
>> *Sent:* Wednesday, September 26, 2012 7:45 AM
>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>> *Subject:* [asterisk-users] SIP Retransmitting REGISTER message
>>
>>  
>>
>> Hi,
>>
>>  
>>
>> I was trying to register a VoIP trunk in Asterisk , where its keep on
>> sending Register message to the server, where I am not getting any response
>> from server. 
>>
>>  
>>
>> But whereas if i register in Xlite softphone the account is getting
>> registered. 
>>
>>  
>>
>> I suspect it could be network related issue, but since in softphone it is
>> getting registered from the same network. 
>>
>>  
>>
>> Any ideas to isolate things would be appreciated. 
>>
>>  
>>
>> Regards,
>>
>> Gopal. 
>>
>>
>> --
>> _
>> -- 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
>>
>> ** **
>>
>> --
>> _
>> -- 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
>>
>
> --
> _
> -- 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
>
Nov 29 16:10:17 VERBOSE[5249] logger.c: Retransmitting #1 (NAT) to 
202.85.243.105:5060:
REGISTER sip:sip.pennytel.com SIP/2.0
Via: SIP/2.0/UDP 117.223.64.95:5060;branch=z9hG4bK566dab4d;rport
From: ;tag=as107f0d7d

Re: [asterisk-users] SIP Retransmitting REGISTER message

[Gopalakrishnan N]

yes this is the link http://www.callwithus.com/configuration am following,
and using the same, except type=friend i am using type=peer,

[general]
register => username:passw...@sip.callwithus.com

[callwithus]
type=peer
host=sip.callwithus.com
username=username
secret=password
qualify=no
insecure=invite
nat=yes

Also Asterisk server has access to Internet. I can able to ping
sip.callwithus.com.

The same account working in different network.

Regards.


On Thu, Sep 27, 2012 at 12:56 PM, SamyGo  wrote:

>
>
>> I have registered in sip.conf
>
>
> wow that was very detailed. I think I asked How have you configured this
> to register ? I'm pretty much sure you've nat related string mis-configured
> in your sip.conf.
>
> Can you tell if your asterisk server has access to internet !! I can see
> the same situation happening with callwithus register attempts !!
>
> See this page from callwithus and configure your asterisk accordingly for
> both accounts.
>
> http://www.callwithus.com/configuration
>
> BR
> Sammy
>
>
>
> On Thu, Sep 27, 2012 at 12:09 PM, Gopalakrishnan N <
> gopalakrishnan...@gmail.com> wrote:
>
>> I have registered in sip.conf and in my network i am not using any port
>> forwarding kind of stuff (NAT), Asterisk server is directly connected to
>> Internet and the Internet router doesn't have any firewall.
>>
>> And attached is asterisk log, that SIP REGISTER messages keep on sending
>> and no response from the server.
>>
>> I am sure that this is some network issue, because the same account i
>> tested in different network (Network B) in some other place and it got
>> registered, even i am able to make call.
>>
>> One thing which i don't understand is in same network (Network A) in
>> xlite phone the account is getting registered and not in Asterisk server.
>>
>> I just want to isolate things why I am not getting any response, or
>> somewhere the response is getting lost! :(
>>
>>
>> Regards,
>> Gopal.
>>
>>
>> On Wed, Sep 26, 2012 at 6:32 PM, SamyGo  wrote:
>>
>>> Hi,
>>> How are you connected to server ? How have you configured your asterisk
>>> server to register to other side ? What about any NAT involved in your
>>> scenario ?Turn on sip debug and share your registrations.
>>>
>>> BR
>>> Sammy
>>> On Sep 26, 2012 5:54 PM, "Danny Nicholas"  wrote:
>>>
>>>> Another possibility – you registered from the softphone first and the
>>>> provider took the IP address from your PC and “locked out” the IP address
>>>> of your Asterisk server.
>>>>
>>>> ** **
>>>>
>>>> *From:* asterisk-users-boun...@lists.digium.com [mailto:
>>>> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gopalakrishnan
>>>> N
>>>> *Sent:* Wednesday, September 26, 2012 7:51 AM
>>>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>>>> *Subject:* Re: [asterisk-users] SIP Retransmitting REGISTER message
>>>>
>>>> ** **
>>>>
>>>> there is no firewall, its just the router gave by the service provider.
>>>> May be the SIP port issue?
>>>>
>>>> ** **
>>>>
>>>> Regards.
>>>>
>>>> ** **
>>>>
>>>> On Wed, Sep 26, 2012 at 6:17 PM, Danny Nicholas 
>>>> wrote:
>>>>
>>>> The Asterisk server and softphone are hitting the firewall from two
>>>> different points.  Start there.
>>>>
>>>>  
>>>>
>>>> *From:* asterisk-users-boun...@lists.digium.com [mailto:
>>>> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gopalakrishnan
>>>> N
>>>> *Sent:* Wednesday, September 26, 2012 7:45 AM
>>>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>>>> *Subject:* [asterisk-users] SIP Retransmitting REGISTER message
>>>>
>>>>  
>>>>
>>>> Hi,
>>>>
>>>>  
>>>>
>>>> I was trying to register a VoIP trunk in Asterisk , where its keep on
>>>> sending Register message to the server, where I am not getting any response
>>>> from server. 
>>>>
>>>>  
>>>>
>>>> But whereas if i register in Xlite softphone the account is getting
>>>> registered. 
>>>>
>>>>  
>>>>
>&g

[asterisk-users] Asterisk with R2D configuration

[Gopalakrishnan N]

Hi,

Has anybody worked on R2D Brazillian setup. I have configured R2 using
OpenR2 with Asterisk.

While doing some analysis I found R2D is already included in libopenr2.

Have anyone tested the same.

Regards,
Gopal.
--
_
-- 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

[asterisk-users] VoIP Incoming Issue

[Gopalakrishnan N]

I have made the SIP bind port to 5070, and already I have one VoIP trunk
configured in my Asterisk 1.6.

Now the problem is after changing the bind port at some point of time, am
not able to dial in the DID number of the VoIP trunk!

Changing the bind port matters for this?

Regards.
--
_
-- 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

Re: [asterisk-users] debug strategy for one-way audio calls

[Gopalakrishnan N]

@Marrie For one way audio as a debug strategy you can enable RTP debug and
see whether you have both way packets flow SENT and GOT.

Regards


On Thu, May 2, 2013 at 6:05 PM, Johan Wilfer  wrote:

> 2013-05-02 13:19, Marie Fischer skrev:
> > Hello everybody,
> >
> > from time to time, we get so-called simplex / one-way audio calls, where
> one party cannot hear the other. The only thing in common is that is does
> happen with calls via SIP trunk, not ISDN and not internal calls. Nothing
> strange in verbose and SIP logs. Could even be some weird intermittent
> firewall issue I guess.
> >
> > Apart from logging all traffic 24/7 via tcpdump (not really convenient),
> can you give me some ideas how to debug this kind of issue?
> >
> > Asterisk 1.8.11-cert10 on CentOS 6.3, if that matters.
> >
>
> Voipmonitor.org is great for debugging voip. You can either use only the
> sniffer (opensource) and use mysql + the pcap files or you can also buy
> the commercial webgui. Either way, it's a great product.
>
> /Johan
>
> --
> _
> -- 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
>
--
_
-- 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

[asterisk-users] GotoIf function

[Gopalakrishnan N]

Hi,

Actually i would like to get the input from the user and he should not try
more than 3 times, he can try more than 3 times, if yes it will get routed
to the next priority and if not it goes to the loopback again from the
beginning.

And following is the one I created, I just want to know whether this will
validate the input and will allow for 3 times

exten => s,1,GotoIfTime(08:00-09:00,mon-fri,*,*?2:avgtech,1)
exten => s,n,Background(voicemessage_1)
exten => s,n(voicemessage2),Background(voicemessage_2)

exten => s,n(begin),Set(wait=2)
exten => s,n,Set(gottries=0)
exten => s,n,Read(get,"silence/1"${wait})

exten => s,n(gotnothing),Set(gottries=$[${gottries}+1]
exten => s,n,GotoIf($[${LEN(${get})} == 0]?reallynothing:gotdigit)
exten => s,n(reallynothing),GotoIf($[${gottries}>3]?done:voicemessage5)
exten => s,n(done),Background(voicemessage3)
exten => s,n,Background(voicemessage4)
exten => s,n,Playback(moh)
exten => s,n, ; Addittional messageing
exten => s,n,Queue(general technical team)

exten => s,n(voicemessage5),Goto(voicemessage2)

exten => s,n(gotdigit),Set(got=${get})
exten => s,n,GotoIf( $[ "${got}" = "1"]?doneinstall)
exten => s,n(doneinstall),Background(voicemessage3)
exten => s,n,Background(voicemessage4)
exten => s,n,Playback(moh)
exten => s,n, ; Addittional messageing
exten => s,n,Queue(installation technical skill)

exten => s,n,GotoIf( $[ "${got}" = "2"]?done2)
exten => s,n(done2),Background(voicemessage6)
exten => s,n,Goto(begin2)
exten => s,n(begin2),Set(wait=2)
exten => s,n,Set(gottries=0)
exten => s,n,Read(get,"silence/1"${wait})
exten => s,n(gotnothing),Set(gottries=$[${gottries}+1]
exten => s,n,GotoIf($[${LEN(${get})} == 0]?reallynothing:gotdigit2)
exten => s,n(reallynothing),GotoIf($[${gottries}>3]?done:option2)
exten => s,n(done),Background(voicemessage3)
exten => s,n,Background(voicemessage4)
exten => s,n,Playback(moh)
exten => s,n, ; Addittional messageing
exten => s,n,Queue(general technical skill)

exten => s,n(option2),Background(voicemessage5)
exten => s,n,Goto(done2)

and so on... for digit 3...

Thanks in advance...

Regards.
--
_
-- 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

Re: [asterisk-users] GotoIf function

[Gopalakrishnan N]

I just want to make some increment... to 3 and yes go to the desired option
not to one more option.




On Thu, May 23, 2013 at 7:19 PM, Gopalakrishnan N <
gopalakrishnan...@gmail.com> wrote:

> Hi,
>
> Actually i would like to get the input from the user and he should not try
> more than 3 times, he can try more than 3 times, if yes it will get routed
> to the next priority and if not it goes to the loopback again from the
> beginning.
>
> And following is the one I created, I just want to know whether this will
> validate the input and will allow for 3 times
>
> exten => s,1,GotoIfTime(08:00-09:00,mon-fri,*,*?2:avgtech,1)
> exten => s,n,Background(voicemessage_1)
> exten => s,n(voicemessage2),Background(voicemessage_2)
>
> exten => s,n(begin),Set(wait=2)
> exten => s,n,Set(gottries=0)
> exten => s,n,Read(get,"silence/1"${wait})
>
> exten => s,n(gotnothing),Set(gottries=$[${gottries}+1]
> exten => s,n,GotoIf($[${LEN(${get})} == 0]?reallynothing:gotdigit)
> exten => s,n(reallynothing),GotoIf($[${gottries}>3]?done:voicemessage5)
> exten => s,n(done),Background(voicemessage3)
> exten => s,n,Background(voicemessage4)
> exten => s,n,Playback(moh)
> exten => s,n, ; Addittional messageing
> exten => s,n,Queue(general technical team)
>
> exten => s,n(voicemessage5),Goto(voicemessage2)
>
> exten => s,n(gotdigit),Set(got=${get})
> exten => s,n,GotoIf( $[ "${got}" = "1"]?doneinstall)
> exten => s,n(doneinstall),Background(voicemessage3)
> exten => s,n,Background(voicemessage4)
> exten => s,n,Playback(moh)
> exten => s,n, ; Addittional messageing
> exten => s,n,Queue(installation technical skill)
>
> exten => s,n,GotoIf( $[ "${got}" = "2"]?done2)
>  exten => s,n(done2),Background(voicemessage6)
> exten => s,n,Goto(begin2)
> exten => s,n(begin2),Set(wait=2)
> exten => s,n,Set(gottries=0)
> exten => s,n,Read(get,"silence/1"${wait})
> exten => s,n(gotnothing),Set(gottries=$[${gottries}+1]
> exten => s,n,GotoIf($[${LEN(${get})} == 0]?reallynothing:gotdigit2)
> exten => s,n(reallynothing),GotoIf($[${gottries}>3]?done:option2)
> exten => s,n(done),Background(voicemessage3)
> exten => s,n,Background(voicemessage4)
> exten => s,n,Playback(moh)
> exten => s,n, ; Addittional messageing
> exten => s,n,Queue(general technical skill)
>
> exten => s,n(option2),Background(voicemessage5)
> exten => s,n,Goto(done2)
>
> and so on... for digit 3...
>
> Thanks in advance...
>
> Regards.
>
--
_
-- 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

Re: [asterisk-users] Error 488 Not Acceptable Here

[Gopalakrishnan N]

488 not acceptable is due to codec error. Make sure you have right codec in
place between the end points.


On Fri, May 24, 2013 at 12:18 AM, Maximilian Grobecker <
m.grobec...@portunity.de> wrote:

> Hi,
>
> Maybe you have not allowed T.38 as acceptable codec ;-)
> You can try with "allow=all" in your sip.conf.
>
>
> Am 22.05.2013 16:39, schrieb Andrew Colin:
> > Hi guys,
> >
> > Any idea why I am getting this error when someone tries to send me a T38
> > Fax?
> >
> > --
> > _
> > -- 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
>
>
> --
>
>
> --
> - Portunity GmbH - Werner-Seelenbinder-Str. 23
> -- 42477 Radevormwald - Germany
> -
> - Portal:  http://www.portunity.de
> -
> - General: Phone: +49 (0)202 - 69555 - 0
> -  eMail/SIP: i...@portunity.de
> -  Fax:   +49 (0)202 - 69555 - 190
> -
> - Support: Phone: +49 (0)202 - 69555 - 300
> -  eMail/SIP: supp...@portunity.de
> -
> - Amtsgericht Koeln HRB 38162
> - USt-Identnummer DE206277867
> - Geschaeftsfuehrung: Bjoern Ruecker, Bernd Schnell
> --
>
> --
> _
> -- 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
>
--
_
-- 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

[asterisk-users] Asterisk 11 dtmf not recognised

[Gopalakrishnan N]

Hi

I have a dialplan as per the following,

extensions.conf
[avgtest]
exten = 100,n,Playback(avgtest/message1)
exten = 100,n,Set(rightPIN=1)
exten = 100,n,Read(inPIN,,1,,5,3) ; Attempts for 5 times with 3 seconds of
timeout
exten = 100,n,GotoIf($["${inPIN}" = "${rightPIN}"]?pin-accepted,1)
exten = 100,n,Hangup() ; Didn't go to pin-accepted, so play badPIN and
hangup
exten=pinaccepted,1,Playback(avgtest/message2) ; correct pin, play

sipconf
[1001]
uername=1001
secret=1001
context=avgtest
disallow=all
allow=ulaw
allow=alaw
dtmfmode=auto
type=friend
host=dynamic
canreinvite=yes
relaxdtmf=yes

This looks very simple but dtmf is not recognised.

