[asterisk-users] Please help test the gender detection module at 575-613-4392

[Asterisk Asterisk]

I need your help: please help test the gender detection module at 575-613-4392.

I wrote a gender detection module and thought I'd try it out. It only takes a 
second. I've been showing 90%+ accuracy and I want
to make sure it's working correctly. Rain and significant background noise 
seems to throw it off, so I still have a bit of work to do.

Have your friends and significant others call too. Also, let me know if you 
have any need for the module.

Justin Newman
nt_jnewman at yahoo.com


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Re: [asterisk-users] Please help test the gender detection module at 575-613-4392

[Asterisk Asterisk]

This module detects gender and approximate age range. I'm working on getting 
it's accuracy to 80%+ on a consistent basis, after implementing filters to 
remove background noise and other artifacts.

It's designed for a number of things. To start, I have several clients 
(primarily mobile content and servers providers) that want to profile and 
generate demographics of their users for selling advertising. They also want to 
understand their user base. Plus, some customers have found that male and 
female users tend to respond differently to different prompts, flows, etc. This 
helps in designing a system that meets needs of many different types of users.

Of course, there are many other uses and I'm sure people can generate some cool 
ideas.

Let me know how it works when you try the test number at 575-613-4392. Also, 
let me know if you have any interest in the module.

Justin
nt_jnewman at yahoo.com





From: Ron Joffe 
To: asterisk-users@lists.digium.com
Cc: Asterisk Asterisk 
Sent: Monday, February 16, 2009 11:05:24 AM
Subject: Re: [asterisk-users] Please help test the gender detection module at 
575-613-4392

That's an interesting module.

Care to elaborate on what you designed it for ?

Thanks,

Ron



On Monday 16 February 2009 13:29, Asterisk Asterisk wrote:
> I need your help: please help test the gender detection module at
> 575-613-4392.
>
> I wrote a gender detection module and thought I'd try it out. It only takes
> a second. I've been showing 90%+ accuracy and I want to make sure it's
> working correctly. Rain and significant background noise seems to throw it
> off, so I still have a bit of work to do.
>
> Have your friends and significant others call too. Also, let me know if you
> have any need for the module.
>
> Justin Newman
> nt_jnewman at yahoo.com



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Re: [asterisk-users] Please help test the gender detection module at 575-613-4392

[Asterisk Asterisk]

That's funny. The way I have it phrased, when I called I started talking to it 
as well! I have some code for short list voice recognition and thought about 
detecting yes and no in there, but I ran out of time...and the prompts were 
already recorded.

Thank you everyone for helping test the module. There have been 200+ calls from 
users on the list and they are still coming in. We're getting about 65%-70% 
success rate. My target is 80%-85% in random sampling and 90%-95% in controlled 
settings.

Update: I'm adjusting the detection ratios tomorrow, so that should improve 
general detection results based on the received data. I'm implementing filters 
to remove the background noise. I'd guess that 5% of those testing are trying 
to fool the system for fun, in one way or another. When the user is unaware of 
sampling, the results are slightly higher. My greeting suggests a less 
masculine phrase, but with a male voice. I suspect this throws off both 
genders' recordings. I probably should have had testers say their own names, 
since testers rarely divert on that.





From: Gondar Monn 
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Sent: Monday, February 16, 2009 9:19:20 PM
Subject: Re: [asterisk-users] Please help test the gender detection module at 
575-613-4392

Looks like my provider is not passing dtmf correctly .. Had a serious 
laugh, system kept asking me if I was ready., ended up finding myself 
talking to the IVR .


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Re: [asterisk-users] Network architecture

[Asterisk Asterisk]

>found out that the best solution is to use OpenSips as SIP

OpenSIPS is a great free software proxy.

>1- Is there any Software limitation on asterisk regarding number of 
>simulltaneous calls?

There isn't any explicit limitation in Asterisk or OpenSIPS that I'm aware of, 
but you are limited to processing power, memory, bandwidth, etc.

>2- Can 1 asterisk server handle 5000 simuitaneous calls if I have the 
>appropriate hardware?

There are a lot of factors to consider, but I'm sure you could do it if you are 
determined. Not the wisest option however - see below.

>3- It's etter to have one asterisk server for hadling 5k simultaneous calls or 
>divide the load on different servers?

I would split it up and keep each server under 50% load during normal activity. 
That way you can handle peak load and balance if one or more servers fail. Try 
not to put more than 200-400 calls on each server, depending on your 
configuration. That would be 100-200 calls per server with 50% load.

For 5,000 concurrent calls, that means 25 servers assuming decent hardware and 
50% load. That might not be an option. You may be able to split up some of the 
servers into multiple VMs -- maybe five servers with five VMs each. 

You may be able to get away with 90% regular load if 5,000 concurrent calls is 
never to be exceeded. Anyway, there are many factors to consider. More 
information is definitely needed.




From: michel freiha 
To: Asterisk Users Mailing List - Non-Commercial Discussion 
; asterisk-users-boun...@lists.digium.com
Sent: Tuesday, February 17, 2009 7:19:58 AM
Subject: [asterisk-users] Network architecture


Hi all,

I'm planning to build a VOIP solution for handling SIP calls coming from 
endpoints registered on a specific SIP proxy...I made some research regarding 
network architecture and found out that the best solution is to use OpenSips as 
SIP proxy for registration and local calls between registered endpoints and use 
asterisk server with a2billing for PSTN calls, rating, routing and all other 
stuff plus a MySQL database...

This architecture convinced me, but I have some questions regarding asterisk 
and I need asterisk expert answers in order to take decision...

1- Is there any Software limitation on asterisk regarding number of 
simulltaneous calls? 
2- Can 1 asterisk server handle 5000 simuitaneous calls if I have the 
appropriate hardware?
3- It's etter to have one asterisk server for hadling 5k simultaneous calls or 
divide the load on different servers?


Waiting your reply

Regards



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Re: [asterisk-users] Please help test the gender detection module at 575-613-4392

[Asterisk Asterisk]

For those who testing the gender detection module via the number provided:

How was the experience, aside from the funny beep?

In your perception, how well did it perform? (I see raw numbers here, but 
perception is important too.)

Do you have any comments, suggestions, or feedback?





From: Asterisk Asterisk 
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Cc: Gondar Monn ; nt_aster...@yahoo.com; 
nt_jnew...@yahoo.com
Sent: Tuesday, February 17, 2009 9:10:38 AM
Subject: Re: [asterisk-users] Please help test the gender detection module at 
575-613-4392


That's funny. The way I have it phrased, when I called I started talking to it 
as well! I have some code for short list voice recognition and thought about 
detecting yes and no in there, but I ran out of time...and the prompts were 
already recorded.

Thank you everyone for helping test the module. There have been 200+ calls from 
users on the list and they are still coming in. We're getting about 65%-70% 
success rate. My target is 80%-85% in random sampling and 90%-95% in controlled 
settings.

Update: I'm adjusting the detection ratios tomorrow, so that should improve 
general detection results based on the received data. I'm implementing filters 
to remove the background noise. I'd guess that 5% of those testing are trying 
to fool the system for fun, in one way or another. When the user is unaware of 
sampling, the results are slightly higher. My greeting suggests a less 
masculine phrase, but with a male voice. I suspect this throws off both 
genders' recordings. I probably should have had testers say their own names, 
since testers rarely divert on that.





From: Gondar Monn 
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Sent: Monday, February 16, 2009 9:19:20 PM
Subject: Re: [asterisk-users] Please help test the gender detection module at 
575-613-4392

Looks like my provider is not passing dtmf correctly .. Had a serious 
laugh, system kept asking me if I was ready., ended up finding myself 
talking to the IVR .


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Re: [asterisk-users] Please help test the gender detection moduleat 575-613-4392

[Asterisk Asterisk]

Accuracy should be 10%-15% better on Wed or Thu.





From: Jason Aarons (US) 
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Sent: Tuesday, February 17, 2009 10:48:07 AM
Subject: Re: [asterisk-users] Please help test the gender detection moduleat 
575-613-4392

 
After helping out it seems I’ve been determined a female(wrongly). 
It was disappointing and I’m considering a visit to the Dr Phil Show to
work out my anger….



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[asterisk-users] Updated modules to be released (FaxDetect, GenderDetect, MachineDetect, others)

[Asterisk Asterisk]

I will be releasing updated versions to many of the detection modules next 
week. They include better support of Asterisk 1.2, 1.4, and 1.6, better 
detection, better parameters, an easier build system, and usability is enhanced.

The updated modules include:

* FaxDetect, LineDetect, and MachineDetect - which many are presently using
* PlayDetect and BackgroundDetect - playback with specification of detection 
modules to use
* GenderDetect, NoiseDetect, and AnswerDetect - new modules

Contact me off the list if you need updated modules or have questions, 
comments, or feedback.

Justin Newman
nt_jnewman at yahoo.com


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Re: [asterisk-users] Please help test the gender detection moduleat 575-613-4392

[Asterisk Asterisk]

Thanks for the feedback. I did some research and it looks like you were calling 
over international lines. It also appears that there was high than average 
static on the line, which is not normal for my system. It's true that I threw 
my recordings together quickly and the beep was supposed to be funny - it was 
actually me saying "beep". However, the static and noise you received was 
probably not from my system.

Nonetheless, I am working on improving the results of detection and will have a 
new release today or tomorrow. I'll post it up on the test systems for people 
to test and build additional data for refinement. Most importantly, I'll be 
adding a background noise filter and fine tuning the male/female results. After 
I get the gender detection done, I'll also be adding age range detection.

Justin

--Original Message--
From: Anselm Martin Hoffmeister
To: nt_jnew...@yahoo.com
Sent: Feb 18, 2009 4:09 AM
Subject: Re: [asterisk-users] Please help test the gender detection moduleat 
575-613-4392

Am Montag, den 16.02.2009, 11:45 -0800 schrieb Asterisk Asterisk:

> Let me know how it works when you try the test number at 575-613-4392.

Hi Justin,

I tried your module half an hour ago, and first of all, the bad sound
quality immediately came to my attention. The prompts where noisy as if
I had called from a mobile (which I did not) - intercontinental lines
may be the cause here.

The pause after the voice recording is quite long, giving the impression
that you somehow missed the (short) voice recording and still wait for
the caller's input: After the beep, I spoke as told, then had to wait
five or so seconds.

Besides, I have been detected as "female", three times in a row. I
should probably go to the bathroom and check, but I think you are in
error there.

BR
Anselm


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Re: [asterisk-users] Please help test the gender detection module at 575-613-4392

[Asterisk Asterisk]

Steve,

>Tried to test and got "call could not be completed as dialed".

Were you able to connect? If not, please try again. Call volume has been 
growing.

>How about a moving stress variable that could be used as a lie detector
of sorts or 
>just to measure how certain parts of a script, or certain
questions may

This is possible. Do you want to call or e-mail to discuss?


>I guess to get a baseline, you would have to ask a few inert questions. 

Yes, I definitely need to do this and will probably add this in for the next 
release.

Justin Newman
nt_jnewman at yahoo.com




From: Steve Totaro 
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Sent: Wednesday, February 18, 2009 10:57:47 AM
Subject: Re: [asterisk-users] Please help test the gender detection module at 
575-613-4392




On Wed, Feb 18, 2009 at 1:28 PM, Steve Totaro  
wrote:




On Mon, Feb 16, 2009 at 2:45 PM, Asterisk Asterisk  
wrote:

This module detects gender and approximate age range. I'm working on getting 
it's accuracy to 80%+ on a consistent basis, after implementing filters to 
remove background noise and other artifacts.

It's designed for a number of things. To start, I have several clients 
(primarily mobile content and servers providers) that want to profile and 
generate demographics of their users for selling advertising. They also want to 
understand their user base. Plus, some customers have found that male and 
female users tend to respond differently to different prompts, flows, etc. This 
helps in designing a system that meets needs of many different types of users.