Am using asterisk 11.

Any suggestions is much appreciated.

Regards
--
_
-- 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

Re: [asterisk-users] Asterisk 11 dtmf not recognised

[Gopalakrishnan N]

Tried info, rfc2833, inband and finally kept as auto.
On 25 May 2013 02:20, "Doug Lytle"  wrote:

> >> dtmfmode=auto
>
> dtmfmode=info
>
> or
>
> dtmfmode=rfc2833
>
> Doug
>
>
> --
> Ben Franklin quote:
>
> "Those who would give up Essential Liberty to purchase a little Temporary
> Safety, deserve neither Liberty nor Safety."
>
> --
> _
> -- 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
>
--
_
-- 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

Re: [asterisk-users] Asterisk 11 dtmf not recognised

[Gopalakrishnan N]

With Asterisk 1.8 I got it working.

Regards


On Sat, May 25, 2013 at 2:37 AM, Gopalakrishnan N <
gopalakrishnan...@gmail.com> wrote:

> Tried info, rfc2833, inband and finally kept as auto.
> On 25 May 2013 02:20, "Doug Lytle"  wrote:
>
>>  >> dtmfmode=auto
>>
>> dtmfmode=info
>>
>> or
>>
>> dtmfmode=rfc2833
>>
>> Doug
>>
>>
>> --
>> Ben Franklin quote:
>>
>> "Those who would give up Essential Liberty to purchase a little Temporary
>> Safety, deserve neither Liberty nor Safety."
>>
>> --
>> _
>> -- 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
>>
>
--
_
-- 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

Re: [asterisk-users] Asterisk 11 dtmf not recognised

[Gopalakrishnan N]

Am using Read application to get the digit, since its recognizing... I
would like to get for 3 attempts and then after 3rd attempt it has to
playback some different message like entries exceeded.

My dialplan as,
exten = 100,1(begin),Playback(letters/a)
exten = 100,n,Set(rightPIN=1)
exten = 100,n,Read(inPIN,,1,skip,3,3) ; Attempts for 5 times with 3 seconds
of timeout
exten = 100,n,GotoIf($["${inPIN}" = "${rightPIN}"]?pin-accepted,1)
exten = 100,n,Playback(letters/c) ; Didn't go to pin-accepted, so play
badPIN and hangup
exten = pin-accepted,1,Playback(letters/b) ; correct pin, play


what happens its keep on asking to enter digit If my DTMF didnt match. Do i
need to use any return function... ?

Actually my goal is to ask for 3 times and if not matched then return to
some other application.

Thanks in advance.


On Sat, May 25, 2013 at 3:19 PM, Gopalakrishnan N <
gopalakrishnan...@gmail.com> wrote:

> With Asterisk 1.8 I got it working.
>
> Regards
>
>
> On Sat, May 25, 2013 at 2:37 AM, Gopalakrishnan N <
> gopalakrishnan...@gmail.com> wrote:
>
>> Tried info, rfc2833, inband and finally kept as auto.
>> On 25 May 2013 02:20, "Doug Lytle"  wrote:
>>
>>>  >> dtmfmode=auto
>>>
>>> dtmfmode=info
>>>
>>> or
>>>
>>> dtmfmode=rfc2833
>>>
>>> Doug
>>>
>>>
>>> --
>>> Ben Franklin quote:
>>>
>>> "Those who would give up Essential Liberty to purchase a little
>>> Temporary Safety, deserve neither Liberty nor Safety."
>>>
>>> --
>>> _
>>> -- 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
>>>
>>
>
--
_
-- 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

Re: [asterisk-users] Asterisk 11 dtmf not recognised

[Gopalakrishnan N]

Finally got it working with 3 attempts by the fialplan,

exten = 300,1,Playback(letters/a)
exten = 300,n,Set(gottries=0)
exten = 300,n(getmore),Set(rightPIN=1)
exten = 300,n,Read(inPIN,,1,skip,3,3) ; Attempts for 3 times with 3 seconds
of timeout
exten = 300,n(gotdigit),GotoIf($["${inPIN}" = "${rightPIN}"]?pin-accepted,1)
exten = 300,n,Set(gottries=$[${gottries}+1];
exten = 300,n,GotoIf($[${LEN(${inPIN})} == 0]?reallynothing:gotdigit)
exten = 300,n(reallynothing),GotoIf($[${gottries}>3]?done:getmore) ;
Attempts for 3 tries if greater than 3 then it will come out or else
getmore will called
exten = 300,n(done),Playback(letters/c) ; Didn't go to pin-accepted, so
play badPIN and hangup
exten = pin-accepted,1,Playback(letters/b) ; correct pin, play

Thanks
 On 25 May 2013 15:38, "Gopalakrishnan N" 
wrote:

> Am using Read application to get the digit, since its recognizing... I
> would like to get for 3 attempts and then after 3rd attempt it has to
> playback some different message like entries exceeded.
>
> My dialplan as,
> exten = 100,1(begin),Playback(letters/a)
> exten = 100,n,Set(rightPIN=1)
> exten = 100,n,Read(inPIN,,1,skip,3,3) ; Attempts for 5 times with 3
> seconds of timeout
> exten = 100,n,GotoIf($["${inPIN}" = "${rightPIN}"]?pin-accepted,1)
> exten = 100,n,Playback(letters/c) ; Didn't go to pin-accepted, so play
> badPIN and hangup
> exten = pin-accepted,1,Playback(letters/b) ; correct pin, play
>
>
> what happens its keep on asking to enter digit If my DTMF didnt match. Do
> i need to use any return function... ?
>
> Actually my goal is to ask for 3 times and if not matched then return to
> some other application.
>
> Thanks in advance.
>
>
> On Sat, May 25, 2013 at 3:19 PM, Gopalakrishnan N <
> gopalakrishnan...@gmail.com> wrote:
>
>> With Asterisk 1.8 I got it working.
>>
>> Regards
>>
>>
>> On Sat, May 25, 2013 at 2:37 AM, Gopalakrishnan N <
>> gopalakrishnan...@gmail.com> wrote:
>>
>>> Tried info, rfc2833, inband and finally kept as auto.
>>> On 25 May 2013 02:20, "Doug Lytle"  wrote:
>>>
>>>>  >> dtmfmode=auto
>>>>
>>>> dtmfmode=info
>>>>
>>>> or
>>>>
>>>> dtmfmode=rfc2833
>>>>
>>>> Doug
>>>>
>>>>
>>>> --
>>>> Ben Franklin quote:
>>>>
>>>> "Those who would give up Essential Liberty to purchase a little
>>>> Temporary Safety, deserve neither Liberty nor Safety."
>>>>
>>>> --
>>>> _
>>>> -- 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
>>>>
>>>
>>
>
--
_
-- 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

[asterisk-users] DTMF recognized after call establishment

[Gopalakrishnan N]

Hi,

I am receiving DTMF without any reason after call establishment.

The log as follows, and I suspect something related to directmedia,
[May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49 is
making progress passing it to SIP/MAN-000a4b48
[May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49
answered SIP/MAN-000a4b48
[May 17 00:33:35] DTMF[4238] channel.c: DTMF end '*' received on
SIP/MyTrunk-000a4b49, duration 0 ms
[May 17 00:33:35] DTMF[4238] channel.c: DTMF end accepted without begin '*'
on SIP/MyTrunk-000a4b49
[May 17 00:33:35] DTMF[4238] channel.c: DTMF end passthrough '*' on
SIP/MyTrunk-000a4b49
[May 17 00:33:36] DTMF[4238] channel.c: DTMF end '8' received on
SIP/MyTrunk-000a4b49, duration 0 ms
[May 17 00:33:36] DTMF[4238] channel.c: DTMF end accepted without begin '8'
on SIP/MyTrunk-000a4b49
[May 17 00:33:36] DTMF[4238] channel.c: DTMF end passthrough '8' on
SIP/MyTrunk-000a4b49
[May 17 00:33:36] DTMF[4104] channel.c: DTMF end '8' received on
SIP/MAN-000a4af0, duration 100 ms
[May 17 00:33:36] DTMF[4104] channel.c: DTMF begin emulation of '8' with
duration 100 queued on SIP/MAN-000a4af0
[May 17 00:33:36] DTMF[4104] channel.c: DTMF end emulation of '8' queued on
SIP/MAN-000a4af0
[May 17 00:33:37] DTMF[4234] channel.c: DTMF end '1' received on
SIP/MAN-000a4b41, duration 100 ms
[May 17 00:33:37] DTMF[4234] channel.c: DTMF begin emulation of '1' with
duration 100 queued on SIP/MAN-000a4b41
[May 17 00:33:37] DTMF[4234] channel.c: DTMF end emulation of '1' queued on
SIP/MAN-000a4b41
[May 17 00:33:55] VERBOSE[4106] pbx.c:   == Spawn extension
(sip-trunk-inbound, 2127773456, 1) exited non-zero on 'SIP/MyTrunk-000a4af3'
[May 17 00:33:56] VERBOSE[4136] pbx.c: -- Executing [h@trunk-outbound:1]
NoOp("SIP/MAN-000a4b09", "16") in new stack
[May 17 00:33:56] VERBOSE[4136] pbx.c:   == Spawn extension
(trunk-outbound, 87457712, 2) exited non-zero on 'SIP/MAN-000a4b09'

Is this some thing related to SIP RE-INVITE?

Thanks.
--
_
-- 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

Re: [asterisk-users] DTMF recognized after call establishment

[Gopalakrishnan N]

So any resolution for this?

I suspect it could be related to RE INVITE


On Tue, May 28, 2013 at 2:09 PM, Asghar Mohammad wrote:

> i had this in past there was an ATA configured to send 9 at the end of
> dialing in my case.
>
>
> On Tue, May 28, 2013 at 8:21 AM, Gopalakrishnan N <
> gopalakrishnan...@gmail.com> wrote:
>
>> Hi,
>>
>> I am receiving DTMF without any reason after call establishment.
>>
>> The log as follows, and I suspect something related to directmedia,
>> [May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49
>> is making progress passing it to SIP/MAN-000a4b48
>> [May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49
>> answered SIP/MAN-000a4b48
>> [May 17 00:33:35] DTMF[4238] channel.c: DTMF end '*' received on
>> SIP/MyTrunk-000a4b49, duration 0 ms
>> [May 17 00:33:35] DTMF[4238] channel.c: DTMF end accepted without begin
>> '*' on SIP/MyTrunk-000a4b49
>> [May 17 00:33:35] DTMF[4238] channel.c: DTMF end passthrough '*' on
>> SIP/MyTrunk-000a4b49
>> [May 17 00:33:36] DTMF[4238] channel.c: DTMF end '8' received on
>> SIP/MyTrunk-000a4b49, duration 0 ms
>> [May 17 00:33:36] DTMF[4238] channel.c: DTMF end accepted without begin
>> '8' on SIP/MyTrunk-000a4b49
>> [May 17 00:33:36] DTMF[4238] channel.c: DTMF end passthrough '8' on
>> SIP/MyTrunk-000a4b49
>> [May 17 00:33:36] DTMF[4104] channel.c: DTMF end '8' received on
>> SIP/MAN-000a4af0, duration 100 ms
>> [May 17 00:33:36] DTMF[4104] channel.c: DTMF begin emulation of '8' with
>> duration 100 queued on SIP/MAN-000a4af0
>> [May 17 00:33:36] DTMF[4104] channel.c: DTMF end emulation of '8' queued
>> on SIP/MAN-000a4af0
>> [May 17 00:33:37] DTMF[4234] channel.c: DTMF end '1' received on
>> SIP/MAN-000a4b41, duration 100 ms
>> [May 17 00:33:37] DTMF[4234] channel.c: DTMF begin emulation of '1' with
>> duration 100 queued on SIP/MAN-000a4b41
>> [May 17 00:33:37] DTMF[4234] channel.c: DTMF end emulation of '1' queued
>> on SIP/MAN-000a4b41
>> [May 17 00:33:55] VERBOSE[4106] pbx.c:   == Spawn extension
>> (sip-trunk-inbound, 2127773456, 1) exited non-zero on
>> 'SIP/MyTrunk-000a4af3'
>> [May 17 00:33:56] VERBOSE[4136] pbx.c: -- Executing [h@trunk-outbound:1]
>> NoOp("SIP/MAN-000a4b09", "16") in new stack
>> [May 17 00:33:56] VERBOSE[4136] pbx.c:   == Spawn extension
>> (trunk-outbound, 87457712, 2) exited non-zero on 'SIP/MAN-000a4b09'
>>
>> Is this some thing related to SIP RE-INVITE?
>>
>> Thanks.
>>
>>
>> --
>> _
>> -- 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
>>
>
>
> --
> _
> -- 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
>
--
_
-- 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

Re: [asterisk-users] DTMF recognized after call establishment

[Gopalakrishnan N]

Let me try with dtmfmode as auto...
On 28 May 2013 19:32, "Asghar Mohammad"  wrote:

> work around was block dtmf.
> set wrong type of dtmf in incoming trunk.
>
>
> On Tue, May 28, 2013 at 11:15 AM, Gopalakrishnan N <
> gopalakrishnan...@gmail.com> wrote:
>
>> So any resolution for this?
>>
>> I suspect it could be related to RE INVITE
>>
>>
>> On Tue, May 28, 2013 at 2:09 PM, Asghar Mohammad wrote:
>>
>>> i had this in past there was an ATA configured to send 9 at the end of
>>> dialing in my case.
>>>
>>>
>>> On Tue, May 28, 2013 at 8:21 AM, Gopalakrishnan N <
>>> gopalakrishnan...@gmail.com> wrote:
>>>
>>>> Hi,
>>>>
>>>> I am receiving DTMF without any reason after call establishment.
>>>>
>>>> The log as follows, and I suspect something related to directmedia,
>>>> [May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49
>>>> is making progress passing it to SIP/MAN-000a4b48
>>>> [May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49
>>>> answered SIP/MAN-000a4b48
>>>> [May 17 00:33:35] DTMF[4238] channel.c: DTMF end '*' received on
>>>> SIP/MyTrunk-000a4b49, duration 0 ms
>>>> [May 17 00:33:35] DTMF[4238] channel.c: DTMF end accepted without begin
>>>> '*' on SIP/MyTrunk-000a4b49
>>>> [May 17 00:33:35] DTMF[4238] channel.c: DTMF end passthrough '*' on
>>>> SIP/MyTrunk-000a4b49
>>>> [May 17 00:33:36] DTMF[4238] channel.c: DTMF end '8' received on
>>>> SIP/MyTrunk-000a4b49, duration 0 ms
>>>> [May 17 00:33:36] DTMF[4238] channel.c: DTMF end accepted without begin
>>>> '8' on SIP/MyTrunk-000a4b49
>>>> [May 17 00:33:36] DTMF[4238] channel.c: DTMF end passthrough '8' on
>>>> SIP/MyTrunk-000a4b49
>>>> [May 17 00:33:36] DTMF[4104] channel.c: DTMF end '8' received on
>>>> SIP/MAN-000a4af0, duration 100 ms
>>>> [May 17 00:33:36] DTMF[4104] channel.c: DTMF begin emulation of '8'
>>>> with duration 100 queued on SIP/MAN-000a4af0
>>>> [May 17 00:33:36] DTMF[4104] channel.c: DTMF end emulation of '8'
>>>> queued on SIP/MAN-000a4af0
>>>> [May 17 00:33:37] DTMF[4234] channel.c: DTMF end '1' received on
>>>> SIP/MAN-000a4b41, duration 100 ms
>>>> [May 17 00:33:37] DTMF[4234] channel.c: DTMF begin emulation of '1'
>>>> with duration 100 queued on SIP/MAN-000a4b41
>>>> [May 17 00:33:37] DTMF[4234] channel.c: DTMF end emulation of '1'
>>>> queued on SIP/MAN-000a4b41
>>>> [May 17 00:33:55] VERBOSE[4106] pbx.c:   == Spawn extension
>>>> (sip-trunk-inbound, 2127773456, 1) exited non-zero on
>>>> 'SIP/MyTrunk-000a4af3'
>>>> [May 17 00:33:56] VERBOSE[4136] pbx.c: -- Executing
>>>> [h@trunk-outbound:1] NoOp("SIP/MAN-000a4b09", "16") in new stack
>>>> [May 17 00:33:56] VERBOSE[4136] pbx.c:   == Spawn extension
>>>> (trunk-outbound, 87457712, 2) exited non-zero on 'SIP/MAN-000a4b09'
>>>>
>>>> Is this some thing related to SIP RE-INVITE?
>>>>
>>>> Thanks.
>>>>
>>>>
>>>> --
>>>> _
>>>> -- 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
>>>>
>>>
>>>
>>> --
>>> _
>>> -- 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
>>>
>>
>>
>> --
>> _
>> -- 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
>>
>
>
> --
> _
> -- 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
>
--
_
-- 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

[asterisk-users] Queue Periodic Announce not working...

[Gopalakrishnan N]

I am having a queue where included periodic announce like the below,

[test]
context = default
member = Agent/1001
member = Agent/1002
music = default
strategy = rrmemory
ringinuse = no
timeout = 15
retry = 1
maxlen = 0
joinempty = yes
leavewhenempty = no
periodic-announce = /var/lib/asterisk/sounds/en/test/AVG-15.wav
periodic-announce-frequency=30
random-periodic-announce=no
relative-periodic-announce=yes
wrapuptime = 10

When am entering into the queue, the CLI shows playing periodic announce
but actually its not playing. Even I do have the file in the proper
directory.

Below is the CLI log,


--  Playing 'avgtest/AVG-13.slin' (language 'en')
[May 30 05:50:07] NOTICE[2711]: chan_sip.c:24257 handle_request_subscribe:
Received SIP subscribe for peer without mailbox: 1001
-- Executing [500@default:10] Queue("SIP/1001-0010", "avg") in new
stack
-- Started music on hold, class 'avgtest', on SIP/1001-0010
-- Stopped music on hold on SIP/1001-0010
-- Playing periodic announcement
-- Started music on hold, class 'avgtest', on SIP/1001-0010


Regards
--
_
-- 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

Re: [asterisk-users] Queue Periodic Announce not working...

[Gopalakrishnan N]

It works.

Thanks
 On 30 May 2013 19:39, "Doug Lytle"  wrote:

> >> periodic-announce = /var/lib/asterisk/sounds/en/test/AVG-15.wav
>
> Try it without the .wav
>
> Doug
>
> --
> Ben Franklin quote:
>
> "Those who would give up Essential Liberty to purchase a little Temporary
> Safety, deserve neither Liberty nor Safety."
>
> --
> _
> -- 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
>
--
_
-- 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

Re: [asterisk-users] Most suitable version for Production ENV

[Gopalakrishnan N]

Asterisk 1.8 is stable
On 1 Jun 2013 16:40, "luke devon"  wrote:

> Hi
>
> As I seen on the Asterisk web site , there is packages called ,
>
> AsteriskLatest Version - 11.4.0
>
> asterisk-11-current.tar.gz
>  and
>
> asterisk-1.8-current.tar.gz
>
> May I now which one is the most suitable for a production environment ?
>
> Thanks in advance
> Luke
>
>
> --
> _
> -- 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
>
--
_
-- 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

[asterisk-users] Codec Mismatch

[Gopalakrishnan N]

Sometimes in huge call volume am facing this type of error,

[Jun  4 08:42:46] WARNING[8459][C-79fa]: channel.c:5075 ast_write:
Codec mismatch on channel Local/8038@xss-call-out-4774;1 setting write
format to slin from ulaw native formats (ulaw)
[Jun  4 08:43:04] WARNING[8285][C-79da]: channel.c:5075 ast_write:
Codec mismatch on channel Local/6513@xss-call-out-4775;1 setting write
format to slin from ulaw native formats (ulaw)
[Jun  4 08:43:10] WARNING[8790][C-7a2c]: channel.c:5075 ast_write:
Codec mismatch on channel Local/18002662279@xss-call-out-4778;1 setting
write format to slin from ulaw native formats (ulaw)
[Jun  4 08:43:23] WARNING[8355][C-79e6]: channel.c:5075 ast_write:
Codec mismatch on channel Local/2896@xss-call-out-4779;1 setting write
format to slin from ulaw native formats (ulaw)
[Jun  4 08:43:25] WARNING[7577][C-798a]: channel.c:5075 ast_write:
Codec mismatch on channel Local/2896@xss-call-out-477a;1 setting write
format to slin from ulaw native formats (ulaw)


basically Asterisk will do the slin to ulaw, hope there should not be any
problem...

But am not sure why am getting this error? will this affect my call?
--
_
-- 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

[asterisk-users] Asterisk HA

[Gopalakrishnan N]

I was go through'ing the following links for HA,

https://wiki.asterisk.org/wiki/display/TOP/Failover+-+Linux - which doesn't
have file syncing.

https://www.johncahill.net/wiki/index.php/2_Node_Active/Passive_cluster -
this one has file syncing with pacemaker

Any other HA applications available or the lsyncd with pacemaker is good?

Regards
--
_
-- 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

Re: [asterisk-users] Executing Script after MixMonitor is called

[Gopalakrishnan N]

Hi Satish,

I tried with sox, without any parameter, just sox filename.wav to
filename.mp3, in linux shell prompt... the file is been converted...

Now If i want to run that command using dialplan,

MixMonitor(filename.wav,m)
Monitor_Exec(sox filename.wav filename.mp3)

Or to use System command?

Regards..


On Fri, Jan 27, 2012 at 11:29 AM, Satish Barot wrote:

> This is how I use a wav to mp3 script on Mixmonitor in my dialplan
> (Asterisk 1.8.7.0).
> ...
> same => n,MixMonitor(${FILENAME},W(4),/var/spool/asterisk/wav2mp3
> ^{FILENAME})
> ...
> and my script is...
>
> #!/bin/bash
>
> WAV="/var/spool/asterisk/monitor/$1"
> MP3=$(echo $1 | sed 's/\.wav$/.mp3/')
> MP3DEST="/var/spool/asterisk/mp3/$MP3"
> /usr/bin/lame "${WAV}" "${MP3DEST}" --silent -b 16 -s 9.6 -m m --bitwidth
> 8 --lowpass 9.6 --resample 8 --lowpass-width 1
>
> --SATISH BAROT
> Ahmedabad,India.
>
>
> On Wed, Jan 25, 2012 at 8:59 PM, Faraj Khasib wrote:
>
>> Hello Guys,
>> I am trying to convert files that are .wac to mp3 after mixmonitor
>> command is called but it doesnt execute the command, I tried the command in
>> terminal it worked, any help please ... below is my dial plan
>> exten=6500,n,Set(MIXMONITOR_EXEC=&& nice -n 19 /usr/local/bin/lame -b 8
>> -t -F -m m --bitwidth 8 --quiet
>> "/var/spool/asterisk/monitor/${CALLFILENAME}.wav"
>> "/var/spool/asterisk/monitor/${CALLFILENAME}.mp3" && rm -f
>> "/var/spool/asterisk/monitor/${CALLFILENAME}.wav")
>> exten=6500,n,MixMonitor(${CALLFILENAME}.wav,b)
>>
>> --
>> _
>> -- 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
>>
>
>
> --
> _
> -- 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
>
--
_
-- 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

[asterisk-users] Asterisk Queue Frame

[Gopalakrishnan N]

What happens when we increase the queue frame size in channels.c

if ((queued_frames + new_frames > 128 || queued_voice_frames +
new_voice_frames > 96)) {

Be default it is 128 and 96 if i increase it to 256 and 192 what will
happen? will it impact to default behavior?


Regards,
Gopal.
--
_
-- 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

Re: [asterisk-users] Asterisk Queue Frame

[Gopalakrishnan N]

actually when i get the message my call volume is around 180 to 200
calls will that matter... and some calls being transferred to other
location and some are making outbound calls, some are inbound...

Is there any way for optimization?


On Fri, Jun 21, 2013 at 5:57 AM, Richard Mudgett wrote:

> On Thu, Jun 20, 2013 at 6:55 PM, Gopalakrishnan N <
> gopalakrishnan...@gmail.com> wrote:
>
>> What happens when we increase the queue frame size in channels.c
>>
>> if ((queued_frames + new_frames > 128 || queued_voice_frames +
>> new_voice_frames > 96)) {
>>
>> Be default it is 128 and 96 if i increase it to 256 and 192 what will
>> happen? will it impact to default behavior?
>>
>
> It looks like you are getting the "Exceptionally long queue length"
> warning message.  The
> change you mention will just increase the allowed size of the queue.
> Doing that won't really
> help much as it will just delay getting the message.  That warning message
> means Asterisk
> is falling behind in processing frames on the channel.
>
> Richard
>
> --
> _
> -- 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
>
--
_
-- 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

[asterisk-users] IAX2 netsock error with name resolution

[Gopalakrishnan N]

Am getting netsock error like this when using IAX2,

Connected to Asterisk 11.2.1 currently running on indiaprimaryast01 (pid =
4270)
  == Using SIP RTP CoS mark 5
-- Executing [2001@Test:1] Dial("SIP/4090-0005",
"SIP/2001@IAX2/IND-MAN,30")
in new stack
[Jun 23 06:31:36] NOTICE[4383][C-0005]: chan_sip.c:29491
sip_request_call: Conflicting extension values given. Using '2001' and not
'IND-MAN'
  == Using SIP RTP CoS mark 5
[Jun 23 06:31:36] ERROR[4383][C-0005]: netsock2.c:269
ast_sockaddr_resolve: getaddrinfo("IAX2", "(null)", ...): Temporary failure
in name resolution
[Jun 23 06:31:36] WARNING[4383][C-0005]: chan_sip.c:6191 create_addr:
No such host: IAX2
[Jun 23 06:31:36] WARNING[4383][C-0005]: app_dial.c:2437
dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -
Subscriber absent)
  == Everyone is busy/congested at this time (1:0/0/1)

My hostname are proper,
in /etc/hostname and /etc/sysconfig/network

Even then am not able to find why am getting this error. Also am able to
ping with my own hostname.

Regards,
--
_
-- 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

Re: [asterisk-users] IAX2 netsock error with name resolution

[Gopalakrishnan N]

After changing my dialplan as suggested, there is no socket error, but
getting Busy/Congested, and the call is hanging up, let me check that
part...

Earlier my dialplan was,
;exten => _2XXX,1,Dial(SIP/${EXTEN}@${MANIAX},30)

and I changed like this exten => _2XXX,1,Dial(${MANIAX}/${EXTEN},30)

whether the SIP matters?

And now since its a SIP extension in other side, am getting failed because
the extension is not able to find.


Regards.


On Sun, Jun 23, 2013 at 5:22 PM, Alec Davis  wrote:

> 
> > -- Executing [2001@Test:1] Dial("SIP/4090-0005",
> "SIP/2001@IAX2/IND-MAN,30") in new stack
> > [Jun 23 06:31:36] NOTICE[4383][C-0005]: chan_sip.c:29491
> sip_request_call: Conflicting extension values given. Using '2001' and not
> 'IND-MAN'
> >   == Using SIP RTP CoS mark 5
> > [Jun 23 06:31:36] ERROR[4383][C-0005]: netsock2.c:269
> ast_sockaddr_resolve: getaddrinfo("IAX2", "(null)", ...): Temporary failure
> in name resolution
> > [Jun 23 06:31:36] WARNING[4383][C-0005]: chan_sip.c:6191 create_addr:
> No such host: IAX2
> > [Jun 23 06:31:36] WARNING[4383][C-0005]: app_dial.c:2437
> dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -
> Subscriber absent)
> >   == Everyone is busy/congested at this time (1:0/0/1)
>
> Try this syntax Dial(IAX2/IND-MAN/2001,30)
> Where IND-MAN is the name of a peer/friend [IND-MAN] defined in iax.conf
> and 2001 is the extension on the remote system 'IND-MAN' where 2001 dials
> SIP/2001
>
> Alec Davis
>
>
> --
> _
> -- 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
>
--
_
-- 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

[asterisk-users] Asterisk trunking between two location

[Gopalakrishnan N]

Am using Asterisk 11.2 in one location and 11.1 in another location.

when I trunk between two servers, the status is unreachable.