Of course, there are many other uses and I'm sure people can generate some cool 
ideas.

Let me know how it works when you try the test number at 575-613-4392. Also, 
let me know if you have any interest in the module.

Justin

nt_jnewman at yahoo.com





From: Ron Joffe 
To: asterisk-users@lists.digium.com
Cc: Asterisk Asterisk 
Sent: Monday, February 16, 2009 11:05:24 AM
Subject: Re: [asterisk-users] Please help test the gender detection module at 
575-613-4392

That's an interesting module.

Care to elaborate on what you designed it for ?

Thanks,

Ron




On Monday 16 February 2009 13:29, Asterisk Asterisk wrote:
> I need your help: please help test the gender detection module at
> 575-613-4392.
>
> I wrote a gender detection module and thought I'd try it out. It only takes
> a second. I've been showing 90%+ accuracy and I want to make sure it's
> working correctly. Rain and significant background noise seems to throw it
> off, so I still have a bit of work to do.
>
> Have your friends and significant others call too. Also, let me know if you
> have any need for the module.
>
> Justin Newman
> nt_jnewman at yahoo.com


Tried to test and got "call could not be completed as dialed".

This sounds very interesting Justin.   

-- 
Thanks,
Steve Totaro 


Justin, how about building some additional functionality.  

How about a moving stress variable that could be used as a lie detector of 
sorts or just to measure how certain parts of a script, or certain questions 
may prove to be more stressful where simply rewording them may have a less 
stressful response?

I guess to get a baseline, you would have to ask a few inert questions. 

-- 
Thanks,
Steve Totaro 
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)



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Re: [asterisk-users] Please help test the gender detection moduleat 575-613-4392

[Asterisk Asterisk]

I hadn't even thought of that, but that's a great idea. I wrote some code that 
does speech recognition based on generated tokens and no learning needed. We 
could certainly apply the gender detection and that sr to a project like this. 
I would store only the token in my current model, but we could store voice too.

I'll send you another email with my contact info. Maybe we can talk offline and 
put something together this week? This would be really cool...voice auth and 
ID. Probably not too much work either.

Justin Newman
nt_jnewman at yahoo.com

-Original Message-
From: Jeff LaCoursiere 
Date: Thu, 19 Feb 2009 00:01:15 
To: Asterisk Users Mailing List - Non-Commercial 
Discussion
Cc: 
Subject: Re: [asterisk-users] Please help test the gender detection module
at 575-613-4392



Hi Justin,

How far is your work from being able to do speaker verification?  Not 
*identification* mind you, but being able to tell that a captured voice is 
the same as another that is stored...

Cheers,

j

On Wed, 18 Feb 2009, Asterisk Asterisk wrote:

> Steve,
>
>> Tried to test and got "call could not be completed as dialed".
>
> Were you able to connect? If not, please try again. Call volume has been 
> growing.
>
>> How about a moving stress variable that could be used as a lie detector
> of sorts or
>> just to measure how certain parts of a script, or certain
> questions may
>
> This is possible. Do you want to call or e-mail to discuss?
>
>
>> I guess to get a baseline, you would have to ask a few inert questions.
>
> Yes, I definitely need to do this and will probably add this in for the next 
> release.
>
> Justin Newman
> nt_jnewman at yahoo.com
>
>
>
> 
> From: Steve Totaro 
> To: Asterisk Users Mailing List - Non-Commercial Discussion 
> 
> Sent: Wednesday, February 18, 2009 10:57:47 AM
> Subject: Re: [asterisk-users] Please help test the gender detection module at 
> 575-613-4392
>
>
>
>
> On Wed, Feb 18, 2009 at 1:28 PM, Steve Totaro 
>  wrote:
>
>
>
>
> On Mon, Feb 16, 2009 at 2:45 PM, Asterisk Asterisk  
> wrote:
>
> This module detects gender and approximate age range. I'm working on getting 
> it's accuracy to 80%+ on a consistent basis, after implementing filters to 
> remove background noise and other artifacts.
>
> It's designed for a number of things. To start, I have several clients 
> (primarily mobile content and servers providers) that want to profile and 
> generate demographics of their users for selling advertising. They also want 
> to understand their user base. Plus, some customers have found that male and 
> female users tend to respond differently to different prompts, flows, etc. 
> This helps in designing a system that meets needs of many different types of 
> users.
>
> Of course, there are many other uses and I'm sure people can generate some 
> cool ideas.
>
> Let me know how it works when you try the test number at 575-613-4392. Also, 
> let me know if you have any interest in the module.
>
> Justin
>
> nt_jnewman at yahoo.com
>
>
>
>
> 
> From: Ron Joffe 
> To: asterisk-users@lists.digium.com
> Cc: Asterisk Asterisk 
> Sent: Monday, February 16, 2009 11:05:24 AM
> Subject: Re: [asterisk-users] Please help test the gender detection module at 
> 575-613-4392
>
> That's an interesting module.
>
> Care to elaborate on what you designed it for ?
>
> Thanks,
>
> Ron
>
>
>
>
> On Monday 16 February 2009 13:29, Asterisk Asterisk wrote:
>> I need your help: please help test the gender detection module at
>> 575-613-4392.
>>
>> I wrote a gender detection module and thought I'd try it out. It only takes
>> a second. I've been showing 90%+ accuracy and I want to make sure it's
>> working correctly. Rain and significant background noise seems to throw it
>> off, so I still have a bit of work to do.
>>
>> Have your friends and significant others call too. Also, let me know if you
>> have any need for the module.
>>
>> Justin Newman
>> nt_jnewman at yahoo.com
>
>
> Tried to test and got "call could not be completed as dialed".
>
> This sounds very interesting Justin.
>
> -- 
> Thanks,
> Steve Totaro
>
>
> Justin, how about building some additional functionality.
>
> How about a moving stress variable that could be used as a lie detector of 
> sorts or just to measure how certain parts of a script, or certain questions 
> may prove to be more stressful where simply rewording them may have a less 
> stressful response?
>
> I guess to get a baseline, you would have to ask a few inert questions.
>
> -- 
> Thanks,
> Steve Totaro
> +18887771888 (Toll Free)
> +12409381212 (Cell)
> +12024369784 (Skype)
>
>
>
>



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Re: [asterisk-users] US DID

[Asterisk Asterisk]

Don't most?





From: Nhadie 
To: Asterisk-users@lists.digium.com
Sent: Wednesday, February 18, 2009 6:19:24 AM
Subject: [asterisk-users] US DID

Hi,

Anyone knows a DID provider that can do both outbound and inbound?

Regards
Nhadie

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Re: [asterisk-users] check if not human

[Asterisk Asterisk]

You can probably use combo of NVLineDetect, NVGenderDetect, and AMD 
(NVMachineDetect).





From: Edwin Quijada 
To: Asterisk Asterisk 
Sent: Thursday, February 19, 2009 12:55:05 PM
Subject: Re: [asterisk-users] check if not human

 
How can I detect how many ring a call to hangup?
Where I can find info about AMD?

*---* 
*-Edwin Quijada 
*-Developer DataBase 
*-JQ Microsistemas 
*-809-849-8087
* " Si deseas lograr cosas excepcionales debes de hacer cosas fuera de lo 
comun" 
*---*





Get Windows Live and get whatever you need, wherever you are. Start here.


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Re: [asterisk-users] call picking and transfers

[Asterisk Asterisk]

You might also check with www.star2star.com (Star2Star Communications). We did 
a call park, pickup, and transfer module with similar functionality. Integrates 
very nicely.

Justin Newman
nt_jnewman at yahoo.com





From: Jeff LaCoursiere 
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Sent: Thursday, February 19, 2009 7:02:58 AM
Subject: Re: [asterisk-users] call picking and transfers



On Wed, 11 Feb 2009, Philipp Kempgen wrote:

> Jeff LaCoursiere schrieb:
>> Working on some niche requests from one of my hotel clients.  asterisk
>> 1.4.20-1 on CentOS, Polycom 501s.
>>
>> The first request is for the Polycom's screen to show the CID of the
>> inbound caller when a call pick is executed, so the picker knows if the
>> call is internal or external.  I have already "worked around" this issue
>> by using ALERT info to give seperate ring tones for outside and inside,
>> but they are used to their old Nortel switch which apparently showed the
>> CID immediately after the pick, and they then knew how to answer the
>> phone.
>>
>> The second is to show CID information on the screen when a call has been
>> answered by the front desk, then a blind transfer sent to an internal
>> phone.  Today they simply see "Front Desk" and there is no indication of
>> who the actual caller is to distinguish internal staff, internal guest
>> room, or outside caller.
>>
>> Has anyone attacked these things with Polycom that might share their
>> approach?
>
> These bugs cover the functionality you need I guess:
>
> http://bugs.digium.com/view.php?id=5014
> http://bugs.digium.com/view.php?id=13827
> http://bugs.digium.com/view.php?id=8824
>
> However none of the patches are likely to be merged into 1.4.
>

Hi,

I am very happy to announce that two patches:

http://bugs.digium.com/view.php?id=8824
http://bugs.digium.com/view.php?id=14206

applied to 1.4.23.1 work perfectly on Polycom, Cisco, and Linksys phones 
to supply:

* CALLED party information showing in display as remote phone is ringing
* blind and attended transfers show the original callers ID info to the 
receiving extension's display
* call picks using **ext or *8 show the picked caller's ID info in the 
display when it is completed.

These features are found in PBX equipment going back at least a decade, so 
I am very happy to see them finally in asterisk.  It does seem that these 
patches will become part of the 1.6 release soon if not already.

Cheers,

j

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Re: [asterisk-users] Busy status of a snom connected to two asterisk servers?

[Asterisk Asterisk]

We have a BLF module that maintains device state across Asterisk servers.

Contact me off the list if interested.

Justin Newman
nt_jnewman at yahoo.com





From: Lenz Emilitri 
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Sent: Thursday, February 19, 2009 6:46:20 AM
Subject: Re: [asterisk-users] Busy status of a snom connected to two asterisk 
servers?


One simple thing that comes to my mind is to have the SNOM connected to only 
one server, and send calls to from the queue on the second server to the first 
server, so that you can enforce a acall limit.
l.


2009/2/19 Rajkumar S 

Hi,

I have a snom 360 connected to two asterisk servers(both 1.6.0.5), via
two identities.  Each asterisk server runs a queue and snom is a
member of queue in both servers. Currently when snom is receiving call
from one asterisk server, it can still receive a call from the other
asterisk, because even though the snom is "busy" attending call from
one asterisk the other server does not know this.

Is there any way to share the actual busy status of snom to both
asterisk server so that when snom is receiving a call from one server
the other will see a busy status and will wait to become free before
connecting the call?

with warm regards,

raj

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Re: [asterisk-users] Please help test the gender detection module at 575-613-4392

[Asterisk Asterisk]

You sure you don't have a pony tail? :) Hehe.


It happens to the best of us. Hopefully after my fine tuning it will happen to 
less of us!




From: Darren Wiebe 
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Sent: Wednesday, February 18, 2009 4:13:46 PM
Subject: Re: [asterisk-users] Please help test the gender detection module at 
575-613-4392

Pretty cool.  I'm almost offended though as I'm not usually guessed as a 
female of the species. :)

Darren Wiebe
dar...@aleph-com.net

Asterisk Asterisk wrote:
> Steve,
>
> >Tried to test and got "call could not be completed as dialed".
>
> Were you able to connect? If not, please try again. Call volume has 
> been growing.
>
> >How about a moving stress variable that could be used as a lie 
> detector of sorts or
> >just to measure how certain parts of a script, or certain questions may
>
> This is possible. Do you want to call or e-mail to discuss?
>
> >I guess to get a baseline, you would have to ask a few inert questions.
>
> Yes, I definitely need to do this and will probably add this in for 
> the next release.
>
> Justin Newman
> nt_jnewman at yahoo.com
>
> 
> *From:* Steve Totaro 
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion 
> 
> *Sent:* Wednesday, February 18, 2009 10:57:47 AM
> *Subject:* Re: [asterisk-users] Please help test the gender detection 
> module at 575-613-4392
>
>
>
> On Wed, Feb 18, 2009 at 1:28 PM, Steve Totaro 
>  <mailto:stot...@totarotechnologies.com>> wrote:
>
>
>
> On Mon, Feb 16, 2009 at 2:45 PM, Asterisk Asterisk
> mailto:nt_aster...@yahoo.com>> wrote:
>
> This module detects gender and approximate age range. I'm
> working on getting it's accuracy to 80%+ on a consistent
> basis, after implementing filters to remove background noise
> and other artifacts.
>
> It's designed for a number of things. To start, I have several
> clients (primarily mobile content and servers providers) that
> want to profile and generate demographics of their users for
> selling advertising. They also want to understand their user
> base. Plus, some customers have found that male and female
> users tend to respond differently to different prompts, flows,
> etc. This helps in designing a system that meets needs of many
> different types of users.
>
> Of course, there are many other uses and I'm sure people can
> generate some cool ideas.
>
> Let me know how it works when you try the test number at
> 575-613-4392. Also, let me know if you have any interest in
> the module.
>
> Justin
>
> nt_jnewman at yahoo.com <http://yahoo.com>
>
>     
> 
> *From:* Ron Joffe  <mailto:ron.jo...@gmail.com>>
> *To:* asterisk-users@lists.digium.com
> <mailto:asterisk-users@lists.digium.com>
> *Cc:* Asterisk Asterisk  <mailto:nt_aster...@yahoo.com>>
> *Sent:* Monday, February 16, 2009 11:05:24 AM
> *Subject:* Re: [asterisk-users] Please help test the gender
> detection module at 575-613-4392
>
> That's an interesting module.
>
> Care to elaborate on what you designed it for ?
>
> Thanks,
>
> Ron
>
>
>
>
> On Monday 16 February 2009 13:29, Asterisk Asterisk wrote:
> > I need your help: please help test the gender detection
> module at
> > 575-613-4392.
> >
> > I wrote a gender detection module and thought I'd try it
> out. It only takes
> > a second. I've been showing 90%+ accuracy and I want to make
> sure it's
> > working correctly. Rain and significant background noise
> seems to throw it
> > off, so I still have a bit of work to do.
> >
> > Have your friends and significant others call too. Also, let
> me know if you
> > have any need for the module.
> >
> > Justin Newman
> > nt_jnewman at yahoo.com <http://yahoo.com>
>
>
> Tried to test and got "call could not be completed as dialed".
>
> This sounds very interesting Justin.  
>
> -- 
> Thanks,
> S

Re: [asterisk-users] Polycom Phones start to break up after being up a LONG time

[Asterisk Asterisk]

That's interesting - I haven't noticed this with any of my installs. What 
version of firmware and SIP?





From: Barry D. Hassler 
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Sent: Friday, February 20, 2009 8:41:33 AM
Subject: [asterisk-users] Polycom Phones start to break up after being up a 
LONG time

Has anyone else encountered this? I have a fairly large installation (~50 
phones, almost all Polycom 501's and a handful of 601's. We're running into a 
number of phones on which the outbound voice (Polycom phone user doesn't hear 
any problems, but the other end does) is breaking up occasionally -- enough to 
be noticeable and make you say "what?". In each case, rebooting the phone has 
resolved the symptoms, but I'd like to know if there is a known problem.

most of these phones would be up for several months now (installed this past 
summer), and unless there are any power outages, would not be restarted 
specifically.

I'm planning on restarting all the phones over the weekend, but as this is a 
24-hour operation, we'd like to avoid interrupting phones at all.

-- 
Barry D. Hassler
President, HCST

http://www.hcst.net/
937-427-9000



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Re: [asterisk-users] Please help test the gender detection module at 575-613-4392

[Asterisk Asterisk]

Let's chat about this on Saturday or Sunday. We could probably combine pieces 
of the vr and gd code to get something working. A peer of mine suggested using 
FFT with a neural network, but I honestly think that backend is going to take 
way too much processing, too much effort, and beat the purpose. I'm sure we can 
get something that works well enough and that fits in the 20ms frames.


Justin




From: Jeff LaCoursiere 
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Cc: nt_jnew...@yahoo.com
Sent: Wednesday, February 18, 2009 4:01:15 PM
Subject: Re: [asterisk-users] Please help test the gender detection module at 
575-613-4392


Hi Justin,

How far is your work from being able to do speaker verification?  Not 
*identification* mind you, but being able to tell that a captured voice is 
the same as another that is stored...

Cheers,

j

On Wed, 18 Feb 2009, Asterisk Asterisk wrote:

> Steve,
>
>> Tried to test and got "call could not be completed as dialed".
>
> Were you able to connect? If not, please try again. Call volume has been 
> growing.
>
>> How about a moving stress variable that could be used as a lie detector
> of sorts or
>> just to measure how certain parts of a script, or certain
> questions may
>
> This is possible. Do you want to call or e-mail to discuss?
>
>
>> I guess to get a baseline, you would have to ask a few inert questions.
>
> Yes, I definitely need to do this and will probably add this in for the next 
> release.
>
> Justin Newman
> nt_jnewman at yahoo.com
>
>
>
> 
> From: Steve Totaro 
> To: Asterisk Users Mailing List - Non-Commercial Discussion 
> 
> Sent: Wednesday, February 18, 2009 10:57:47 AM
> Subject: Re: [asterisk-users] Please help test the gender detection module at 
> 575-613-4392
>
>
>
>
> On Wed, Feb 18, 2009 at 1:28 PM, Steve Totaro 
>  wrote:
>
>
>
>
> On Mon, Feb 16, 2009 at 2:45 PM, Asterisk Asterisk  
> wrote:
>
> This module detects gender and approximate age range. I'm working on getting 
> it's accuracy to 80%+ on a consistent basis, after implementing filters to 
> remove background noise and other artifacts.
>
> It's designed for a number of things. To start, I have several clients 
> (primarily mobile content and servers providers) that want to profile and 
> generate demographics of their users for selling advertising. They also want 
> to understand their user base. Plus, some customers have found that male and 
> female users tend to respond differently to different prompts, flows, etc. 
> This helps in designing a system that meets needs of many different types of 
> users.
>
> Of course, there are many other uses and I'm sure people can generate some 
> cool ideas.
>
> Let me know how it works when you try the test number at 575-613-4392. Also, 
> let me know if you have any interest in the module.
>
> Justin
>
> nt_jnewman at yahoo.com
>
>
>
>
> 
> From: Ron Joffe 
> To: asterisk-users@lists.digium.com
> Cc: Asterisk Asterisk 
> Sent: Monday, February 16, 2009 11:05:24 AM
> Subject: Re: [asterisk-users] Please help test the gender detection module at 
> 575-613-4392
>
> That's an interesting module.
>
> Care to elaborate on what you designed it for ?
>
> Thanks,
>
> Ron
>
>
>
>
> On Monday 16 February 2009 13:29, Asterisk Asterisk wrote:
>> I need your help: please help test the gender detection module at
>> 575-613-4392.
>>
>> I wrote a gender detection module and thought I'd try it out. It only takes
>> a second. I've been showing 90%+ accuracy and I want to make sure it's
>> working correctly. Rain and significant background noise seems to throw it
>> off, so I still have a bit of work to do.
>>
>> Have your friends and significant others call too. Also, let me know if you
>> have any need for the module.
>>
>> Justin Newman
>> nt_jnewman at yahoo.com
>
>
> Tried to test and got "call could not be completed as dialed".
>
> This sounds very interesting Justin.
>
> -- 
> Thanks,
> Steve Totaro
>
>
> Justin, how about building some additional functionality.
>
> How about a moving stress variable that could be used as a lie detector of 
> sorts or just to measure how certain parts of a script, or certain questions 
> may prove to be more stressful where simply rewording them may have a less 
> stressful response?
>
> I guess to get a baseline, you would have to ask a few inert questions.
>
> -- 
> Thanks,
> Steve Totaro
> +18887771888 (To

Re: [asterisk-users] Please help test the gender detection module at 575-613-4392

[Asterisk Asterisk]

>BTW, I love the beep.

It's me saying "bp". :)





From: Steve Totaro 
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Sent: Thursday, February 19, 2009 4:23:05 PM
Subject: Re: [asterisk-users] Please help test the gender detection module at 
575-613-4392

It got my gender correct the two times I tested, even with the TV loud in the 
background.

BTW, I love the beep.



On Thu, Feb 19, 2009 at 5:54 PM, Asterisk Asterisk  
wrote:

You sure you don't have a pony tail? :) Hehe.


It happens to the best of us. Hopefully after my fine tuning it will happen to 
less of us!




From: Darren Wiebe 

To: Asterisk Users Mailing List - Non-Commercial Discussion 

Sent: Wednesday, February 18, 2009 4:13:46 PM

Subject: Re: [asterisk-users] Please help test the gender detection module at 
575-613-4392


Pretty cool.  I'm almost offended though as I'm not usually guessed as a 
female of the species. :)

Darren Wiebe
dar...@aleph-com.net

Asterisk Asterisk wrote:
> Steve,
>
> >Tried to test and got "call could not be completed as dialed".
>
> Were you able to connect? If not, please try again. Call volume has 
> been growing.
>
> >How about a moving stress variable that could be used as a lie 
> detector of sorts or
> >just to measure how certain parts of a script, or certain questions may
>
> This is possible. Do you want to call or e-mail to discuss?
>
> >I guess to get a baseline, you would have to ask a few inert questions.
>
> Yes, I definitely need to do this and will probably add this in for 
> the next release.
>
> Justin Newman
> nt_jnewman at yahoo.com
>
> 
> *From:* Steve Totaro 
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion 
> 
> *Sent:* Wednesday, February 18, 2009 10:57:47 AM
> *Subject:* Re: [asterisk-users] Please help test the gender detection 
> module at 575-613-4392
>
>
>
> On Wed, Feb 18, 2009 at 1:28 PM, Steve Totaro 
>  <mailto:stot...@totarotechnologies.com>> wrote:
>
>
>
> On Mon, Feb 16, 2009 at 2:45 PM, Asterisk Asterisk
> mailto:nt_aster...@yahoo.com>> wrote:
>
> This module detects gender and approximate age range. I'm
> working on getting it's accuracy to 80%+ on a consistent
> basis, after implementing filters to remove background noise
> and other artifacts.
>
> It's designed for a number of things. To start, I have several
> clients (primarily mobile content and servers providers) that
> want to profile and generate demographics of their users for
> selling advertising. They also want to understand their user
> base. Plus, some customers have found that male and female
> users tend to respond differently to different prompts, flows,
> etc. This helps in designing a system that meets needs of many
> different types of users.
>
> Of course, there are many other uses and I'm sure people can
> generate some cool ideas.
>
> Let me know how it works when you try the test number at
> 575-613-4392. Also, let me know if you have any interest in
> the module.
>
> Justin
>
> nt_jnewman at yahoo.com <http://yahoo.com>
>
> 
> 
> *From:* Ron Joffe  <mailto:ron.jo...@gmail.com>>
> *To:* asterisk-users@lists.digium.com
> <mailto:asterisk-users@lists.digium.com>
> *Cc:* Asterisk Asterisk  <mailto:nt_aster...@yahoo.com>>
> *Sent:* Monday, February 16, 2009 11:05:24 AM
> *Subject:* Re: [asterisk-users] Please help test the gender
> detection module at 575-613-4392
>
> That's an interesting module.
>
> Care to elaborate on what you designed it for ?
>
> Thanks,
>
> Ron
>
>
>
>
> On Monday 16 February 2009 13:29, Asterisk Asterisk wrote:
> > I need your help: please help test the gender detection
> module at
> > 575-613-4392.
> >
> > I wrote a gender detection module and thought I'd try it
> out. It only takes
> > a second. I've been showing 90%+ accuracy and I want to make
> sure it's
> > working correctly. Rain and significant background noise
> seems to throw it
> > off, so I still have a bit of work to do.
> >
> > Have your friend

Re: [asterisk-users] check if not human

[Asterisk Asterisk]

NVGenderDetect is new, but you can find NVLineDetect on the web.