But with different server with 11.2 and 11.2 it works fine.

I tried both IAX and SIP.

the trunk in sip.conf what i have is,
[serverb]
type=friend
username=serverb
secret=serverb
host=10.10.10.5
port=5060
context=default
insecure=port,invite
dtmfmode=rfc2833
relaxdtmf=yes
directmedia=no
qualify=3000
nat=force_rport,comedia
disallow=all
allow=g729
allow=ulaw
allow=alaw
deny=0.0.0.0/0.0.0.0
permit=10.10.10.5/255.255.255.0

Is there any issue with 11.1?
--
_
-- 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

Re: [asterisk-users] Asterisk trunking between two location

[Gopalakrishnan N]

Also tried one more scenario, particularly from one IP to other IP not
registering.

For example like 10.10.10.5 to 10.20.10.5

If it is 10.10.10.5 to 10.30.2.5 - working
If it is 10.30.2.5 to 10.20.10.4 works fine.

really strange... I suspect some issue on the network side...

Problem is there is no packet loss.. with mtr it is fine, tracepath is
fine, ping is fine... :(


On Wed, Jul 3, 2013 at 1:05 AM, Gopalakrishnan N <
gopalakrishnan...@gmail.com> wrote:

> Am using Asterisk 11.2 in one location and 11.1 in another location.
>
> when I trunk between two servers, the status is unreachable.
>
> But with different server with 11.2 and 11.2 it works fine.
>
> I tried both IAX and SIP.
>
> the trunk in sip.conf what i have is,
> [serverb]
> type=friend
> username=serverb
> secret=serverb
> host=10.10.10.5
> port=5060
> context=default
> insecure=port,invite
> dtmfmode=rfc2833
> relaxdtmf=yes
> directmedia=no
> qualify=3000
> nat=force_rport,comedia
> disallow=all
> allow=g729
> allow=ulaw
> allow=alaw
> deny=0.0.0.0/0.0.0.0
> permit=10.10.10.5/255.255.255.0
>
> Is there any issue with 11.1?
>
--
_
-- 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

Re: [asterisk-users] Asterisk trunking between two location

[Gopalakrishnan N]

[servera]
type=friend
username=servera
secret=servera
host=10.30.2.5
port=5060
context=Manila
insecure=port,invite
dtmfmode=rfc2833
relaxdtmf=yes
directmedia=no
qualify=yes
disallow=all
allow=g729
allow=ulaw
allow=alaw
deny=0.0.0.0/0.0.0.0
permit=10.30.2.5/255.255.255.0

If i use host=dynamic, it wont communicate each other and will result to
unmonitored


and the IP segment is two different segment. where am able to ping each
other.



On Wed, Jul 3, 2013 at 1:29 AM, Asghar Mohammad  wrote:

> hi,
> paste server a trunk also, if you want register why you are not using
> host=dynamic?
> both servers are on 10.10.10.0 ? if no then check your deny permit seting.
>
>
> On Tue, Jul 2, 2013 at 9:53 PM, Gopalakrishnan N <
> gopalakrishnan...@gmail.com> wrote:
>
>> Also tried one more scenario, particularly from one IP to other IP not
>> registering.
>>
>> For example like 10.10.10.5 to 10.20.10.5
>>
>> If it is 10.10.10.5 to 10.30.2.5 - working
>> If it is 10.30.2.5 to 10.20.10.4 works fine.
>>
>> really strange... I suspect some issue on the network side...
>>
>> Problem is there is no packet loss.. with mtr it is fine, tracepath is
>> fine, ping is fine... :(
>>
>>
>> On Wed, Jul 3, 2013 at 1:05 AM, Gopalakrishnan N <
>> gopalakrishnan...@gmail.com> wrote:
>>
>>> Am using Asterisk 11.2 in one location and 11.1 in another location.
>>>
>>> when I trunk between two servers, the status is unreachable.
>>>
>>> But with different server with 11.2 and 11.2 it works fine.
>>>
>>> I tried both IAX and SIP.
>>>
>>> the trunk in sip.conf what i have is,
>>> [serverb]
>>> type=friend
>>> username=serverb
>>> secret=serverb
>>> host=10.10.10.5
>>> port=5060
>>> context=default
>>> insecure=port,invite
>>> dtmfmode=rfc2833
>>> relaxdtmf=yes
>>> directmedia=no
>>> qualify=3000
>>> nat=force_rport,comedia
>>> disallow=all
>>> allow=g729
>>> allow=ulaw
>>> allow=alaw
>>> deny=0.0.0.0/0.0.0.0
>>> permit=10.10.10.5/255.255.255.0
>>>
>>> Is there any issue with 11.1?
>>>
>>
>>
>> --
>> _
>> -- 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
>>
>
>
> --
> _
> -- 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
>
--
_
-- 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

Re: [asterisk-users] Asterisk trunking between two location

[Gopalakrishnan N]

can't we use without register command both way as peer to peer?


On Wed, Jul 3, 2013 at 1:45 AM, Asghar Mohammad  wrote:

> 1. you permiting 10.10.10.0 on b but you should permit 10.30.2.0 on b and
> 10.10.10.0 on a.
> 2. use host=dynamic type=friend on  side A and host=ip type=peer on side B.
> 3. general section in sip.conf of side B register with server A.
>
> please see comments in sip.conf
> ;dynamic_exclude_static = yes   ; Disallow all dynamic hosts from
> registering
> ; as any IP address used for staticly
> defined
> ; hosts.  This helps avoid the
> configuration
> ; error of allowing your users to register
> at
> ; the same address as a SIP provider.
>
>
>
> On Tue, Jul 2, 2013 at 10:04 PM, Gopalakrishnan N <
> gopalakrishnan...@gmail.com> wrote:
>
>> [servera]
>> type=friend
>> username=servera
>> secret=servera
>> host=10.30.2.5
>> port=5060
>> context=Manila
>> insecure=port,invite
>> dtmfmode=rfc2833
>> relaxdtmf=yes
>> directmedia=no
>> qualify=yes
>> disallow=all
>> allow=g729
>> allow=ulaw
>> allow=alaw
>> deny=0.0.0.0/0.0.0.0
>> permit=10.30.2.5/255.255.255.0
>>
>> If i use host=dynamic, it wont communicate each other and will result to
>> unmonitored
>>
>>
>> and the IP segment is two different segment. where am able to ping each
>> other.
>>
>>
>>
>> On Wed, Jul 3, 2013 at 1:29 AM, Asghar Mohammad wrote:
>>
>>> hi,
>>> paste server a trunk also, if you want register why you are not using
>>> host=dynamic?
>>> both servers are on 10.10.10.0 ? if no then check your deny permit
>>> seting.
>>>
>>>
>>> On Tue, Jul 2, 2013 at 9:53 PM, Gopalakrishnan N <
>>> gopalakrishnan...@gmail.com> wrote:
>>>
>>>> Also tried one more scenario, particularly from one IP to other IP not
>>>> registering.
>>>>
>>>> For example like 10.10.10.5 to 10.20.10.5
>>>>
>>>> If it is 10.10.10.5 to 10.30.2.5 - working
>>>> If it is 10.30.2.5 to 10.20.10.4 works fine.
>>>>
>>>> really strange... I suspect some issue on the network side...
>>>>
>>>> Problem is there is no packet loss.. with mtr it is fine, tracepath is
>>>> fine, ping is fine... :(
>>>>
>>>>
>>>> On Wed, Jul 3, 2013 at 1:05 AM, Gopalakrishnan N <
>>>> gopalakrishnan...@gmail.com> wrote:
>>>>
>>>>> Am using Asterisk 11.2 in one location and 11.1 in another location.
>>>>>
>>>>> when I trunk between two servers, the status is unreachable.
>>>>>
>>>>> But with different server with 11.2 and 11.2 it works fine.
>>>>>
>>>>> I tried both IAX and SIP.
>>>>>
>>>>> the trunk in sip.conf what i have is,
>>>>> [serverb]
>>>>> type=friend
>>>>> username=serverb
>>>>> secret=serverb
>>>>> host=10.10.10.5
>>>>> port=5060
>>>>> context=default
>>>>> insecure=port,invite
>>>>> dtmfmode=rfc2833
>>>>> relaxdtmf=yes
>>>>> directmedia=no
>>>>> qualify=3000
>>>>> nat=force_rport,comedia
>>>>> disallow=all
>>>>> allow=g729
>>>>> allow=ulaw
>>>>> allow=alaw
>>>>> deny=0.0.0.0/0.0.0.0
>>>>> permit=10.10.10.5/255.255.255.0
>>>>>
>>>>> Is there any issue with 11.1?
>>>>>
>>>>
>>>>
>>>> --
>>>> _
>>>> -- 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
>>>>
>>>
>>>
>>> --
>>> _
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>

Re: [asterisk-users] Asterisk trunking between two location

[Gopalakrishnan N]

I tried creating two trunks with following,
*1st Location*
[10.30.2.5]
type=friend
username=indman01
secret=indman01
host=dynamic
port=5060
context=Manila
insecure=port,invite
dtmfmode=rfc2833
relaxdtmf=yes
directmedia=no
qualify=yes
disallow=all
allow=g729
allow=ulaw

*2nd Location*
[10.20.111.48]
type=friend
username=manind01
secret=manind01
host=dynamic
port=5060
context=india
insecure=port,invite
dtmfmode=rfc2833
relaxdtmf=yes
directmedia=no
qualify=yes
nat=force_rport,comedia
disallow=all
allow=g729
allow=ulaw
allow=alaw

My dialplan is like this
exten => _2XXX,1,Dial(SIP/10.30.2.5/${EXTEN})
exten => _2XXX,n,Hangup

And the output I get is
 Executing [2001@Test:1] Dial("SIP/3081-27d2", "SIP/10.30.2.5/2001") in
new stack
[Jul  2 16:49:57] WARNING[15766][C-2b94]: app_dial.c:2437
dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -
Subscriber absent)
  == Everyone is busy/congested at this time (1:0/0/1)
-- Executing [2001@Test:2] Hangup("SIP/3081-27d2", "") in new stack
  == Spawn extension (Test, 2001, 2) exited non-zero on 'SIP/3081-27d2'

Actually the trunk which i mentioned in my first email, it was working...
and from today it is not

Still breaking... what could be the reason... !



On Wed, Jul 3, 2013 at 2:05 AM, Asghar Mohammad  wrote:

> yes you can. just create trunks on both side with static ip and in dial
> use trunk name.
> exten => _X.,1,Dial(SIP/trunka/${EXTEN}) on side b and exten =>
> _X.,1,Dial(SIP/trunkb/${EXTEN}) on side a.
> make a call from a to b and one from b to and post cli log here or upload
> anyware else.
>
>
> On Tue, Jul 2, 2013 at 10:25 PM, Gopalakrishnan N <
> gopalakrishnan...@gmail.com> wrote:
>
>> can't we use without register command both way as peer to peer?
>>
>>
>> On Wed, Jul 3, 2013 at 1:45 AM, Asghar Mohammad wrote:
>>
>>> 1. you permiting 10.10.10.0 on b but you should permit 10.30.2.0 on b
>>> and 10.10.10.0 on a.
>>> 2. use host=dynamic type=friend on  side A and host=ip type=peer on side
>>> B.
>>> 3. general section in sip.conf of side B register with server A.
>>>
>>> please see comments in sip.conf
>>> ;dynamic_exclude_static = yes   ; Disallow all dynamic hosts from
>>> registering
>>> ; as any IP address used for staticly
>>> defined
>>> ; hosts.  This helps avoid the
>>> configuration
>>> ; error of allowing your users to
>>> register at
>>> ; the same address as a SIP provider.
>>>
>>>
>>>
>>> On Tue, Jul 2, 2013 at 10:04 PM, Gopalakrishnan N <
>>> gopalakrishnan...@gmail.com> wrote:
>>>
>>>> [servera]
>>>> type=friend
>>>> username=servera
>>>> secret=servera
>>>> host=10.30.2.5
>>>> port=5060
>>>> context=Manila
>>>> insecure=port,invite
>>>> dtmfmode=rfc2833
>>>> relaxdtmf=yes
>>>> directmedia=no
>>>> qualify=yes
>>>> disallow=all
>>>> allow=g729
>>>> allow=ulaw
>>>> allow=alaw
>>>> deny=0.0.0.0/0.0.0.0
>>>> permit=10.30.2.5/255.255.255.0
>>>>
>>>> If i use host=dynamic, it wont communicate each other and will result
>>>> to unmonitored
>>>>
>>>>
>>>> and the IP segment is two different segment. where am able to ping each
>>>> other.
>>>>
>>>>
>>>>
>>>> On Wed, Jul 3, 2013 at 1:29 AM, Asghar Mohammad wrote:
>>>>
>>>>> hi,
>>>>> paste server a trunk also, if you want register why you are not using
>>>>> host=dynamic?
>>>>> both servers are on 10.10.10.0 ? if no then check your deny permit
>>>>> seting.
>>>>>
>>>>>
>>>>> On Tue, Jul 2, 2013 at 9:53 PM, Gopalakrishnan N <
>>>>> gopalakrishnan...@gmail.com> wrote:
>>>>>
>>>>>> Also tried one more scenario, particularly from one IP to other IP
>>>>>> not registering.
>>>>>>
>>>>>> For example like 10.10.10.5 to 10.20.10.5
>>>>>>
>>>>>> If it is 10.10.10.5 to 10.30.2.5 - working
>>>>>> If it is 10.30.2.5 to 10.20.10.4 works fine.
>>>>>>
>>>>>> really strange... I suspect some issue on the network side...
>>>>>&g

Re: [asterisk-users] Asterisk trunking between two location

[Gopalakrishnan N]

still the peer shows unreachable let me restart and give a try...


On Wed, Jul 3, 2013 at 2:49 AM, Asghar Mohammad  wrote:

> *1st Location*
> [manila]
> type=peer
> username=indman01
> secret=indman01
> host=10.30.2.5 <-- ip of 2nd location
> port=5060
> context=Manila
> insecure=port,invite
> dtmfmode=rfc2833
> relaxdtmf=yes
> directmedia=no
> qualify=yes
> disallow=all
> allow=g729
> allow=ulaw
>
> 1st location dialplan
> exten => _2XXX,1,Dial(SIP/manila/${EXTEN} <http://10.30.2.5/$%7BEXTEN%7D>)
> exten => _2XXX,n,Hangup
>
> *2nd Location*
> [india]
> type=friend
> username=manind01
> secret=manind01
> host=dynamic
> port=5060
> context=10.20.111.48 <- ip of 1st location
>  insecure=port,invite
> dtmfmode=rfc2833
> relaxdtmf=yes
> directmedia=no
> qualify=yes
> nat=force_rport,comedia
> disallow=all
> allow=g729
> allow=ulaw
> allow=alaw
>
> 2st location dialplan
> exten => _2XXX,1,Dial(SIP/india/${EXTEN} <http://10.30.2.5/$%7BEXTEN%7D>)
> exten => _2XXX,n,Hangup
>
> then you should handle the call when it arrive in any server
> let me know if it work.
>
>
> On Tue, Jul 2, 2013 at 10:56 PM, Gopalakrishnan N <
> gopalakrishnan...@gmail.com> wrote:
>
>> I tried creating two trunks with following,
>> *1st Location*
>> [10.30.2.5]
>> type=friend
>> username=indman01
>> secret=indman01
>> host=dynamic
>> port=5060
>> context=Manila
>> insecure=port,invite
>> dtmfmode=rfc2833
>> relaxdtmf=yes
>> directmedia=no
>> qualify=yes
>> disallow=all
>> allow=g729
>> allow=ulaw
>>
>> *2nd Location*
>> [10.20.111.48]
>> type=friend
>> username=manind01
>> secret=manind01
>> host=dynamic
>> port=5060
>> context=india
>> insecure=port,invite
>> dtmfmode=rfc2833
>> relaxdtmf=yes
>> directmedia=no
>> qualify=yes
>> nat=force_rport,comedia
>> disallow=all
>> allow=g729
>> allow=ulaw
>> allow=alaw
>>
>> My dialplan is like this
>> exten => _2XXX,1,Dial(SIP/10.30.2.5/${EXTEN}<http://10.30.2.5/$%7BEXTEN%7D>
>> )
>> exten => _2XXX,n,Hangup
>>
>> And the output I get is
>>  Executing [2001@Test:1] Dial("SIP/3081-27d2", "SIP/10.30.2.5/2001")
>> in new stack
>> [Jul  2 16:49:57] WARNING[15766][C-2b94]: app_dial.c:2437
>> dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -
>> Subscriber absent)
>>   == Everyone is busy/congested at this time (1:0/0/1)
>> -- Executing [2001@Test:2] Hangup("SIP/3081-27d2", "") in new
>> stack
>>   == Spawn extension (Test, 2001, 2) exited non-zero on
>> 'SIP/3081-27d2'
>>
>> Actually the trunk which i mentioned in my first email, it was working...
>> and from today it is not
>>
>> Still breaking... what could be the reason... !
>>
>>
>>
>> On Wed, Jul 3, 2013 at 2:05 AM, Asghar Mohammad wrote:
>>
>>> yes you can. just create trunks on both side with static ip and in dial
>>> use trunk name.
>>> exten => _X.,1,Dial(SIP/trunka/${EXTEN}) on side b and exten =>
>>> _X.,1,Dial(SIP/trunkb/${EXTEN}) on side a.
>>> make a call from a to b and one from b to and post cli log here or
>>> upload anyware else.
>>>
>>>
>>> On Tue, Jul 2, 2013 at 10:25 PM, Gopalakrishnan N <
>>> gopalakrishnan...@gmail.com> wrote:
>>>
>>>> can't we use without register command both way as peer to peer?
>>>>
>>>>
>>>> On Wed, Jul 3, 2013 at 1:45 AM, Asghar Mohammad wrote:
>>>>
>>>>> 1. you permiting 10.10.10.0 on b but you should permit 10.30.2.0 on b
>>>>> and 10.10.10.0 on a.
>>>>> 2. use host=dynamic type=friend on  side A and host=ip type=peer on
>>>>> side B.
>>>>> 3. general section in sip.conf of side B register with server A.
>>>>>
>>>>> please see comments in sip.conf
>>>>> ;dynamic_exclude_static = yes   ; Disallow all dynamic hosts from
>>>>> registering
>>>>> ; as any IP address used for staticly
>>>>> defined
>>>>> ; hosts.  This helps avoid the
>>>>> configuration
>>>>> ; error of allowing your users to
>>>>> register at
>>>>> 

Re: [asterisk-users] Asterisk trunking between two location

[Gopalakrishnan N]

By having different server, i made it work. I suspect some network issue...


On Wed, Jul 3, 2013 at 3:27 AM, Asghar Mohammad  wrote:

> make a call and post cli log
>
>
> On Tue, Jul 2, 2013 at 11:54 PM, Gopalakrishnan N <
> gopalakrishnan...@gmail.com> wrote:
>
>> still the peer shows unreachable let me restart and give a try...
>>
>>
>> On Wed, Jul 3, 2013 at 2:49 AM, Asghar Mohammad wrote:
>>
>>> *1st Location*
>>> [manila]
>>> type=peer
>>> username=indman01
>>> secret=indman01
>>> host=10.30.2.5 <-- ip of 2nd location
>>> port=5060
>>> context=Manila
>>> insecure=port,invite
>>> dtmfmode=rfc2833
>>> relaxdtmf=yes
>>> directmedia=no
>>> qualify=yes
>>> disallow=all
>>> allow=g729
>>> allow=ulaw
>>>
>>> 1st location dialplan
>>> exten => _2XXX,1,Dial(SIP/manila/${EXTEN}<http://10.30.2.5/$%7BEXTEN%7D>
>>> )
>>> exten => _2XXX,n,Hangup
>>>
>>> *2nd Location*
>>> [india]
>>> type=friend
>>> username=manind01
>>> secret=manind01
>>> host=dynamic
>>> port=5060
>>> context=10.20.111.48 <- ip of 1st location
>>>  insecure=port,invite
>>> dtmfmode=rfc2833
>>> relaxdtmf=yes
>>> directmedia=no
>>> qualify=yes
>>> nat=force_rport,comedia
>>> disallow=all
>>> allow=g729
>>> allow=ulaw
>>> allow=alaw
>>>
>>> 2st location dialplan
>>> exten => _2XXX,1,Dial(SIP/india/${EXTEN} <http://10.30.2.5/$%7BEXTEN%7D>
>>> )
>>> exten => _2XXX,n,Hangup
>>>
>>> then you should handle the call when it arrive in any server
>>> let me know if it work.
>>>
>>>
>>> On Tue, Jul 2, 2013 at 10:56 PM, Gopalakrishnan N <
>>> gopalakrishnan...@gmail.com> wrote:
>>>
>>>> I tried creating two trunks with following,
>>>> *1st Location*
>>>> [10.30.2.5]
>>>> type=friend
>>>> username=indman01
>>>> secret=indman01
>>>> host=dynamic
>>>> port=5060
>>>> context=Manila
>>>> insecure=port,invite
>>>> dtmfmode=rfc2833
>>>> relaxdtmf=yes
>>>> directmedia=no
>>>> qualify=yes
>>>> disallow=all
>>>> allow=g729
>>>> allow=ulaw
>>>>
>>>> *2nd Location*
>>>> [10.20.111.48]
>>>> type=friend
>>>> username=manind01
>>>> secret=manind01
>>>> host=dynamic
>>>> port=5060
>>>> context=india
>>>> insecure=port,invite
>>>> dtmfmode=rfc2833
>>>> relaxdtmf=yes
>>>> directmedia=no
>>>> qualify=yes
>>>> nat=force_rport,comedia
>>>> disallow=all
>>>> allow=g729
>>>> allow=ulaw
>>>> allow=alaw
>>>>
>>>> My dialplan is like this
>>>> exten => _2XXX,1,Dial(SIP/10.30.2.5/${EXTEN}<http://10.30.2.5/$%7BEXTEN%7D>
>>>> )
>>>> exten => _2XXX,n,Hangup
>>>>
>>>> And the output I get is
>>>>  Executing [2001@Test:1] Dial("SIP/3081-27d2", "SIP/10.30.2.5/2001")
>>>> in new stack
>>>> [Jul  2 16:49:57] WARNING[15766][C-2b94]: app_dial.c:2437
>>>> dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -
>>>> Subscriber absent)
>>>>   == Everyone is busy/congested at this time (1:0/0/1)
>>>> -- Executing [2001@Test:2] Hangup("SIP/3081-27d2", "") in new
>>>> stack
>>>>   == Spawn extension (Test, 2001, 2) exited non-zero on
>>>> 'SIP/3081-27d2'
>>>>
>>>> Actually the trunk which i mentioned in my first email, it was
>>>> working... and from today it is not
>>>>
>>>> Still breaking... what could be the reason... !
>>>>
>>>>
>>>>
>>>> On Wed, Jul 3, 2013 at 2:05 AM, Asghar Mohammad wrote:
>>>>
>>>>> yes you can. just create trunks on both side with static ip and in
>>>>> dial use trunk name.
>>>>> exten => _X.,1,Dial(SIP/trunka/${EXTEN}) on side b and exten =>
>>>>> _X.,1,Dial(SIP/trunkb/${EXTEN}) on side a.
>>>>> make a call from a to b and one from b to and post cli log here or
>>>>>

Re: [asterisk-users] Executing Script after MixMonitor is called

[Gopalakrishnan N]

I tried with hangup cause but my script is not executed... also I tried the
same script with mix monitor itself no sucess.

The script what I have is, am converting wav file to flac format..
On 11 Jun 2013 11:17, "Satish Barot"  wrote:

> And yes if you want to use System application in your dialplan then have
> System in your h extension
>
> System(/PathToSox/sox -r 8000 -c 1 /PathToWavFile/filename.wav 
> /PathToMp3FileToBE Stored/filename.mp3)
>
>
>
>
>
>
> On Tue, Jun 11, 2013 at 10:38 AM, Satish Barot 
> wrote:
>
>> Hi Gopamkrishnan,
>>
>> Check the 'command' argument for Mixmonitor. Mixmonitor itself has a
>> facility to execute a command when recording is over.
>>
>> *In my case, 'wav2mp3' is a script which gets executed and converts recorded 
>> wav audio file to mp3. I pass ${FILENAME} as an argument to my script.
>> *
>>
>> *You should have something like 
>> *MixMonitor(filename.wav,m,/PathToYourScript/YourScriptName^filename.wav) in 
>> your dialplan.
>>
>> Hope this helps.
>>
>> --Satish Barot
>>
>>
>> Ahmedabad, India
>>
>>
>>
>>
>>
>> On Tue, Jun 11, 2013 at 9:31 AM, Gopalakrishnan N <
>> gopalakrishnan...@gmail.com> wrote:
>>
>>> Hi Satish,
>>>
>>> I tried with sox, without any parameter, just sox filename.wav to
>>> filename.mp3, in linux shell prompt... the file is been converted...
>>>
>>> Now If i want to run that command using dialplan,
>>>
>>> MixMonitor(filename.wav,m)
>>> Monitor_Exec(sox filename.wav filename.mp3)
>>>
>>> Or to use System command?
>>>
>>> Regards..
>>>
>>>
>>> On Fri, Jan 27, 2012 at 11:29 AM, Satish Barot <
>>> satish4aster...@gmail.com> wrote:
>>>
>>>> This is how I use a wav to mp3 script on Mixmonitor in my dialplan
>>>> (Asterisk 1.8.7.0).
>>>> ...
>>>> same => n,MixMonitor(${FILENAME},W(4),/var/spool/asterisk/wav2mp3
>>>> ^{FILENAME})
>>>> ...
>>>> and my script is...
>>>>
>>>> #!/bin/bash
>>>>
>>>> WAV="/var/spool/asterisk/monitor/$1"
>>>> MP3=$(echo $1 | sed 's/\.wav$/.mp3/')
>>>> MP3DEST="/var/spool/asterisk/mp3/$MP3"
>>>> /usr/bin/lame "${WAV}" "${MP3DEST}" --silent -b 16 -s 9.6 -m m
>>>> --bitwidth 8 --lowpass 9.6 --resample 8 --lowpass-width 1
>>>>
>>>> --SATISH BAROT
>>>> Ahmedabad,India.
>>>>
>>>>
>>>> On Wed, Jan 25, 2012 at 8:59 PM, Faraj Khasib 
>>>> wrote:
>>>>
>>>>> Hello Guys,
>>>>> I am trying to convert files that are .wac to mp3 after mixmonitor
>>>>> command is called but it doesnt execute the command, I tried the command 
>>>>> in
>>>>> terminal it worked, any help please ... below is my dial plan
>>>>> exten=6500,n,Set(MIXMONITOR_EXEC=&& nice -n 19 /usr/local/bin/lame -b
>>>>> 8 -t -F -m m --bitwidth 8 --quiet
>>>>> "/var/spool/asterisk/monitor/${CALLFILENAME}.wav"
>>>>> "/var/spool/asterisk/monitor/${CALLFILENAME}.mp3" && rm -f
>>>>> "/var/spool/asterisk/monitor/${CALLFILENAME}.wav")
>>>>> exten=6500,n,MixMonitor(${CALLFILENAME}.wav,b)
>>>>>
>>>>> --
>>>>> _
>>>>> -- 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
>>>>>
>>>>
>>>>
>>>> --
>>>> _
>>>> -- 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
>>>>
>>>
>>>
>>> --
>>> _
>>> -- 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
>>>
>>
>>
>
> --
> _
> -- 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
>
--
_
-- 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

Re: [asterisk-users] Executing Script after MixMonitor is called

[Gopalakrishnan N]

exten => _4X.,1,Set(START_TIME=${STRFTIME(${EPOCH},,%Y-%m-%d_%H:%M:%S)})
exten => _4X.,n,Set(MIXMONITOR_FILENAME=${EXTEN}-${START_TIME}-OUT)
;exten =>
_4X.,n,MixMonitor(IND_PRI/${CDR(accountcode)}/OUT/${STRFTIME(${EPOCH},,%Y-%m)}/${STRFTIME(${EPOCH},,%d)}/${MIXMONITOR_FILENAME}.wav,m,/root/flac.sh
${MIXMONITOR_FILENAME}.wav)
exten =>
_4X.,n,MixMonitor(IND_PRI/${CDR(accountcode)}/OUT/${STRFTIME(${EPOCH},,%Y-%m)}/${STRFTIME(${EPOCH},,%d)}/${MIXMONITOR_FILENAME}.wav,b,/root/flac.sh
${MIXMONITOR_FILENAME}.wav)
exten =>
_4X.,n,Set(CDR(userfield)=IND_PRI/${CDR(accountcode)}/OUT/${STRFTIME(${EPOCH},,%Y-%m)}/${STRFTIME(${EPOCH},,%d)}/${MIXMONITOR_FILENAME})
exten => _4X.,n,Dial(SIP/${EXTEN},30)
exten => _4X.,n,Hangup