From: David fire 
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Sent: Thursday, February 19, 2009 3:00:14 PM
Subject: Re: [asterisk-users] check if not human

NVLineDetect, NVGenderDetect what is that?

amd info voip-info.org or asterisk.org support asterisk book.

i bougth one to support the cause!!!

David


2009/2/19 Asterisk Asterisk 

You can probably use combo of NVLineDetect, NVGenderDetect, and AMD 
(NVMachineDetect).





From: Edwin Quijada 
To: Asterisk Asterisk 
Sent: Thursday, February 19, 2009 12:55:05 PM
Subject: Re: [asterisk-users] check if not human



How can I detect how many ring a call to hangup?
Where I can find info about AMD?

*---* 
*-Edwin Quijada 
*-Developer DataBase 
*-JQ Microsistemas 
*-809-849-8087
* " Si deseas lograr cosas excepcionales debes de hacer cosas fuera de lo 
comun" 
*---*





Get Windows Live and get whatever you need, wherever you are. Start here.

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-- 
(\__/) 
(='.'=)This is Bunny. Copy and paste bunny into your 
(")_(")signature to help him gain world domination. 


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Re: [asterisk-users] Please help test the gender detection module at 575-613-4392

[Asterisk Asterisk]

We've had a 65% success rate across the board (actually 35% incorrect). I'm 
working on bringing that up to 85% or better.





From: Ira 
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Sent: Thursday, February 19, 2009 5:18:07 PM
Subject: Re: [asterisk-users] Please help test the gender detection module at 
575-613-4392

At 04:23 PM 2/19/2009, you wrote:
>It got my gender correct the two times I tested, even with the TV 
>loud in the background.

It got me wrong twice, but so do about 30% of the people who call.

Ira 


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Re: [asterisk-users] Please help test the gender detection module at 575-613-4392

[Asterisk Asterisk]

Yes, the gender is asked at the end.





From: Jeff LaCoursiere 
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Cc: nt_jnew...@yahoo.com
Sent: Friday, February 20, 2009 9:41:38 AM
Subject: Re: [asterisk-users] Please help test the gender detection module at 
575-613-4392


How do you know if they are correct or not?  Gender is sometimes mistaken 
by humans answering the phone... can we expect the algorithm to do better 
than that?  I actually haven't called yet (blush) - are you asking at the 
end if it correctly identified you?

Cheers,

j

On Fri, 20 Feb 2009, Asterisk Asterisk wrote:

> We've had a 65% success rate across the board (actually 35% incorrect). I'm 
> working on bringing that up to 85% or better.
>
>
>
>
> 
> From: Ira 
> To: Asterisk Users Mailing List - Non-Commercial Discussion 
> 
> Sent: Thursday, February 19, 2009 5:18:07 PM
> Subject: Re: [asterisk-users] Please help test the gender detection module at 
> 575-613-4392
>
> At 04:23 PM 2/19/2009, you wrote:
>> It got my gender correct the two times I tested, even with the TV
>> loud in the background.
>
> It got me wrong twice, but so do about 30% of the people who call.
>
> Ira
>
>
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>
>
>
>

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Re: [asterisk-users] Please help test the gender detection module at 575-613-4392

[Asterisk Asterisk]

You have some good points.

>Justin Newman isn't exactly "someone we don't know". However I only

I agree that my name wasn't clear, but I was trying to avoid getting a bunch of 
spam myself. I'm not sure if I've personally ever spammed the list and I'm 
pretty supportive of the community. I have been part of these lists for many 
many years.

>* The message starts by asking you to call a number.

That was the help needed and it worked. There have been more than 500 different 
callers now and they keep coming in. I'm going to need help with a second round 
of testing, after I release the updates today and Sunday, but I haven't figured 
out how to entice people to test again. I thought about doing an outbound call 
and most people probably wouldn't care, but I'm anti-spam myself and that 
sounds like spam to me! Any thoughts?

>* No page with further information. Such a page helps to provide further

I'm open for suggestions. Let me know how I can be more helpful. By the way, 
thanks everyone for YOUR help!

>* Is this module intended to be free software? (As it is not mentioned,
>  I guess: "no"). If not, I'm less motivated to help to it.


Yes, it's free as in GPL. I'm fine tuning and then releasing to the community. 
My customer paid to have the base written with the assumption that it would be 
released GPL, with further community work down the road. Let me know if you 
have any thoughts, feedback, or suggestions.




From: Tzafrir Cohen 
To: asterisk-users@lists.digium.com
Sent: Friday, February 20, 2009 10:48:10 AM
Subject: Re: [asterisk-users] Please help test the gender detection module at 
575-613-4392

Slightly off-topic,

On Mon, Feb 16, 2009 at 10:29:57AM -0800, Asterisk Asterisk wrote:
> I need your help: please help test the gender detection module at 
> 575-613-4392.
> 
> I wrote a gender detection module and thought I'd try it out. It only 
> takes a second. I've been showing 90%+ accuracy and I want to make 
> sure it's working correctly. Rain and significant background noise 
> seems to throw it off, so I still have a bit of work to do.
> 
> Have your friends and significant others call too. Also, let me know 
> if you have any need for the module.
> 
> Justin Newman
> nt_jnewman at yahoo.com

When someone I don't know asks me to do something I'd hesitate a bit. 
Justin Newman isn't exactly "someone we don't know". However I only
found that out by reading the signature of the message. This message
has several indicators of a spam message:

* The declared name is a completely generic one. This is normally
  indicative of some newbies who have not yet "earned" a name on the
  list.

* The message starts by asking you to call a number.

* No page with further information. Such a page helps to provide further
  verification: I can tell that whoever wrote this message and whoever
  wrote the page are the same.

* Is this module intended to be free software? (As it is not mentioned,
  I guess: "no"). If not, I'm less motivated to help to it.

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

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Re: [asterisk-users] Please help test the gender detection module at 575-613-4392

[Asterisk Asterisk]

Steve -- You are absolutely right and I appreciate your feedback. Let me know 
if you have any interest being involved with this project.

In response, I should clarify by pointing out that the company who engaged me 
in this effort was calling for 80% accuracy. Their target audience does not 
include minors, so children were not a consideration. They are using SIP lines 
with great quality not far from decent digital and analog lines. We could 
probably average better than 90% in their environment. However, I'd rather 
overachieve than restate those numbers, so they work for me.

I have another 1-2 hours of work to get the module close to the 85% range in my 
environment. I know what I need to do, but may need filter help. I've just been 
trying to find time. With a good clean signal, I'm sure people can get 90% or 
better. On this next set of tests, I'd also like to collect more information, 
including age and zip. Figuring out how to do these things without affecting 
the tests seems to take more time than the code itself.

The numbers: the 65% detection rate I stated earlier is increasing and it 
probably closer to 70% now. Our actual detection rate is higher, but this gives 
us a idea of how the module will perform in various settings. There are several 
negative factors affecting results, including the following:
* Pos: I'm running against Asterisk 1.4.9 on a old laptop, a crappy SIP line, 
and Comcast with a substandard SIP service
* Noise: We started testing in the bay area, where it has been raining for a 
week; that background noise threw off many of the tests
* Noise: Many failures came from international callers; noisy lines and 
background noise (like my wife singing) throw it off
* Code: The code isn't done yet and there is room for some improvement; the 
code needs to be fine tuned for some side cases
* Play: Many testers were playing with the system; that includes my wife who 
claims she can sound like a male (why in the world??)

Many of the failures came from noisey lines, playing (like my wife pretending 
to be a male), and international callers, from what I've observed. We've had 
the following approximate demographics: male 73%+ (including false female) and 
domestic 90%+ (without hidden numbers). That fits my estimates of the list 
demographics combined with the peers and friends I asked to test, so the honor 
system appears to be working (honesty prevails).

If anyone needs the data output by this module (test data for the past few 
days), I'd be happy to share. It includes the date/time, gender detected, 
whether that's correct according to the tester, the winning ratio, and the 
energy levels broken up into 25hz bands. If there is additional information 
that I should record, which would be benefitial to the community, let me know 
and I will work on incorporating.


Justin Newman  (yes, this is the real deal, not spam!)
nt_jnewman at yahoo.com




From: Steve Underwood 
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Sent: Friday, February 20, 2009 7:27:12 PM
Subject: Re: [asterisk-users] Please help test the gender detection module at 
575-613-4392

Hi,

Asterisk Asterisk wrote:
> We've had a 65% success rate across the board (actually 35% 
> incorrect). I'm working on bringing that up to 85% or better.
Good gender recognisers get >90% success on PSTN lines. In restricted 
contexts they can get up to 98%. These figures are not entirely honest, 
as they assume no children call. Recognisers get terrible results for 
boys who's voices have yet to break.

Regards,
Steve

>
> 
> *From:* Ira 
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion 
> 
> *Sent:* Thursday, February 19, 2009 5:18:07 PM
> *Subject:* Re: [asterisk-users] Please help test the gender detection 
> module at 575-613-4392
>
> At 04:23 PM 2/19/2009, you wrote:
> >It got my gender correct the two times I tested, even with the TV
> >loud in the background.
>
> It got me wrong twice, but so do about 30% of the people who call.
>
> Ira
>


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Re: [asterisk-users] Please help test the gender detection module at 575-613-4392

[Asterisk Asterisk]

Tzafrir- Again, very good points. See my responses below.


>What does this have to do with using 'Asterisk Asterisk' instead of
>'Justin Newman'?

I was sick of all the junk mail in my old accounts and working with digest mode 
was a pain, so I quickly created a new yahoo.com account a few days ago. I was 
just concerned with creating a new account at that time, so I quickly put in 
"Asterisk Asterisk". However, I just went and updated the name and signature. 
Hope this helps and the fame will follow!

>Have those been deployed already?

I deployed a few small changes as planned and I'll finish the bulk of it this 
weekend.

>Where can I find it, then?

Honestly, I just wrote the app and was asking folks to test to see if it was 
even worthwhile at this point. It's working pretty well and I can see that 
there is interest, so I'll get it posted soon. I have always released my gpl 
modules in the past, so I'm not sure why this is a concern. Plus, I'd hope that 
any posting works well enough before pushing a whole lot of noise on to the 
wiki and source sites.

>This code is NOT included with Asterisk at this point, however it is

Yes, I do prefer to release add-ons. Asterisk seems pretty bulky to me and I 
prefer the alternative to Digium's disclaimer.

>SourceForge, GoogleCode or whatever. Having the programs available under
>version control means a lot to others. All three provide a simple wiki
>for your documentation (the latter two have some more advanced options).

I'll add it to sourceforge and my site, along with some others. It is already 
in cvs. I'll post the documentation on voip-info.org like I have done in the 
past. My source packages in the past two years also include much better 
documentation and a cleaner build system. If you want to help with any of this 
posting or management, you'll help would be a breath of fresh air. I'm usually 
very busy writing code or doing business, which an get in the way.

>Furthermore, your Voip-Info user page comes up high when searching for
>'Justin Newman asterisk'. Why not put there some more useful details?

There are quite a few Justin Newman's now. I used to think my name was not 
common. A quick search will reveal some interesting stories. One Justin Newman 
was killed in Chicago so that another man could fake his death. I was once 
pulled over in Washington for speeding down I-5 in my bimmer and the officer 
asked if I had a warrant out for my arrest due to name likeness. There are also 
several of us Justin Newman's in the VOIP world.

Anyhow, I hope some of my efforts above benefit the community. Feel free to 
catch me on IM.

---
Justin Newman
Sr Software Engineer
Envy Software LLC
nt_asterisk at yahoo.com (general list)
nt_jnewman at yahoo.com (personal)
justin_newman (skype im)
justin_newman (yahoo im)




From: Tzafrir Cohen 
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Sent: Saturday, February 21, 2009 1:17:48 AM
Subject: Re: [asterisk-users] Please help test the gender detection module at 
575-613-4392

On Fri, Feb 20, 2009 at 11:16:06AM -0800, Asterisk Asterisk wrote:
> You have some good points.
> 
> >Justin Newman isn't exactly "someone we don't know". However I only
> 
> I agree that my name wasn't clear, but I was trying to avoid getting a 
> bunch of spam myself. I'm not sure if I've personally ever spammed the 
> list and I'm pretty supportive of the community. I have been part of 
> these lists for many many years.

What does this have to do with using 'Asterisk Asterisk' instead of
'Justin Newman'?

> 
> >* The message starts by asking you to call a number.
> 
> That was the help needed and it worked. There have been more than 500 
> different callers now and they keep coming in. I'm going to need help 
> with a second round of testing, after I release the updates today and 
> Sunday, but I haven't figured out how to entice people to test again. 

Have those been deployed already?

> 
> >* No page with further information. Such a page helps to provide further
> 
> I'm open for suggestions. Let me know how I can be more helpful. By the 
> way, thanks everyone for YOUR help!
> 

See below regarding project hosting.

> >* Is this module intended to be free software? (As it is not mentioned,
> >  I guess: "no"). If not, I'm less motivated to help to it.
> 
> 
> Yes, it's free as in GPL. I'm fine tuning and then releasing to the 
> community. My customer paid to have the base written with the 
> assumption that it would be released GPL, with further community work 
> down the road. Let me know if you have any thoughts, feedback, or 
> suggestions.

Where

Re: [asterisk-users] Please help test the gender detection module at 575-613-4392

[Asterisk Asterisk]

What's your caller ID? :)

Thanks for your help.





From: Darren Wiebe 
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Sent: Friday, February 20, 2009 9:08:03 PM
Subject: Re: [asterisk-users] Please help test the gender detection module at 
575-613-4392

Asterisk Asterisk wrote:
> You have some good points.
>
> >Justin Newman isn't exactly "someone we don't know". However I only
>
> I agree that my name wasn't clear, but I was trying to avoid getting a 
> bunch of spam myself. I'm not sure if I've personally ever spammed the 
> list and I'm pretty supportive of the community. I have been part of 
> these lists for many many years.
>
> >* The message starts by asking you to call a number.
>
> That was the help needed and it worked. There have been more than 500 
> different callers now and they keep coming in. I'm going to need help 
> with a second round of testing, after I release the updates today and 
> Sunday, but I haven't figured out how to entice people to test again. 
> I thought about doing an outbound call and most people probably 
> wouldn't care, but I'm anti-spam myself and that sounds like spam to 
> me! Any thoughts?
>
-- Snipped --

I'll be happy to try it again to see if I've become a male yet. :)

Darren Wiebe
dar...@aleph-com.net


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[asterisk-users] Asterisk 1.4.0 Installation error on Red Hat Linux 9.0-Urgent

[Asterisk Asterisk]

Hey,
   
  Implementing Asterisk on local Lan spread over 2 campuses on two different 
cities is our  graduation project. 
   
  Having done all the research and reading stuff. I started with the practical 
work.
  Not getting a hand on the  linux digium. I installed Red Hat linux 9.0. I was 
able to install zaptel1.4.0 and libpri1.4.0 successfully. When i attempted to 
install linux and put in following commands.
   
  # cd /usr/src/asterisk-version
  # make clean
  # make
   
  at the "make" command i got following error:
   
  
   The configure script must be make before running make
  
  make: *** [makeopts] Error 1 
   
   
  Please somebody could help me out.
   
  Regards,
   
  The Szabistians

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[asterisk-users] Fwd: Can't hear any sound

[Asterisk Asterisk]

Note: forwarded message attached.
 Send instant messages to your online friends http://uk.messenger.yahoo.com --- Begin Message ---
Hey,

I am new to asterisk and softphones. I am able to install astersik and 2 XLite  
softphones on three PCs with linux feora core 6. I have also written a basic 
dial plan to make calls between two clients.But when i dial from a pc to 
another PC the calls goes through i can hear the ring tone and also recieve 
call but i can't hear any voice.

Please could anyone help me.

Regards,

Szabstians

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[asterisk-users] Can't hear any sound (This time in plain text)

[Asterisk Asterisk]

Hey,

I am a new to asterisk and softphones. Ihave recently
installed and configured linux and 2 xlite clients all
in  linux fedora core 6. I have also made a dial plan
for the two users. But when i dial from one xlite
client to another i can hear the ring tone but when i
answer the call i can not hear any sound. 

I have checked my microphone and its working fine.

Please could anyone help me on this issue.

Sorry,I did not know that the list could not open any
attachments.

Regards Szabistians

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[asterisk-users] Problem configuring voice conference

[Asterisk Asterisk]

Hey!

I am trying to configure the voice onference with
MeetMe application for my internal users. I have my
server and 4 clients on same LAN and following is my
extensions.conf file:

[globals]
Ahsen=SIP/222
Tahami=SIP/444
Uzair=SIP/333
Wasif=SIP/555


[internal]
exten => 1234,1,Macro(voicemail,${Ahsen})
exten => 4321,1,Macro(voicemail,${Uzair})
exten => 5678,1,Macro(voicemail,${Tahami})
exten => 8765,1,Macro(voicemail,${Wasif})


;For Call Conferencing
;Here the syntax is exten => extension(normally the
conf room no.),prority,
;MeetMe(conf. room no.,i(announces when people enter
and exit the conference),password)

exten => 700,1,MeetMe(600,i,1234)

;limit the conference room to 10 participants
exten => 700,1,MeetMeCount(600,CONFCOUNT)
exten => 700,2,GotoIf($[${CONFCOUNT} <= 10]?3:100)
exten => 700,3,MeetMe(600,i,1234)
exten => 700,100,Playback(conf-full)


[macro-voicemail]
exten => s,1,Dial(${ARG1},10)
exten => s,2,VoiceMail([EMAIL PROTECTED])
exten => s,102,VoiceMail([EMAIL PROTECTED])

;So usrs can dial 500 to access their voicemail
exten => 500,1,VoiceMailMain( )

But from any client i dial extension 700 to initiate
teh conference i get the following error at asterisk
CLI:

[Mar 12 15:41:38] WARNING[2756]: pbx.c:1779
pbx_extension_helper: No application 'MeetMe' for
extension (internal, 700, 1)


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[asterisk-users] Problem configuring voice conference

[Asterisk Asterisk]

Hey i installed zaptel and when i tried to install
asterisk and ran command menuselect it showed me that
there are some discrepencies that are not being
fullfilled for meetme application, but i have also
installed ztdummy when i installed zaptel. I am
totally stuck and nowhere to go what should i do.

--- Paul Hales <[EMAIL PROTECTED]> wrote:

> 
> Sure, but you will probably have to recompile
> Asterisk to get all the
> extra bits.
> 
> Should only take you 10 minutes.
> 
> later,
> 
> PaulH
> 
> On Mon, 2007-03-12 at 06:54 +, Asterisk Asterisk
> wrote:
> > Hey! Thanks you are absolutely rite could i
> install
> > ity now after i have compiled and installed
> asterisk
> > or not.
> > 
> > 
> > Send instant messages to your online friends
> http://uk.messenger.yahoo.com 
> 
> 


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[Asterisk-Users] TMD card to buy.

[asterisk asterisk]

Hello ,
I want to install a littel office . I have some question regarding to it.
My office has 8 analog line in and we would 20 line out (analog ).
So as a sow I need 2 TMD fxo card  and 5 TMD txs card , Am I right ?
Can I use these card in one PC ?, or I need more PC s ? Can I install driver in one PC for 7 TMD cards ?
Iam from Hungary , where can I purchase cards ? Is there any company in Hungary or nearby hungary ?
 
Thanks a lot !
 
 
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[Asterisk-Users] Linux OS platforms

[asterisk asterisk]

I have a question regarding to OS platform.
As I see on Wiki -s homepage there are many type of linux version.And in some of them there are reported errors (regarding to asterisk ) for exemole in rad hat .
Can you tell me what is the best linux paltform ,( version ), which is supported by digiroom card (T1 and TMD )and asterisk run on it stable ?.
Which linux is prefereable ? for asterisk ?
Hanks , 
Roby.
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[Asterisk-Users] Wildcard TE110P works with 2 channel ISDN ?

[asterisk asterisk]

Hello ,
I have a question regarding to PRI card (Wildcard TE110P).We want ot use this card in Hungary .So if we have a PRI line (30 B channel (64Kb) and 2 D channel) is is good (I think)
But what happends then when we have only BRI (2-D channel + 1-D channel for signaling) ?
Does it works this card (Wildcard TE110P) with 2 lines , 4  lines ISDN ? Or just with PRI (30 B channel) ?
Can you tell me or where I can find some more info abaut E1 card ?Thank you.
Regards Kallos Robert.
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[Asterisk-Users] Extension context question

[asterisk asterisk]

I have a very simple question .
I have 2 internal extension 301 and 300 sip phone . I want to these extesion can call each other, and ext 300 can call outside to pstn, and ext 301 to call internatonal.  
How can I do that ? 
 
[x1]exten => 300,1,Dial(SIP/300)
include => pstnlocal
[x2]exten => 301,1,Dial(SIP/301)
include =>international
[pstnlocal]
exten => _9xxx,1,Dial(Zap/g1/${EXTEN}) 
[international]
exten => _900.,1,Dial(Zap/g1/${EXTEN}) 
 
 
So it is good but in this case I cann t call the local phone .And if I include context x1 in x2 and x2 in x1 the ext 300 will be able to call international no.
 
Can anyone help me ?
 
Thanks.
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Re: [Asterisk-Users] Extension context question

[asterisk asterisk]

Yes Pstn local start with 9 and pstn international starts with 00 .That is ok.
I can make call form 300 to pstn local, and from ext 301 to pstn international, that is ok .But in this exemple I can not call form ext 300 to 301 and form 301 to 300. 
 
It is possibile  to have 2 diferent group of extension, with diferent tyipe of permision (like here x1 pstn local , and x2 pstn international) but extension can call each other regardless form the group. It does not matther in which group is the pone I can call any phone form group x1 or x2.
But if I want to call outside it depends on permision of group in wich I am (x1 or x2).
 
Ronald Wiplinger <[EMAIL PROTECTED]> wrote:
asterisk asterisk wrote:> I have a very simple question .>> I have 2 internal extension 301 and 300 sip phone . I want to these > extesion can call each other, and ext 300 can call outside to pstn, > and ext 301 to call internatonal. >> How can I do that ?>> >include pstnlocal at either [x2] or [international]Remember, that international starts with 00 while local never has 00 - right?byeRonald> [x1]> exten => 300,1,Dial(SIP/300)>> include => pstnlocal>> [x2]> exten => 301,1,Dial(SIP/301)>> include =>international>include => pstnlocal> [pstnlocal]>> exten => _9xxx,1,Dial(Zap/g1/${EXTEN}) >> [international]>> exten 
 =>
 _900.,1,Dial(Zap/g1/${EXTEN}) >include => pstnlocal> >> So it is good but in this case I cann t call the local phone .And if I > include context x1 in x2 and x2 in x1 the ext 300 will be able to call > international no.>> >> Can anyone help me ?>> >> Thanks.>> > Do You Yahoo!?> Yahoo! Small Business - Try our new Resources site! > >>>>>___>Asterisk-Users mailing list>Asterisk-Users@lists.digium.com>http://lists.digium.com/mailman/listinfo/asterisk-users>To UNSUBSCRIBE or update options visit:>
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[Asterisk-Users] Asterisk realtime , asterisk extensions not load form db.