Regards
On 4 Jul 2013 11:18, "Satish Barot"  wrote:

> On Thu, Jul 4, 2013 at 1:30 AM, Gopalakrishnan N <
> gopalakrishnan...@gmail.com> wrote:
>
>> I tried with hangup cause but my script is not executed... also I tried
>> the same script with mix monitor itself no sucess.
>>
>> The script what I have is, am converting wav file to flac format..
>> On 11 Jun 2013 11:17, "Satish Barot"  wrote:
>>
>>> And yes if you want to use System application in your dialplan then have
>>> System in your h extension
>>>
>>> System(/PathToSox/sox -r 8000 -c 1 /PathToWavFile/filename.wav 
>>> /PathToMp3FileToBE Stored/filename.mp3)
>>>
>>> On Tue, Jun 11, 2013 at 10:38 AM, Satish Barot <
>>> satish4aster...@gmail.com> wrote:
>>>
>>>> Hi Gopamkrishnan,
>>>>
>>>> Check the 'command' argument for Mixmonitor. Mixmonitor itself has a
>>>> facility to execute a command when recording is over.
>>>>
>>>> *In my case, 'wav2mp3' is a script which gets executed and converts 
>>>> recorded wav audio file to mp3. I pass ${FILENAME} as an argument to my 
>>>> script.*
>>>>
>>>> *You should have something like 
>>>> *MixMonitor(filename.wav,m,/PathToYourScript/YourScriptName^filename.wav) 
>>>> in your dialplan.
>>>>
>>>> Hope this helps.
>>>>
>>>> --Satish Barot
>>>>
>>>>
>>>> Ahmedabad, India
>>>>
>>>>
>>>> On Tue, Jun 11, 2013 at 9:31 AM, Gopalakrishnan N <
>>>> gopalakrishnan...@gmail.com> wrote:
>>>>
>>>>> Hi Satish,
>>>>>
>>>>> I tried with sox, without any parameter, just sox filename.wav to
>>>>> filename.mp3, in linux shell prompt... the file is been converted...
>>>>>
>>>>> Now If i want to run that command using dialplan,
>>>>>
>>>>> MixMonitor(filename.wav,m)
>>>>> Monitor_Exec(sox filename.wav filename.mp3)
>>>>>
>>>>> Or to use System command?
>>>>>
>>>>> Regards..
>>>>>
>>>>>
>>>>> On Fri, Jan 27, 2012 at 11:29 AM, Satish Barot <
>>>>> satish4aster...@gmail.com> wrote:
>>>>>
>>>>>> This is how I use a wav to mp3 script on Mixmonitor in my dialplan
>>>>>> (Asterisk 1.8.7.0).
>>>>>> ...
>>>>>> same => n,MixMonitor(${FILENAME},W(4),/var/spool/asterisk/wav2mp3
>>>>>> ^{FILENAME})
>>>>>> ...
>>>>>> and my script is...
>>>>>>
>>>>>> #!/bin/bash
>>>>>>
>>>>>> WAV="/var/spool/asterisk/monitor/$1"
>>>>>> MP3=$(echo $1 | sed 's/\.wav$/.mp3/')
>>>>>> MP3DEST="/var/spool/asterisk/mp3/$MP3"
>>>>>> /usr/bin/lame "${WAV}" "${MP3DEST}" --silent -b 16 -s 9.6 -m m
>>>>>> --bitwidth 8 --lowpass 9.6 --resample 8 --lowpass-width 1
>>>>>>
>>>>>> --SATISH BAROT
>>>>>> Ahmedabad,India.
>>>>>>
>>>>>>
>>>>>> On Wed, Jan 25, 2012 at 8:59 PM, Faraj Khasib >>>>> > wrote:
>>>>>>
>>>>>>> Hello Guys,
>>>>>>> I am trying to convert files that are .wac to mp3 after mixmonitor
>>>>>>> command is called but it doesnt execute the command, I tried the 
>>>>>>> command in
>>>>>>> terminal it worked, any help please ... below is my dial plan
>>>>>>> exten=6500,n,Set(MIXMONITOR_EXEC=&& nice -n 19 /usr/local/bin/lame
>>>>>>> -b 8 -t -F -m m --bitwidth 8 --quiet
>>>>>>> "/var/spool/asterisk/monitor/${CALLFILENAME}.wav"
>>>>>>> "/var/spool/asterisk/monitor/${CALLFILENAME}.mp3" && rm -f
>>>>>>> "/var/spool/asterisk/monitor/${CALLFILENAME}.wav")
>>>>>>> exten=6500,n,MixMonitor(${CALLFILENAME}.wav,b)
>>>>>>>
>>>>>>>
>>
> Show your latest dialplan and script.
>
> --Satish Barot
> Ahmedabad, India
>
> --
> _
> -- 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
>
--
_
-- 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

[asterisk-users] Asterisk crash

[Gopalakrishnan N]

Suddenly my asterisk restarted automatically and came up in seven seconds,

While checking core dump I see some message related to snmp.

No symbol table info available.
#5 0x7fc7e6249faa in agent_thread (arg=) at
snmp/agent.c:206
__PRETTY_FUNCTION__ = "agent_thread"
#6 0x0056dd0b in dummy_start (data=) at
utils.c:1028
__cancel_buf = {__cancel_jmp_buf = {{__cancel_jmp_buf = {89647040,
7553562169405615537, 140735377460432, 140496194722240, 4, 7,
-7540143656030687823,
7553561768520461745}, __mask_was_saved = 0}}, __pad = {0x7fc7d1c74e90, 0x0,
0x0, 0x0}}
__cancel_arg = 0x7fc7d1c75700
not_first_call = 
ret = 
a = {start_routine = 0x7fc7e6249eb0 , data = 0x0, name =
0x7fc7d1c74d70 "\300\347W\005"}
#7 0x7fc830e54851 in start_thread () from /lib64/libpthread.so.0
No symbol table info available.
#8 0x7fc8323c611d in clone () from /lib64/libc.so.6
No symbol table info available.
(gdb) quit

Will this be related to snmp?

Regards
--
_
-- 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

Re: [asterisk-users] Asterisk crash

[Gopalakrishnan N]

Ok thanks posting now
On 5 Jul 2013 03:09, "Matthew Jordan"  wrote:

>
> On Thu, Jul 4, 2013 at 3:30 PM, Gopalakrishnan N <
> gopalakrishnan...@gmail.com> wrote:
>
>> Suddenly my asterisk restarted automatically and came up in seven seconds,
>>
>> While checking core dump I see some message related to snmp.
>>
>> No symbol table info available.
>> #5 0x7fc7e6249faa in agent_thread (arg=) at
>> snmp/agent.c:206
>> __PRETTY_FUNCTION__ = "agent_thread"
>> #6 0x0056dd0b in dummy_start (data=) at
>> utils.c:1028
>> __cancel_buf = {__cancel_jmp_buf = {{__cancel_jmp_buf = {89647040,
>> 7553562169405615537, 140735377460432, 140496194722240, 4, 7,
>> -7540143656030687823,
>> 7553561768520461745}, __mask_was_saved = 0}}, __pad = {0x7fc7d1c74e90,
>> 0x0, 0x0, 0x0}}
>> __cancel_arg = 0x7fc7d1c75700
>> not_first_call = 
>> ret = 
>> a = {start_routine = 0x7fc7e6249eb0 , data = 0x0, name =
>> 0x7fc7d1c74d70 "\300\347W\005"}
>> #7 0x7fc830e54851 in start_thread () from /lib64/libpthread.so.0
>> No symbol table info available.
>> #8 0x7fc8323c611d in clone () from /lib64/libc.so.6
>> No symbol table info available.
>> (gdb) quit
>>
>> Will this be related to snmp?
>>
>>
>> Possibly, but not necessarily. Without seeing the whole backtrace it's
> hard to say for certain.
>
> The Asterisk wiki has instructions on how to properly get a backtrace from
> a core dump created by Asterisk:
>
> https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace
>
> Please do file an issue in the issue tracker - https://issues.asterisk.org- 
> crashes are always bugs.
>
> Thanks!
>
> Matt
>
> --
> Matthew Jordan
> Digium, Inc. | Engineering Manager
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at: http://digium.com & http://asterisk.org
>
> --
> _
> -- 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
>
--
_
-- 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

Re: [asterisk-users] Executing Script after MixMonitor is called

[Gopalakrishnan N]

I tried with the ^ symbol but still there is no success.

And regards to the path, actually my file is in path /root, is that to be
in /usr/sbin or somewhere?

Basically am able to see the application executed in the CLI, like the
below,

 Executing [4090@test:1] Set("SIP/4092-003b",
"START_TIME=2013-07-05_14:43:11") in new stack
-- Executing [4090@test:2] Set("SIP/4092-003b",
"MIXMONITOR_FILENAME=4090-2013-07-05_14:43:11-OUT") in new stack
-- Executing [4090@test:3] MixMonitor("SIP/4092-003b",
"IND_PRI/4092/OUT/2013-07/05/4090-2013-07-05_14:43:11-OUT.wav,b,/root/flac.sh
4090-2013-07-05_14:43:11-OUT") in new stack
-- Executing [4090@test:4] Set("SIP/4092-003b",
"CDR(userfield)=/var/spool/asterisk/monitor/IND_PRI/4092/OUT/2013-07/05/4090-2013-07-05_14:43:11-OUT.wav")
in new stack
-- Executing [4090@test:5] Dial("SIP/4092-003b", "SIP/4090,30") in
new stack
  == Using SIP RTP CoS mark 5
  == Begin MixMonitor Recording SIP/4092-003b
-- Called SIP/4090
-- SIP/4090-003c is ringing
-- SIP/4090-003c answered SIP/4092-003b
-- fixed jitterbuffer created on channel SIP/4090-003c
-- fixed jitterbuffer created on channel SIP/4092-003b
-- Executing [h@test:1] MYSQL("SIP/4092-003b", "Connect connid
localhost root Iopex1063 Logs") in new stack
-- Executing [h@test:2] MYSQL("SIP/4092-003b", "Query resultid 1
insert into
call_log(accountcode,start,end,src,dst,uniqueid,userfield,hangupcause)
values("4092","2013-07-05
14:43:11",now(),30993091,4090,1373049791.59,"/var/spool/asterisk/monitor/IND_PRI/4092/OUT/2013-07/05/4090-2013-07-05_14:43:11-OUT.wav",16)")
in new stack
-- Executing [h@test:3] MYSQL("SIP/4092-003b", "Disconnect 1") in
new stack
-- fixed jitterbuffer destroyed on channel SIP/4090-003c
  == Spawn extension (test, 4090, 5) exited non-zero on 'SIP/4092-003b'
-- fixed jitterbuffer destroyed on channel SIP/4092-003b
  == MixMonitor close filestream
  == *Executing [/root/flac.sh 4090-2013-07-05_14:43:11-OUT]*
  == End MixMonitor Recording SIP/4092-003b

But the file is not converted, I suspect it could be a path issue.



Regards


On Fri, Jul 5, 2013 at 10:59 AM, Satish Barot wrote:

> On Fri, Jul 5, 2013 at 1:45 AM, Gopalakrishnan N <
> gopalakrishnan...@gmail.com> wrote:
>
>> exten => _4X.,1,Set(START_TIME=${STRFTIME(${EPOCH},,%Y-%m-%d_%H:%M:%S)})
>> exten => _4X.,n,Set(MIXMONITOR_FILENAME=${EXTEN}-${START_TIME}-OUT)
>> ;exten =>
>> _4X.,n,MixMonitor(IND_PRI/${CDR(accountcode)}/OUT/${STRFTIME(${EPOCH},,%Y-%m)}/${STRFTIME(${EPOCH},,%d)}/${MIXMONITOR_FILENAME}.wav,m,/root/flac.sh
>> ${MIXMONITOR_FILENAME}.wav)
>> exten =>
>> _4X.,n,MixMonitor(IND_PRI/${CDR(accountcode)}/OUT/${STRFTIME(${EPOCH},,%Y-%m)}/${STRFTIME(${EPOCH},,%d)}/${MIXMONITOR_FILENAME}.wav,b,/root/flac.sh
>> ${MIXMONITOR_FILENAME}.wav)
>> exten =>
>> _4X.,n,Set(CDR(userfield)=IND_PRI/${CDR(accountcode)}/OUT/${STRFTIME(${EPOCH},,%Y-%m)}/${STRFTIME(${EPOCH},,%d)}/${MIXMONITOR_FILENAME})
>> exten => _4X.,n,Dial(SIP/${EXTEN},30)
>> exten => _4X.,n,Hangup
>>
>> Regards
>> On 4 Jul 2013 11:18, "Satish Barot"  wrote:
>>
>>> On Thu, Jul 4, 2013 at 1:30 AM, Gopalakrishnan N <
>>> gopalakrishnan...@gmail.com> wrote:
>>>
>>>> I tried with hangup cause but my script is not executed... also I tried
>>>> the same script with mix monitor itself no sucess.
>>>>
>>>> The script what I have is, am converting wav file to flac format..
>>>> On 11 Jun 2013 11:17, "Satish Barot"  wrote:
>>>>
>>>>> And yes if you want to use System application in your dialplan then
>>>>> have System in your h extension
>>>>>
>>>>> System(/PathToSox/sox -r 8000 -c 1 /PathToWavFile/filename.wav 
>>>>> /PathToMp3FileToBE Stored/filename.mp3)
>>>>>
>>>>>
>>>>>
>>>>> On Tue, Jun 11, 2013 at 10:38 AM, Satish Barot <
>>>>> satish4aster...@gmail.com> wrote:
>>>>>
>>>>>> Hi Gopamkrishnan,
>>>>>>
>>>>>> Check the 'command' argument for Mixmonitor. Mixmonitor itself has a
>>>>>> facility to execute a command when recording is over.
>>>>>>
>>>>>> *In my case, 'wav2mp3' is a script which gets executed and converts 
>>>>>> recorded wav audio file to mp3. I pass ${FILENAME} as an argument to my 
>

Re: [asterisk-users] External Recording Server for Asterisk Voicemail

[Gopalakrishnan N]

If you want to store in external, why can't you have a NAS device and mount
to Asterisk server, let the mounted be a part in asterisk.conf, so that
voicemail will get recorded in external server...