[asterisk asterisk]

Hello !
I try to run asterisk with real time config from database.
I use AMP to configure .
Everythig it ok , I can set new sip and iax extensions, I can see them on mysql db , as well is amp .
But these extension I cannt use in asterisk .
I have seen some new conf file 
;sip_additional.conf;iax_additional.conf;extensions_additional.conf;meetme_additional.conf
.If I reload asterisk these extensions are not load .
I have these tabele in mysql ,sip, iax, extensions.
my extconfig.conf file 
[settings]
;uncomment to load queues.conf via the db engine.;queues.conf => odbc; => ,[,table_name]
voicemail.conf => mysql,asterisk,voicemail_table   ; it is goodsip.conf => mysql,asterisk,sip_tableiax => mysql,asterisk,iaxextensions.conf => mysql,asterisk,extensions
 
How can I set asterisk to load config form database ?
Is there anyone who have the same problem ?
Thans for help !
Roby.
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RE: [Asterisk-Users] Asterisk realtime , asterisk extensions not load form db.

[asterisk asterisk]

Yes I've checked . these pakeche I have instaled.
But it does not work.
 
 
 
echo "libxml2" rpm -qa|grep libxml2 echo "libtiff" rpm -qa|grep libtiff echo "libtiff-devel" rpm -qa|grep libtiff-devel echo "httpd" rpm -qa|grep httpd echo "mysql" rpm -qa|grep mysql echo "mysql -devel" rpm -qa|grep mysql-devel echo "mysql -server" rpm -qa|grep mysql-server echo "php" rpm -qa|grep php echo "php -mysql" rpm -qa|grep php-mysql echo "openssl" rpm -qa|grep openssl echo "openssl -devel"&nb
 sp;rpm
 -qa|grep openssl-devel echo "kernel -source" rpm -qa|grep kernel-source echo "perl" rpm -qa|grep perl echo "perl -CPAN" rpm -qa|grep perl-CPAN echo "cvs" rpm -qa|grep cvs echo "bison" rpm -qa|grep bison echo "ncurses -devel" rpm -qa|grep ncurses-devel echo "audiofile -devel" rpm -qa|grep audiofile-devel echo "-" Roman Zhovtulya <[EMAIL PROTECTED]> wrote:


You've got to check if you have all the required mysql libraries installed (mysql client and mysql-devel)
 
 


-Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of asterisk asteriskSent: Donnerstag, 3. März 2005 10:13To: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Asterisk realtime ,asterisk extensions not load form db.


Hello !
I try to run asterisk with real time config from database.
I use AMP to configure .
Everythig it ok , I can set new sip and iax extensions, I can see them on mysql db , as well is amp .
But these extension I cannt use in asterisk .
I have seen some new conf file 
;sip_additional.conf;iax_additional.conf;extensions_additional.conf;meetme_additional.conf
.If I reload asterisk these extensions are not load .
I have these tabele in mysql ,sip, iax, extensions.
my extconfig.conf file 
[settings]
;uncomment to load queues.conf via the db engine.;queues.conf => odbc; => ,[,table_name]
voicemail.conf => mysql,asterisk,voicemail_table   ; it is goodsip.conf => mysql,asterisk,sip_tableiax => mysql,asterisk,iaxextensions.conf => mysql,asterisk,extensions
 
How can I set asterisk to load config form database ?
Is there anyone who have the same problem ?
Thans for help !
Roby.
 
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[Asterisk-Users] Asterisk realtime , asterisk extensions not load form db.

[asterisk asterisk]

Yes I've checked . these pakeche I have instaled.
But it does not work.
 
 
 
echo "libxml2" rpm -qa|grep libxml2 echo "libtiff" rpm -qa|grep libtiff echo "libtiff-devel" rpm -qa|grep libtiff-devel echo "httpd" rpm -qa|grep httpd echo "mysql" rpm -qa|grep mysql echo "mysql -devel" rpm -qa|grep mysql-devel echo "mysql -server" rpm -qa|grep mysql-server echo "php" rpm -qa|grep php echo "php -mysql" rpm -qa|grep php-mysql echo "openssl" rpm -qa|grep openssl echo "openssl -devel"&am
 p;nb
 sp;rpm -qa|grep openssl-devel echo "kernel -source" rpm -qa|grep kernel-source echo "perl" rpm -qa|grep perl echo "perl -CPAN" rpm -qa|grep perl-CPAN echo "cvs" rpm -qa|grep cvs echo "bison" rpm -qa|grep bison echo "ncurses -devel" rpm -qa|grep ncurses-devel echo "audiofile -devel" rpm -qa|grep audiofile-devel echo "-" Roman Zhovtulya <[EMAIL PROTECTED]> wrote: 


You've got to check if you have all the required mysql libraries installed (mysql client and mysql-devel)
 
 


-Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of asterisk asteriskSent: Donnerstag, 3. März 2005 10:13To: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Asterisk realtime ,asterisk extensions not load form db.


Hello !
I try to run asterisk with real time config from database.
I use AMP to configure .
Everythig it ok , I can set new sip and iax extensions, I can see them on mysql db , as well is amp .
But these extension I cannt use in asterisk .
I have seen some new conf file 
;sip_additional.conf;iax_additional.conf;extensions_additional.conf;meetme_additional.conf
.If I reload asterisk these extensions are not load .
I have these tabele in mysql ,sip, iax, extensions.
my extconfig.conf file 
[settings]
;uncomment to load queues.conf via the db engine.;queues.conf => odbc; => ,[,table_name]
voicemail.conf => mysql,asterisk,voicemail_table   ; it is goodsip.conf => mysql,asterisk,sip_tableiax => mysql,asterisk,iaxextensions.conf => mysql,asterisk,extensions
 
How can I set asterisk to load config form database ?
Is there anyone who have the same problem ?
Thans for help !
Roby.
 
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[Asterisk-Users] Grandstream Message button

[asterisk asterisk]

Hi 
I confiured Gasnstream phone 100. Firmware ver:Program--1.0.5.16    Bootloader--1.0.0.21    HTML--1.0.0.41 ï   VOC--1.0.0.7.
 
It workes well everything. If I got a message it blinks. My voicemail no 555 .If I call 555, I can hear voicemail . But I can not configure Message Button on the phone. I set via html  Voice Mail UserID:555.  If I press message button does not work.
Can you help me ?
 
Thanks.
 
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[Asterisk-Users] Smal ofice pbx

[asterisk asterisk]

Hi ,
I have 3 ISDN BRI and 4 analog line .
I would like a smal ofice with 30 exension.
 
Can you give me it is possibile to work together isdn and analog in a same pc (PBX).
Which isdn and analog card aou recommand ? Is there any support for these card ?
 
Thaks.
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[Asterisk-Users] features.conf and CVS

[asterisk asterisk]

This is my features.conf
[general]
parkext => 700  ; What ext. to dial to park
parkpos => 701-720  ; What extensions to park calls on
context => parkedcalls  ; Which context parked calls are in
parkingtime => 45   ; Number of seconds a call can be parked for
; (default is 45 seconds)
transferdigittimeout => 3   ; Number of seconds to wait between digits
when transfering a call
courtesytone = beep ; Sound file to play to the parked caller
; when someone dials a parked call
adsipark = yes  ; if you want ADSI parking announcements

[featuremap]
blindxfer=> ##
automon => *1
atxfer => *2

I out in dial the options wWtT and I use the CVS  CVS-v1-0-08/16/05-08:47:44
If I press the # on my phone asterisk wants to trasfer the call and if
I press *1 it hung up.
any idea?
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Re: [Asterisk-Users] x100p question for incomming calls

[asterisk asterisk]

Check your extensions.conf on the context setted on zapata.conf
probably you have the command answer you should remove it.

On 8/16/05, Hubert Hoefsloot <[EMAIL PROTECTED]> wrote:
> This must be a question asked before but can't find it so here I go:
> 
> I have a Asterisk box connected, thou a x100p, to a PSTN PBX. When we
> get a incomming call on that PBX the phones in the office wil ring and
> there will also be a ring signal on the x100p. At my current
> configuration the call wil be answered by the Asterisk box.
> 
> Is it possible to configure Asterisk to not answer the line and only
> "extend" the ring signal to the callgroup?
> 
> TIA
> 
> Hubert
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[asterisk-users] DTMF Detection Problems with certain phones incoming zap channels

[asterisk asterisk]

Hello,

I'm having a problem with the autoattendant. It won't recognize the
DTMF signals from certain  people that call in. I have relaxed DTMF,
upgraded Asterisk from 1.2 to 1.2.12 to 1.2.12.1 as well as the zaptel
drivers. I have stopped X from running then only thing I didn't do
that was on Digium's support website was to reconpile vpmdtmfsupport
to 0 in wctdm24xxp.c or wct4xxp.c

Extansions.conf

[incoming]
exten=>s,1,Answer
exten=>s,2,Wait,1
exten=>s,3,Set(TIMEOUT(digit)=3)
exten=>s,4,Set(TIMEOUT(response)=10)
exten=>s,5,Background(welcome)
;exten=>s,2,WaitExten
exten=>i,1,Playback(invalid)
exten=>i,n,Goto(incoming,s,1)
exten=>6,1,Set(TIMEOUT(digit)=6)
exten=>6,2,Set(TIMEOUT(response)=10)
exten=>6,3,Background(dir-intro)
;exten=>6,2,WaitExten
exten=>0,1,Dial(SIP/12|20)
exten=>0,2,VoiceMail(u0)
exten=>t,1,Goto(incoming,s,1)
exten=>h,1,Hangup()
exten=>627,1,Background(johnis)
exten=>627,2,Goto(mastro,10,1)
exten=>372,1,Background(fradeis)
exten=>372,2,Goto(frade,11,1)
exten=>386,1,Background(eileenis)
exten=>386,2,Goto(eileen,12,1)
exten=>10,1,Goto(mastro,10,1)
exten=>11,1,Goto(frade,11,1)
exten=>12,1,Goto(eileen,12,1)
exten=>13,1,Goto(conference,13,2)
exten=>8500,1,VoiceMailMain
exten=>17,1,Dial(SIP/s12|20)

[internal]
exten=>10,1,Goto(mastro,10,1)
exten=>11,1,Goto(frade,11,1)
exten=>12,1,Goto(eileen,12,1)
exten=>13,1,Goto(conference,13,1)
exten=>110,1,VoiceMail(u10)
exten=>111,1,VoiceMail(u11)
exten=>112,1,VoiceMail(u12)

exten=>_NXX,1,Dial,Zap/g1/w${EXTEN}w
exten=>_NXX,103,Congestion
exten=>_NXX,104,Hangup()

exten=>_1NXXNXX,1,Dial,Zap/G1/w${EXTEN}w
exten=>_1NXXNXX,103,Congestion
exten=>_1NXXNXX,104,Hangup()

exten=>_011.,1,Dial,Zap/G1/w${EXTEN}w
exten=>_011.,103,Congestion
exten=>_011.,104,Hangup()

exten=>8500,1,VoiceMailMain

[mastro]
exten=>10,1,Dial(SIP/10|20)
exten=>10,2,Set(TIMEOUT(digit)=3)
exten=>10,3,Set(TIMEOUT(response)=10)
exten=>10,4,Background(john)
;exten=>10,3,WaitExten
exten=>10,5,VoiceMail(u10)
exten=>10,101,VoiceMail(b10)
exten=>1,1,VoiceMail(u10)
exten=>2,1,Goto(eileen,12,1)
exten=>3,1,Dial(Zap/G1/w19175450294w|10)
exten=>i,1,Playback(invalid)
exten=>i,n,Goto(mastro,10,2)
exten=>t,1,Congestion(5)
exten=>h,1,Hangup()
exten=>8500,1,VoiceMailMain

[frade]
exten=>11,1,Dial(SIP/11|20)
exten=>11,2,Dial(SIP/s11|20)
exten=>11,3,Set(TIMEOUT(digit)=3)
exten=>11,4,Set(TIMEOUT(response)=10)
exten=>11,5,Background(manny)
;exten=>11,4,WaitExten
exten=>11,6,VoiceMail(u11)
exten=>11,101,VoiceMail(b11)
exten=>1,1,VoiceMail(u11)
exten=>2,1,Dial(Zap/G1/w15165325627w|10)
exten=>i,1,Playback(invalid)
exten=>i,n,Goto(frade,11,3)
exten=>t,1,Congestion(5)
exten=>h,1,Hangup()
exten=>8500,1,VoiceMailMain

[eileen]
exten=>12,1,Dial(SIP/12|20)
exten=>12,2,VoiceMail(u12)
exten=>12,101,VoiceMail(b12)
exten=>t,1,Congestion(5)
exten=>h,1,Hangup()
exten=>8500,1,VoiceMailMain

[conference]
exten=>13,1,Dial(SIP/13|20)
exten=>t,1,Congestion(5)
exten=>h,1,Hangup()
exten=>8500,1,VoiceMailMain

Zapata.conf

[channels]
signalling=fxs_ks
context=incoming
language=en
rxgain=25
txgain=0
usecallerid=no
;callprogress=yes
;hanguponpolarityswitch=yes
;callerid=asreceived
;cidsignalling=bell
;cidstart=ring
echocancel=128
echotraining=800
echocancelwhenbridged=yes
transfer=yes
immediate=yes
;relaxdtmf=yes
;busydetect=yes
;busycount=5
group=1
channel=>1-4
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Re: [asterisk-users] DTMF Detection Problems with certain phones incoming zap channels

[asterisk asterisk]

I did turn it on and off as it does not seem to make a difference. On 9/19/06, Jay R. Ashworth <[EMAIL PROTECTED]
> wrote:On Tue, Sep 19, 2006 at 05:29:17PM -0400, asterisk asterisk wrote:> I'm having a problem with the autoattendant. It won't recognize the
> DTMF signals from certain  people that call in. I have relaxed DTMF,Are you sure?> Zapata.conf>> [channels]> signalling=fxs_ks[ ... ]> ;relaxdtmf=yesI'm no expert, but it looks like you haven't.  Unless you did earlier,
and turned it back off.  Or I'm an idiot.  :-)Cheers-- jra--Jay R. Ashworth[EMAIL PROTECTED]Designer  Baylink RFC 2100
Ashworth & AssociatesThe Things I Think'87 e24St Petersburg FL USA  http://baylink.pitas.com +1 727 647 1274"That's women for you; you divorce them, and 10 years later,
  they stop having sex with you."  -- Jennifer Crusie; _Fast_Women--Bandwidth and Colocation provided by Easynews.com
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[asterisk-users] Goggle voice incoming dialplan

[asterisk asterisk]

Hi,

I am a question to handle incoming goggle voice. I have put several GV
accounts into the jabber.conf. How can I direct different accounts to
different extensions?

Help with example is much appreciate

Thanks,

CK
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[asterisk-users] Web based call back

[asterisk asterisk]

Hi,

I am looking for a simple solution to do this.

I wish to have the user to enter their preferred method of connection i.e.
for the cheapest solution to their desktop phone or mobile phone, then plan
callfile based on the number that user provided and dial to the user.

Any suggestions?

CK
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Re: [asterisk-users] Goggle voice incoming dialplan

[asterisk asterisk]

Thanks and will try.

On Thu, Jun 16, 2011 at 11:28 AM, Jamie A. Stapleton <
jstaple...@computer-business.com> wrote:

>
> exten => accou...@gmail.com,1,Answer()
> exten => accou...@gmail.com,n,Wait(2)
> exten => accou...@gmail.com,n,SendDTMF(1)
> exten => accou...@gmail.com,n,Dial(SIP/device1)
>
> exten => accou...@gmail.com,1,Answer()
> exten => accou...@gmail.com,n,Wait(2)
> exten => accou...@gmail.com,n,SendDTMF(1)
> exten => accou...@gmail.com,n,Dial(SIP/device2)
>
> 
>
> From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk asterisk
> Sent: Wednesday, June 15, 2011 11:24 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Goggle voice incoming dialplan
>
>
> Hi,
>
> I am a question to handle incoming goggle voice. I have put several GV
> accounts into the jabber.conf. How can I direct different accounts to
> different extensions?
>
> Help with example is much appreciate
>
> Thanks,
>
> CK
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   http://www.asterisk.org/hello
>
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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] Goggle voice incoming dialplan

[asterisk asterisk]

Do anyone know how to receiving incoming call from GV number associated with
an non gmail.com account? I have custom domains under google and would like
to receiving calls via asterisk.

The google chat function is missing in these GV accounts.

On Thu, Jun 16, 2011 at 11:30 AM, asterisk asterisk wrote:

> Thanks and will try.
>
>
> On Thu, Jun 16, 2011 at 11:28 AM, Jamie A. Stapleton <
> jstaple...@computer-business.com> wrote:
>
>>
>> exten => accou...@gmail.com,1,Answer()
>> exten => accou...@gmail.com,n,Wait(2)
>> exten => accou...@gmail.com,n,SendDTMF(1)
>> exten => accou...@gmail.com,n,Dial(SIP/device1)
>>
>> exten => accou...@gmail.com,1,Answer()
>> exten => accou...@gmail.com,n,Wait(2)
>> exten => accou...@gmail.com,n,SendDTMF(1)
>> exten => accou...@gmail.com,n,Dial(SIP/device2)
>>
>> ____
>>
>> From: asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk asterisk
>> Sent: Wednesday, June 15, 2011 11:24 PM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: [asterisk-users] Goggle voice incoming dialplan
>>
>>
>> Hi,
>>
>> I am a question to handle incoming goggle voice. I have put several GV
>> accounts into the jabber.conf. How can I direct different accounts to
>> different extensions?
>>
>> Help with example is much appreciate
>>
>> Thanks,
>>
>> CK
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>   http://www.asterisk.org/hello
>>
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>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
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Re: [asterisk-users] Goggle voice incoming dialplan

[asterisk asterisk]

Can this non gmail.com GV number be terminated at some sip accounts so that
I can bridge to it via asterisk as client?

On Fri, Jun 17, 2011 at 11:48 AM, William Stillwell <
will...@stillwellsoft.com> wrote:

> Only GV numbers that can terminate to a Google Chat Account can be
> connected directly to asterisk.
>
> ** **
>
> Otherwise you will need to get a free SIP Account, and route calls to it.*
> ***
>
> ** **
>
> ** **
>
> ** **
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *asterisk asterisk
> *Sent:* Thursday, June 16, 2011 11:39 PM
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Goggle voice incoming dialplan
>
> ** **
>
> Do anyone know how to receiving incoming call from GV number associated
> with an non gmail.com account? I have custom domains under google and
> would like to receiving calls via asterisk.
>
> The google chat function is missing in these GV accounts.
>
> On Thu, Jun 16, 2011 at 11:30 AM, asterisk asterisk 
> wrote:
>
> Thanks and will try.
>
> ** **
>
> On Thu, Jun 16, 2011 at 11:28 AM, Jamie A. Stapleton <
> jstaple...@computer-business.com> wrote:
>
>
> exten => accou...@gmail.com,1,Answer()
> exten => accou...@gmail.com,n,Wait(2)
> exten => accou...@gmail.com,n,SendDTMF(1)
> exten => accou...@gmail.com,n,Dial(SIP/device1)
>
> exten => accou...@gmail.com,1,Answer()
> exten => accou...@gmail.com,n,Wait(2)
> exten => accou...@gmail.com,n,SendDTMF(1)
> exten => accou...@gmail.com,n,Dial(SIP/device2)
>
> 
>
> From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk asterisk
> Sent: Wednesday, June 15, 2011 11:24 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Goggle voice incoming dialplan
>
>
>
> Hi,
>
> I am a question to handle incoming goggle voice. I have put several GV
> accounts into the jabber.conf. How can I direct different accounts to
> different extensions?
>
> Help with example is much appreciate
>
> Thanks,
>
> CK
>
> 
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
> ** **
>
> ** **
>
> --
> _
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Re: [asterisk-users] Web based call back

[asterisk asterisk]

Thanks. Will need some time to look into.


On Thu, Jun 16, 2011 at 3:56 PM, Tzafrir Cohen wrote:

> On Thu, Jun 16, 2011 at 11:26:21AM +0800, asterisk asterisk wrote:
> > Hi,
> >
> > I am looking for a simple solution to do this.
> >
> > I wish to have the user to enter their preferred method of connection
> i.e.
> > for the cheapest solution to their desktop phone or mobile phone, then
> plan
> > callfile based on the number that user provided and dial to the user.
> >
> > Any suggestions?
>
> But doing so *is* simple. See a simple example attached. It relies on an
> assumption that the origination IP address authenticates a user and also
> the user's location (specifically: the phone).
>
> You would probably need your own schema for that. But the actual dialing
> is very simple.
>
> --
>   Tzafrir Cohen
> icq#16849755  jabber:tzafrir.co...@xorcom.com
> +972-50-7952406   mailto:tzafrir.co...@xorcom.com
> http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir
>
> --
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Re: [asterisk-users] Goggle voice incoming dialplan

[asterisk asterisk]

Could you elaborate on how you can associate those non-gmail accounts with
gchat account?

On Fri, Jun 17, 2011 at 2:38 PM, Warren Selby  wrote:

> On Thu, Jun 16, 2011 at 10:58 PM, asterisk asterisk 
> wrote:
>
>> Can this non gmail.com GV number be terminated at some sip accounts so
>> that I can bridge to it via asterisk as client?
>>
>>
> Yes, I've setup some GV numbers on my google apps accounts (@selbytech.com,
> for example), and associated those with gchat accounts (
> wcse...@selbytech.com), and successfully received calls on my asterisk
> using this solution.
>
> --
> Thanks,
> --Warren Selby, dCAP
> http://www.SelbyTech.com <http://www.selbytech.com>
>
>
> --
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[asterisk-users] error in GUI access

[asterisk asterisk]

I have this error after upgrading to 1.8.4.4 on my centos 5.6 32it

When using GUI to access, I got this error

*** glibc detected *** /usr/sbin/asterisk: double free or corruption
(!prev): 0x0919c070 ***

The server cannot be connected via GUI and the asterisk CLI dropped and exit
into linux command line.

Appreciate if help can be provided

CK
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Re: [asterisk-users] error in GUI access

[asterisk asterisk]

Hi,

I did not find any file with a or i with your suggested commands.

Any other clues?