Will it makes sense... !

Thanks.


On Mon, Jul 15, 2013 at 4:19 PM, Amit Salunkhe wrote:

> Hello All,
>
> I'm planning to use Asterisk only for voicemail Application and Recording
> will be done at different server.
>
> When user changing his personal greeting or leaving voicemail Call need to
> throw to external Voicemnail recording server over SIP til the time
> recording complete.
>
> While throwing Cal from Asterisk to application box i have to use SIP
> request which having some string in R-URI. Please let me know is this
> possible with configuration example.
>
>
>
> Regards
> Amit
>
> --
> _
> -- 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
>
--
_
-- 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

[asterisk-users] FLAC script to convert from wav to FLAC and also with other 3 to 4 formats

[Gopalakrishnan N]

Hi,

Below link is the script which i found while surfing, this script basically
converts your voice file to flac format, where the file is reduced to 50%.

http://legroom.net/files/software/convtoflac.sh

The quality is really good, I tested. this...

In large production environment this script can be used, only challenging
part, please make sure the CPU usage is within the limit while conversion.

Can be used like this,
exten =>
_4X.,n,MixMonitor(IND_PRI/${CDR(accountcode)}/OUT/${STRFTIME(${EPOCH},,%Y-%m)}/${STRFTIME(${EPOCH},,%d)}/${MIXMONITOR_FILENAME}.wav,b,/root/
flac.sh ${MIXMONITOR_FILENAME}.wav)

Regards,
Gopal.
--
_
-- 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

Re: [asterisk-users] auto answer

[Gopalakrishnan N]

yes its not asterisk configuration, its phone feature and phone
configuration.


On Wed, Jul 17, 2013 at 3:27 PM, bilal ghayyad  wrote:

> So it is not at asterisk configuration?
>
> Regards
> Bilal
>
>   --
>  *From:* A J Stiles 
>
> *To:* bilal ghayyad ; Asterisk Users Mailing List -
> Non-Commercial Discussion 
> *Sent:* Wednesday, July 17, 2013 12:57 PM
>
> *Subject:* Re: [asterisk-users] auto answer
>
> On Wednesday 17 July 2013, bilal ghayyad wrote:
> > But this not in the sip.conf, this in the extensions.conf, right?
> >
> > Regards
> > Bilal
>
> No.  This would be set up in the phone's own configuration file, which in
> turn
> depends on the make and model of phone  (and its location depends on your
> site
> setup).
>
> --
> AJS
>
> Answers come *after* questions.
>
>
>
> --
> _
> -- 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
>
--
_
-- 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

Re: [asterisk-users] auto answer

[Gopalakrishnan N]

If am not wrong even without doing any setting in asterisk side, if the
phone has Auto Answer it works.. !

Correct me if am wrong.


On Wed, Jul 17, 2013 at 9:14 PM, Steve Edwards wrote:

> Please don't top post.
>
>
> On Wed, 17 Jul 2013, bilal ghayyad wrote:
>
>  So it is not at asterisk configuration?
>>
>
> 1) The phone has to be configured to allow it.
>
> 2) Asterisk has to set the appropriate SIP header for your specific model
> phone prior to 'dialing' the phone for each call. I.e. the added SIP header
> for a Cisco is different than for a Polycom.
>
> --
> Thanks in advance,
> --**--**
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
> Newline  Fax: +1-760-731-3000
>
>
> --
> __**__**_
> -- 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
>
--
_
-- 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

[asterisk-users] Random dead calls

[Gopalakrishnan N]

Hi,

Am getting dead or silence calls at sometimes for my agents, when I checked
my CDR the caller-id shows my vendor's name and some shows as real customer
name.

When I call back again the real customer's number its reaching, the
answering machine owned by customer.

I have a confusion, or how to find out are these numbers are from any auto
dialer or from real customers.

Thanks.
--
_
-- 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

[asterisk-users] Asterisk SIP Trunk between two Asterisk Servers

[Gopalakrishnan N]

Hi,

Am making a simple SIP trunk between two Asterisk server,

Server 1
sip.conf
[usman02]
type=peer
username=usman02
secret=usman02
host=10.30.2.58
context=man02-trunk
port=5060
qualify=yes
disallow=all
;allow=g729
allow=g729
;allow=alaw
nat=force_rport,comedia
dtmfmode=rfc2833
relaxdtmf=yes
insecure=invite,port

extensions.conf
[man02-trunk]
exten => _1X.,1,Dial(SIP/usman02/${EXTEN})
exten => _1X.,n,Hangup


Server2
sip.conf
[usman02]
type=peer
username=usman02
secret=usman02
host=10.10.10.81
context=us02-trunk-inbound
port=5060
qualify=yes
disallow=all
allow=g729
;allow=ulaw
;allow=alaw
nat=force_rport,comedia
dtmfmode=rfc2833
relaxdtmf=yes
insecure=port,invite

extensions.conf
[us02-trunk-inbound]
exten => _X.,Dial(SIP/${EXTEN},60)


Now when I dial from server1, in the server 2 am getting the error as,
[Aug 18 09:22:49] WARNING[2779][C-08db]: chan_sip.c:16266 check_auth:
username mismatch, have <2001>, digest has 

things are fine.. but I dont know where the mistake is...!

Can you some one advise me... !

Thanks.
--
_
-- 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

Re: [asterisk-users] Asterisk SIP Trunk between two Asterisk Servers

[Gopalakrishnan N]

Even I tried the type as friend.. but no use...


On Mon, Aug 19, 2013 at 12:27 AM, Gopalakrishnan N <
gopalakrishnan...@gmail.com> wrote:

> Hi,
>
> Am making a simple SIP trunk between two Asterisk server,
>
> Server 1
> sip.conf
> [usman02]
> type=peer
> username=usman02
> secret=usman02
> host=10.30.2.58
> context=man02-trunk
> port=5060
> qualify=yes
> disallow=all
> ;allow=g729
> allow=g729
> ;allow=alaw
> nat=force_rport,comedia
> dtmfmode=rfc2833
> relaxdtmf=yes
> insecure=invite,port
>
> extensions.conf
> [man02-trunk]
> exten => _1X.,1,Dial(SIP/usman02/${EXTEN})
> exten => _1X.,n,Hangup
>
>
> Server2
> sip.conf
> [usman02]
> type=peer
> username=usman02
> secret=usman02
> host=10.10.10.81
> context=us02-trunk-inbound
> port=5060
> qualify=yes
> disallow=all
> allow=g729
> ;allow=ulaw
> ;allow=alaw
> nat=force_rport,comedia
> dtmfmode=rfc2833
> relaxdtmf=yes
> insecure=port,invite
>
> extensions.conf
> [us02-trunk-inbound]
> exten => _X.,Dial(SIP/${EXTEN},60)
>
>
> Now when I dial from server1, in the server 2 am getting the error as,
> [Aug 18 09:22:49] WARNING[2779][C-08db]: chan_sip.c:16266 check_auth:
> username mismatch, have <2001>, digest has 
>
> things are fine.. but I dont know where the mistake is...!
>
> Can you some one advise me... !
>
> Thanks.
>
--
_
-- 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

Re: [asterisk-users] Asterisk SIP Trunk between two Asterisk Servers

[Gopalakrishnan N]

Thanks for the comments.

Without changing anything, adding fromuser=usman02 in both side worked for
me..

Thanks.


On Mon, Aug 19, 2013 at 1:01 AM, Andrew Colin  wrote:

>  change server two to host = dynamic
>
> then add register = on server 1
>
> On 8/18/2013 6:29 PM, Gopalakrishnan N wrote:
>
> Even I tried the type as friend.. but no use...
>
>
> On Mon, Aug 19, 2013 at 12:27 AM, Gopalakrishnan N <
> gopalakrishnan...@gmail.com> wrote:
>
>> Hi,
>>
>>  Am making a simple SIP trunk between two Asterisk server,
>>
>>  Server 1
>> sip.conf
>>  [usman02]
>> type=peer
>> username=usman02
>> secret=usman02
>> host=10.30.2.58
>> context=man02-trunk
>> port=5060
>> qualify=yes
>> disallow=all
>> ;allow=g729
>> allow=g729
>> ;allow=alaw
>> nat=force_rport,comedia
>> dtmfmode=rfc2833
>> relaxdtmf=yes
>> insecure=invite,port
>>
>>  extensions.conf
>>  [man02-trunk]
>> exten => _1X.,1,Dial(SIP/usman02/${EXTEN})
>> exten => _1X.,n,Hangup
>>
>>
>>  Server2
>> sip.conf
>>  [usman02]
>> type=peer
>> username=usman02
>> secret=usman02
>> host=10.10.10.81
>> context=us02-trunk-inbound
>> port=5060
>> qualify=yes
>> disallow=all
>> allow=g729
>> ;allow=ulaw
>> ;allow=alaw
>> nat=force_rport,comedia
>> dtmfmode=rfc2833
>> relaxdtmf=yes
>>  insecure=port,invite
>>
>>  extensions.conf
>>  [us02-trunk-inbound]
>> exten => _X.,Dial(SIP/${EXTEN},60)
>>
>>
>>  Now when I dial from server1, in the server 2 am getting the error as,
>> [Aug 18 09:22:49] WARNING[2779][C-08db]: chan_sip.c:16266 check_auth:
>> username mismatch, have <2001>, digest has 
>>
>>  things are fine.. but I dont know where the mistake is...!
>>
>>  Can you some one advise me... !
>>
>>  Thanks.
>>
>
>
>
> --
> _
> -- 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
>
>
>
--
_
-- 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

Re: [asterisk-users] Asterisk SIP Trunk between two Asterisk Servers

[Gopalakrishnan N]

ok thanks Asghar Mohammad


On Mon, Aug 19, 2013 at 1:05 AM, Asghar Mohammad wrote:

> just remove username.
> type peer authenticate by ip
>
>
> On Sun, Aug 18, 2013 at 7:01 PM, Andrew Colin  wrote:
>
>>  change server two to host = dynamic
>>
>> then add register = on server 1
>>
>> On 8/18/2013 6:29 PM, Gopalakrishnan N wrote:
>>
>> Even I tried the type as friend.. but no use...
>>
>>
>> On Mon, Aug 19, 2013 at 12:27 AM, Gopalakrishnan N <
>> gopalakrishnan...@gmail.com> wrote:
>>
>>> Hi,
>>>
>>>  Am making a simple SIP trunk between two Asterisk server,
>>>
>>>  Server 1
>>> sip.conf
>>>  [usman02]
>>> type=peer
>>> username=usman02
>>> secret=usman02
>>> host=10.30.2.58
>>> context=man02-trunk
>>> port=5060
>>> qualify=yes
>>> disallow=all
>>> ;allow=g729
>>> allow=g729
>>> ;allow=alaw
>>> nat=force_rport,comedia
>>> dtmfmode=rfc2833
>>> relaxdtmf=yes
>>> insecure=invite,port
>>>
>>>  extensions.conf
>>>  [man02-trunk]
>>> exten => _1X.,1,Dial(SIP/usman02/${EXTEN})
>>> exten => _1X.,n,Hangup
>>>
>>>
>>>  Server2
>>> sip.conf
>>>  [usman02]
>>> type=peer
>>> username=usman02
>>> secret=usman02
>>> host=10.10.10.81
>>> context=us02-trunk-inbound
>>> port=5060
>>> qualify=yes
>>> disallow=all
>>> allow=g729
>>> ;allow=ulaw
>>> ;allow=alaw
>>> nat=force_rport,comedia
>>> dtmfmode=rfc2833
>>> relaxdtmf=yes
>>>  insecure=port,invite
>>>
>>>  extensions.conf
>>>  [us02-trunk-inbound]
>>> exten => _X.,Dial(SIP/${EXTEN},60)
>>>
>>>
>>>  Now when I dial from server1, in the server 2 am getting the error as,
>>> [Aug 18 09:22:49] WARNING[2779][C-08db]: chan_sip.c:16266
>>> check_auth: username mismatch, have <2001>, digest has 
>>>
>>>  things are fine.. but I dont know where the mistake is...!
>>>
>>>  Can you some one advise me... !
>>>
>>>  Thanks.
>>>
>>
>>
>>
>> --
>> _
>> -- 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
>>
>>
>>
>> --
>> _
>> -- 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
>>
>
>
> --
> _
> -- 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
>
--
_
-- 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

[asterisk-users] Ingress and Egress

[Gopalakrishnan N]

Hi,

Can Ingress and Egress can be used in Asterisk, so that Jitter can be
calculated...!

Regards
--
_
-- 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

Re: [asterisk-users] Ingress and Egress

[Gopalakrishnan N]

Basically I have some background noise like keyboard stoke or clicking
sound in random basis, I need to measure that, when I check my IPLC its
fine, and with my Telco service provider its fine...

So am trying to conclude with some solution... trying to identify the root
cause.

Any advice would be appreciated.

Thanks.


On Wed, Aug 21, 2013 at 4:46 PM, jg  wrote:

> You do not need to calculate the jitter values yourself. For a quick check
> you can use the CLI cmd "sip show channelstats". For external monitoring
> you could capture the RTCP AMI events.
>
> jg
>
> --
> __**__**_
> -- 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
>
--
_
-- 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

Re: [asterisk-users] How to get the original SIP result code

[Gopalakrishnan N]

You can use AMI Commands and run sip set debug from that you have to
capture the response code.

http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Command

Regards,


On Thu, Aug 22, 2013 at 10:43 PM, Mordechay Kaganer wrote:

> B.H.
>
> Hello, i'm using AMI Originate action (with async=true) to send outgoing
> calls to a SIP trunk (using asterisk-java library to connect to AMI).
>
> The problem is that in case of failed originate, OriginateResponse event
> is returning only the reason code which is sometimes not sufficient to
> determine the real cause of failure. Also, there's no way to link between
> the channel that was trying to dial and failed and the original Originate
> request, because OriginateResponse is issued only after the failed channel
> was hang up. Only successful OriginateResponse will contain the unique id
> of the established channel.
>
> Is there any way that my AMI application can get the original SIP response
> of the failed Originate action?
>
> I'm using Asterisk 1.8.22 and slightly tweaked asterisk-java (
> https://blogs.reucon.com/asterisk-java/) 1.0.0.
>
>
> --
> כתיבה וחתימה טובה לשנה טובה ומתוקה בגשמיות וברוחניות!
> משיח NOW!
> Moshiach is coming very soon, prepare yourself!
> יחי אדוננו מורינו ורבינו מלך המשיח לעולם ועד!
>
> --
> _
> -- 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
>
--
_
-- 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

[asterisk-users] Kepress while on Queue

[Gopalakrishnan N]

Hi,

Will Keypress option will work when am in the queue and hearing MoH?

Lets say a caller is waiting in queue and while he is hearing MoH, can he
key in some DTMF and go to some other queue? is that possible?

Regards
--
_
-- 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

Re: [asterisk-users] Kepress while on Queue

[Gopalakrishnan N]

oh great thanks...


On Wed, Aug 28, 2013 at 1:12 PM, Satish Barot wrote:

> Yes you can. Check the 'context' parameter in queues.conf. When caller
> presses a single digit extension while waiting in a queue, (s)he'll be
> taken out of queue to this context. Then you can send caller to different
> queue from this context.
>
> --Satish Barot
> Ahmedabad, India.
> +919978599700
>
>
> On Wed, Aug 28, 2013 at 12:59 AM, Gopalakrishnan N <
> gopalakrishnan...@gmail.com> wrote:
>
>> Hi,
>>
>> Will Keypress option will work when am in the queue and hearing MoH?
>>
>> Lets say a caller is waiting in queue and while he is hearing MoH, can he
>> key in some DTMF and go to some other queue? is that possible?
>>
>> Regards
>>
>> --
>> _
>> -- 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
>>
>
>
> --
> _
> -- 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
>
--
_
-- 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

Re: [asterisk-users] Kepress while on Queue

[Gopalakrishnan N]

also if am not wrong queue timeout will also applicable for this.. !


On Wed, Aug 28, 2013 at 11:37 PM, Gopalakrishnan N <
gopalakrishnan...@gmail.com> wrote:

> oh great thanks...
>
>
> On Wed, Aug 28, 2013 at 1:12 PM, Satish Barot 
> wrote:
>
>> Yes you can. Check the 'context' parameter in queues.conf. When caller
>> presses a single digit extension while waiting in a queue, (s)he'll be
>> taken out of queue to this context. Then you can send caller to
>> different queue from this context.
>>
>> --Satish Barot
>> Ahmedabad, India.
>> +919978599700
>>
>>
>> On Wed, Aug 28, 2013 at 12:59 AM, Gopalakrishnan N <
>> gopalakrishnan...@gmail.com> wrote:
>>
>>> Hi,
>>>
>>> Will Keypress option will work when am in the queue and hearing MoH?
>>>
>>> Lets say a caller is waiting in queue and while he is hearing MoH, can
>>> he key in some DTMF and go to some other queue? is that possible?
>>>
>>> Regards
>>>
>>> --
>>> _
>>> -- 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
>>>
>>
>>
>> --
>> _
>> -- 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
>>
>
>
--
_
-- 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

[asterisk-users] G729 CPU Utilization

[Gopalakrishnan N]

Hi,

How much CPU utilization will it take when I use G729 transcoding via
hardware based transcoder.

Regards
--
_
-- 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

[asterisk-users] Bad Magic Internal Error

[Gopalakrishnan N]

What does this mean of bad magic internal error, SIP to SIP calling is
fine, when I use SIP via GSM I have this, and asterisk restarts
automatically. Asterisk version which am using is 11.1.2.



Regards
--
_
-- 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

Re: [asterisk-users] Multi-Voicemail Message?

[Gopalakrishnan N]

You can have something like this,
exten => _,1, Answer
exten => _, 2, voicemail ($EXTEN)
 On 25 Sep 2013 05:04, "Tim Nelson"  wrote:

> Greetings-
>
> I have an odd scenario where I need to dial an extension (lets call it
> 555), the system prompts for a list of voicemail boxes, then once complete,
> allows the caller to leave a voicemail that is sent to all voicemail boxes
> previously specified.
>
> How would you do this? Obviously calling Voicemail(), but how to get input
> for multiple extensions/voicemails, and delimit them properly for passing
> to Voicemail()?
>
> All ideas welcome. Thanks!
>
> --Tim
>
> --
> _
> -- 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
>
-- 
_
-- 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

[asterisk-users] Channel not releasing immediately for Attended Transfer

[Gopalakrishnan N]

I have a situation where Asterisk is not releasing the channel for Attended
transfer immediately once I transferred and hangup from my side. The call
is still ongoing and disconnecting after the third party disconnected.

I see that its bug in the Asterisk, but not sure its fixed in version
11.2.1.

Any one facing this issue?

Regards.
-- 
_
-- 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

Re: [asterisk-users] Sangoma transcoding card bug - drops audio samples

[Gopalakrishnan N]

If you are getting like this dropped packets then nothing to worry.. thisis
just an cli message in my case I face this but there is no voice delay
in actual call.
On 22 Nov 2013 21:11, "Eric Wieling"  wrote:

> Are you getting errors like this?
>
>
>
> [Nov 22 10:39:36] WARNING[6307][C-09a1]: codec_sangoma.c:969
> sangoma_frameout: [2724][ulawtog729] Got Seq 7400 but expecting 2154 (time
> since last read = 0ms), dropped 5246 packets
>
>
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Grzegorz Garlewicz
> *Sent:* Friday, November 22, 2013 2:55 AM
> *To:* asterisk-users@lists.digium.com
> *Subject:* [asterisk-users] Sangoma transcoding card bug - drops audio
> samples
>
>
>
>
> There is a serious bug in Sangoma transcoding cards. The card has an
> internal, small jitter buffer and it drops samples
>
> from the audio stream when there is high jitter in the network. The
> bandwidth is cheap now so for me the only reason
>
> to use transcoding is where I have low-bandwidth-high-jitter links.
> Sangoma said they will not fix it and we had to go back
>
> to software transconding.
>
>
> Do you have any experience with using Digium cards in such scenario?
>
> --
> _
> -- 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
>
-- 
_
-- 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

[asterisk-users] AGI Script not working

[Gopalakrishnan N]

I have a Perl AGI script updating some values to database like recorded
file path, unique ID and callerid. When I run the script with test
dialplan, its not updating to database.

Whereas database connection is fine, when I run agi debug I see only Tx
packets not Rx packets, firewall is also OFF.

Any other specific reason why there is no Rx.

The same script working in one more Asterisk machine.

Regards
-- 
_
-- 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

Re: [asterisk-users] Answering agent

[Gopalakrishnan N]

Alao enable cel table that will have all the information
On 29 Nov 2013 23:25, "Todd R."  wrote:

> I do this by writing custom CDR. I write the agents extension write into
> the CDR records. This makes is easy to just parse through the CDR and get
> all the info you need about the call.
>
> Google something like "asterisk custom CDR"
>
>
>
>
> > On Nov 29, 2013, at 11:36 AM, "Leandro Dardini" 
> wrote:
> >
> > Hello friends,
> > when a call arrives in the queue, a CDR record is created, but there is
> no info about which agent has picked up the call. I can find that info only
> in queue_log.
> >
> > Is there a way to have that info in the CDR or maybe in a variable in
> the "h" context, when the call is ended?
> >
> > Leandro
> > --
> > _
> > -- 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
>
> --
> _
> -- 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
>
-- 
_
-- 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

Re: [asterisk-users] AGI Script not working

[Gopalakrishnan N]

Library is Asterisk Perl library and module DBI. The same script working in
different machine with same Asterisk version and same Perl version. Am able
to see Tx and Rx from script.




On Mon, Dec 2, 2013 at 8:08 AM, Eric Wieling  wrote:

> Sounds like you are violating the AGI protocol.   Which Perl AGI library
> are you using?
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of Gopalakrishnan N
> Sent: Saturday, November 30, 2013 1:27 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] AGI Script not working
>
> I have a Perl AGI script updating some values to database like recorded
> file path, unique ID and callerid. When I run the script with test
> dialplan, its not updating to database.
>
> Whereas database connection is fine, when I run agi debug I see only Tx
> packets not Rx packets, firewall is also OFF.
>
> Any other specific reason why there is no Rx.
>
> The same script working in one more Asterisk machine.
>
> Regards
>
> --
> _
> -- 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
>
-- 
_
-- 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

Re: [asterisk-users] AGI Script not working

[Gopalakrishnan N]

Thanks... I got it working actually I found with this command /usr/bin/perl
-d  from this I got to know that my library is missing and
installed Asterisk-perl module and now its fine.

Once again thank you.


On Mon, Dec 2, 2013 at 3:05 PM, Gopalakrishnan N <
gopalakrishnan...@gmail.com> wrote:

> Library is Asterisk Perl library and module DBI. The same script working
> in different machine with same Asterisk version and same Perl version. Am
> able to see Tx and Rx from script.
>
>
>
>
> On Mon, Dec 2, 2013 at 8:08 AM, Eric Wieling  wrote:
>
>> Sounds like you are violating the AGI protocol.   Which Perl AGI library
>> are you using?
>>
>>
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] On Behalf Of Gopalakrishnan N
>> Sent: Saturday, November 30, 2013 1:27 AM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: [asterisk-users] AGI Script not working
>>
>> I have a Perl AGI script updating some values to database like recorded
>> file path, unique ID and callerid. When I run the script with test
>> dialplan, its not updating to database.
>>
>> Whereas database connection is fine, when I run agi debug I see only Tx
>> packets not Rx packets, firewall is also OFF.
>>
>> Any other specific reason why there is no Rx.
>>
>> The same script working in one more Asterisk machine.
>>
>> Regards
>>
>> --
>> _
>> -- 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
>>
>
>
-- 
_
-- 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

Re: [asterisk-users] Change the preferred audio playback format

[Gopalakrishnan N]

Hope basically depends on the codec Asterisk will playback the file
automatically
On 23 Jan 2014 19:25, "Gareth Blades" 
wrote:

> On 23/01/14 13:38, Ishfaq Malik wrote:
>
>> Hi
>>
>> Is there any way to change the preferred audio playback format in
>> asterisk (I'm using 1.8.25.0)
>> i.e. first check for gsm, if doesn't exits then check for slin?
>>
>
> It should pick whichever source format requires the least cpu to transcode
> into the desired output format.
> So generally that means if there is a source available in the same format
> as the output then it will use it otherwise it will use slin 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
>
-- 
_
-- 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

Re: [asterisk-users] Integration of OpenVXI

[Gopalakrishnan N]

Anyone using Voiceglue with latest Asterisk 11.6 certified version?


On Mon, Jun 20, 2011 at 10:00 PM, Jean-Denis Girard wrote:

> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Hi,
>
> Le 20/06/2011 04:40, Gopal krishnan a écrit :
> > Have anybody integrated
> > OpenVXI http://www.speech.cs.cmu.edu/openvxi/ with Asterisk?
>
> Voiceglue works for me: http://www.voiceglue.org/
>
>
> Thanks,
> - --
> Jean-Denis Girard
>
> SysNux  Systèmes  Linux  en Polynésie française
> http://www.sysnux.pf/   Tél: +689 50 10 40 / GSM: +689 79 75 27
> -BEGIN PGP SIGNATURE-
>
> iEYEARECAAYFAk3/dbgACgkQuu7Rv+oOo/hemACdEN4qLhxLl9LJGpdGIfd8zZ0B
> PAsAnRxitrzwt5RhWPeo/iwVuYqfeKNh
> =LpwD
> -END PGP SIGNATURE-
>
> --
> _
> -- 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
>
-- 
_
-- 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

[asterisk-users] Voice XML Asterisk Integration

[Gopalakrishnan N]

Which is the best way around to integrate Asterisk with VoiceXML like
VoiceGlue...! Am using Asterisk 11.2.1.

Regards.
-- 
_
-- 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

Re: [asterisk-users] SIP OPTIONS "storm"?

[Gopalakrishnan N]

SIP options message is due to check the peer registration is keepalive. As
per my understanding it might be because of network flap may be wireshark
trace can give you any clue.

Regards
On 13 Feb 2014 23:41, "Tim Nelson"  wrote:

> Greetings-
>
> I recently experienced an odd situation. I have an Asterisk 11.5.0 system
> (Box A) with a SIP peering to another Asterisk 1.8.23.0 system (Box B). At
> some point, Box A started sending over 65Mbps of SIP OPTIONS packets to Box
> A. I do have qualify=yes for the peer on both sides, and the qualifyfreq is
> not set (aka default of 60secs).
>
> Of course, logs on Box A were not set to show debug info, so there is no
> indication of a problem. Logs on Box B show no issues, only at a very
> specific start time, there are suddenly tons of:
>
> [2014-02-13 00:12:50] DEBUG[31516] chan_sip.c: Allocating new SIP dialog
> for 2a338cf5518531e31190bd4b7826d137@x.y.z.166:5060 - OPTIONS (No RTP)
>
> I've done quite a bit of searching, but am not finding anything of
> consequence. Also, the Asterisk changelogs are not providing anything that
> would indicate this is known and fixed, at least for the 11.x branch.
>
> Thoughts/suggestions? Thanks!
>
> --Tim
>
> --
> _
> -- 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
>
-- 
_
-- 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

Re: [asterisk-users] Asterisk Crash in asterisk 10.0.0.

[Gopalakrishnan N]

Enable debugging module and backtrace and re-compile so that you will
bactrace of the crash logs.

Regards
On 14 Feb 2014 10:29, "Arun Ram"  wrote:

> Hi guys,
>  I need a desperate help from you regarding this asterisk crash issue.
>
>
>
> On Thu, Feb 13, 2014 at 5:48 PM, Arun Ram  wrote:
>
>> Hi,
>>
>> I  am facing asterisk crash issue  in my  Asterisk 10.0.0. safe 
>> asteriskgenerated a core dump in  /tmp path . I  viewed the core dump using
>> viewcore in linux.
>>
>>  *can anyone tell the reason for the crash .  waiting eagerly for an
>> answer from asterisk support guys*.* please the find the core dump
>> attachment too* ..
>>
>>
>> *Below is the information in core dump *
>>
>> --
>>
>>
>> *Thanks & RegardsArunram.c*
>>
>>
>> *The Power of someone has the power to do something.. anything !!*
>>
>
>
>
> --
>
>
> *Thanks & RegardsArunram.c*
>
>
> *The Power of someone has the power to do something.. anything !!*
>
> --
> _
> -- 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
>
-- 
_
-- 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