CK

On Fri, Jul 1, 2011 at 6:23 PM, A J Stiles wrote:

> On Friday 01 Jul 2011, asterisk asterisk wrote:
> > I have this error after upgrading to 1.8.4.4 on my centos 5.6 32it
> >
> > When using GUI to access, I got this error
> >
> > *** glibc detected *** /usr/sbin/asterisk: double free or corruption
> > (!prev): 0x0919c070 ***
> >
> > The server cannot be connected via GUI and the asterisk CLI dropped and
> > exit into linux command line.
>
> Ooo-er.  Last time I got an error like this, it turned out that the box had
> been compromised with a rootkit.
>
> Luckily, most rootkits give themselves away in trying to make themselves
> hard
> to detect / remove:  first they replace some system utilities  (which, on
> Debian, also breaks colour directory listings)  with specially munged ones
> (for instance, an ls command that will deliberately not show any of the
> rootkit's own extra files; a ps that will not show the extra processes; a
> netstat that will not show the rootkit's network connections; and so forth)
> and then they set the extended attributes on the new files to prevent them
> from being overwritten.  So checking extended attributes can give you a
> clue
> that all is not well.
>
> Try
>
> # lsattr /bin
> # lsattr /usr/bin
> # lsattr /sbin
> # lsattr /usr/sbin
>
> All files should have a row of - signs in the left hand column.  Any "a"
> or "i" in a file's attributes indicates that the file has had its extended
> attributes modified, and you should be suspicious.
>
> Note:  ignore any errors such as "lsattr: Operation not supported While
> reading flags on /bin/nc"  (this just means the file is a symbolic link,
> and
> these don't have extended attributes).
>
> --
> AJS
>
> Answers come *after* questions.
>
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[asterisk-users] Chan_mobile

[asterisk asterisk]

I am encountering problem recently with the chan_mobile that the bluetooth
connection between the asterisk and my Nokia E71 mobile phone frequently
connect and disconnect within seconds. As a result, I can't make any call
using Mobile/E71/{exten:2}.

Any suggested cause?
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[asterisk-users] DTMF problem

[asterisk asterisk]

>From time to time, I have a DTMF problem. The asterisk cannot recognize my
handset key pressed if I press 9 to start with. However, if I press with 6,
it is ok.

On the other hand, if DMTF key is generated from softphone, it is ok.

My dialplan is as follow

exten => 1002,1,Answer
exten => 1002,n,Wait(2)
exten => 1002,n,Background(thank-you-for-calling)
exten => 1002,n,Background(vm-enter-num-to-call)
exten => 1002,n,WaitExten()
exten => 1002,n,Hangup
exten => i,1,Background(pbx-invalid)
exten => i,2,Goto(1002,1)
exten => t,1,Background(vm-goodbye)
exten => t,2,Hangup

Thanks for the help in advance.
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[asterisk-users] Questions on Dahdi

[asterisk asterisk]

I have naive question. I do not have any hardware on my asterisk host. All I
have are either SIP trunk for DID or hardware ATA which bridges the asterisk
to PSTN. Do I need Dahdi install? Do i have ztdummy for timing issue? I
encounter problem in this when I try to install Dahdi latest but I found it
is not running, Instead it runs when service starts but I can't find its
status when I type in service dahdi status.

I am using Asterisk 1.8.7 on centos 5.7 32 bit.

CK
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[asterisk-users] Questions on IAX client

[asterisk asterisk]

Hi,

I used to use Zoiper IAX to connect to my asterisk server from remote site.
On asterisk CLI, I can see my zoiper client registered and stay on line.
HOwever, I don't know why now I can't call this client. It always show up as
"Unable to create channel IAX2 (Cause 20 Unknown)

I am using Asterisk 1.8.7.1

CK
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Re: [asterisk-users] Skype For Asterisk (SFA)

[asterisk asterisk]

I can tell you that siptosis is workable but the support has been dropped
recently as well.

It is a great program and especially the paid version with trunk builder
i.e. you can have multiple skype instances

On Wed, Nov 16, 2011 at 8:01 PM, Abdul Basit  wrote:

> Any has Skype For Asterisk (SFA) license.
>
> http://www.digium.com/en/products/software/skypeforasterisk.php
>
> PLEASE NOTE: Skype for Asterisk is no longer available for sale. Skype for
> Asterisk will be supported for two more years, until July 26, 2013.
>
> I want to test this thing. Any Idea. any free solution.
>
> there is one http://nerdvittles.com/index.php?p=784
>
> Tying to test but dont know if its workable or not.
>
> I will appreciate if any one can share his testing/implementation.
>
> --
> Regards,
>
> Abdul Basit | +92 32 1416 4196 | +92 30 0841 1445
>
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Re: [asterisk-users] Connecting Skype to Asterisk

[asterisk asterisk]

The only inexpensive way is to get siptosis but the developer has stopped
the support and upgrade unfortunately. I have been using it for two years
or more.

Excellent quality and works very well

On Sat, Oct 13, 2012 at 5:17 AM, Philip Bennefall wrote:

> From what I gather, it costs extra for each channel even for direct Skype
> to Asterisk calls. Since my plan was to use this for business purposes, I'd
> need at least something like 30 channels which would be way out of my
> monthly budget unfortunately.
>
> Kind regards,
>
> Philip Bennefall
> - Original Message - From: "Duncan Turnbull" <
> dun...@e-simple.co.nz>
> To: ; "Asterisk Users Mailing List - Non-Commercial
> Discussion" 
> 
> >
> Cc: 
> Sent: Friday, October 12, 2012 11:08 PM
> Subject: Re: [asterisk-users] Connecting Skype to Asterisk
>
>
>
>
> On 13/10/2012, at 7:54 AM, Christopher Harrington  wrote:
>
>  On Fri, Oct 12, 2012 at 1:44 PM, Philip Bennefall 
>> wrote:
>>
>>> Hi all,
>>>
>>> I have an Asterisk PBX under development, that I would like to link to a
>>> Skype account if possible. The idea is that people would call a
>>> particular
>>> Skype username, and be redirected to my SIP and through that to
>>> Asterisk. Is
>>> this doable? I have looked around and saw the Skype for Asterisk driver,
>>> but
>>> of course that has been discontinued. Are there any other options? I
>>> would
>>> prefer not to have to go through the regular PSTN telephone network but
>>> directly from Skype to Asterisk via SIP. If you have any tips on how to
>>> configure my sip.conf to get this working, this would also be highly
>>> appreciated.
>>>
>>>
>> It looks like this is what you want:
>> http://www.skype.com/intl/en/**business/skype-connect/
>>
>>  This is pretty straight forward to use for inbound skype business user
> names and outbound either to pstn, skype numbers are a little more to setup
>
> There is a monthly cost but its not much and if you have skype users out
> there its a good way for them to connect in
>
>
>> --
>> -Chris Harrington
>> ACSDi Office: 763.559.5800
>> Mobile Phone: 612.326.4248
>>
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>
>
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[asterisk-users] Motif/XMPP for Google Voice

[asterisk asterisk]

Dear all,

I wish to ask a question of the new Motif Channel in asterisk 11.

I successfully compile the binary and run without error. However, when
dialing out, no external connection  only ringing.

Any suggestions?

I follow the set up in wiki

CK
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[asterisk-users] Random crash of the machine ? due to Asterisk 11

[asterisk asterisk]

I experience random crash of machine (full hang, requiring a hard reset)
after trying to test run Asterisk 11.

The machine is a centos 5.8 32 bits pc with 1G ram. Asterisk is compiled
from the source and no other software has been installed

Anyone experience similar situation?
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Re: [asterisk-users] Random crash of the machine ? due to Asterisk 11

[asterisk asterisk]

No, I put it in Xen VPS with Centos 5.8. Only things I added are skype
support using siptosis and java.

Asterisk 11 is complied with no issue, siptosis and skype call no issues.
But hangs unexpectedly.

Any clue is welcome?

On Thu, Nov 8, 2012 at 2:10 PM, Julian Lyndon-Smith wrote:

> are you running dahdi ?
>
> We're using 11, System uptime: 3 weeks, 22 hours, 42 minutes, 19
> seconds, 231452 calls processed
>
> We did, however, have a problem with dahdi freezing the machine
>
> Julian
>
> On 7 November 2012 22:32, asterisk asterisk  wrote:
> > I experience random crash of machine (full hang, requiring a hard reset)
> > after trying to test run Asterisk 11.
> >
> > The machine is a centos 5.8 32 bits pc with 1G ram. Asterisk is compiled
> > from the source and no other software has been installed
> >
> > Anyone experience similar situation?
> >
> > --
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> >http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> --
> Julian Lyndon-Smith
> IT Director, Dot R Limited
>
> "I don’t care if it works on your machine!  We are not shipping your
> machine!”
>
> The kangaroo dances: http://www.youtube.com/watch?v=MAWl5iYOaUg
>
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[asterisk-users] watchdog like functions

[asterisk asterisk]

I wish to ask if there is way to keep IAX trunk connection up. I have a
small server on Xen VPS but notice that my IAX trunk drops after some time.

I understand there is cron job to function as sip watchdog.

My asterisk is 11.0.1

Thanks for suggestions.

CK
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[asterisk-users] DTMF not detected, time out

[asterisk asterisk]

Hi,

I encounter this problem recently after quite some months of my asterisk.

I have a SIP trunk for dial in and out.
When dial-in, it matches the callerid number and decides. If matched, it
will either go into an a very brief IVR. The IVR allows caller to dial
internal extension.
All along it is working well both from outside call and internal users.
Now for unknown reason, it fails with a timeout and hangup. It is the only
message I can see at the console.
But internal user can do this without any problem.

Appreciate if someone can help.

CK
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Re: [asterisk-users] DTMF not detected, time out

[asterisk asterisk]

In the past it was set as auto and worked. I change to RFC2833 but did not
work.

How can I check further?



On Wed, Feb 16, 2011 at 10:30 AM, Faisal Hanif  wrote:

> Check if dtmfmode is properly set on SIP trunk ask with your carrier which
> dmtfmode they support.
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *asterisk asterisk
> *Sent:* Wednesday, February 16, 2011 5:39 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] DTMF not detected, time out
>
>
>
> Hi,
>
> I encounter this problem recently after quite some months of my asterisk.
>
> I have a SIP trunk for dial in and out.
> When dial-in, it matches the callerid number and decides. If matched, it
> will either go into an a very brief IVR. The IVR allows caller to dial
> internal extension.
> All along it is working well both from outside call and internal users.
> Now for unknown reason, it fails with a timeout and hangup. It is the only
> message I can see at the console.
> But internal user can do this without any problem.
>
> Appreciate if someone can help.
>
> CK
>
> --
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Re: [asterisk-users] DTMF not detected, time out

[asterisk asterisk]

It is somehow back to normal. Nothing change. May be the sip provider
problem. However, it lasts for quite a while.

Thanks

On Wed, Feb 16, 2011 at 12:04 PM, Faisal Hanif  wrote:

> You can also append add dtmf logging to cosole and see if dtmf is coming
> from carrier.
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *asterisk asterisk
> *Sent:* Wednesday, February 16, 2011 8:58 AM
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] DTMF not detected, time out
>
>
>
> In the past it was set as auto and worked. I change to RFC2833 but did not
> work.
>
> How can I check further?
>
>
> On Wed, Feb 16, 2011 at 10:30 AM, Faisal Hanif  wrote:
>
> Check if dtmfmode is properly set on SIP trunk ask with your carrier which
> dmtfmode they support.
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *asterisk asterisk
> *Sent:* Wednesday, February 16, 2011 5:39 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] DTMF not detected, time out
>
>
>
> Hi,
>
> I encounter this problem recently after quite some months of my asterisk.
>
> I have a SIP trunk for dial in and out.
> When dial-in, it matches the callerid number and decides. If matched, it
> will either go into an a very brief IVR. The IVR allows caller to dial
> internal extension.
> All along it is working well both from outside call and internal users.
> Now for unknown reason, it fails with a timeout and hangup. It is the only
> message I can see at the console.
> But internal user can do this without any problem.
>
> Appreciate if someone can help.
>
> CK
>
>
> --
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>
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[asterisk-users] Problem in dialing out

[asterisk asterisk]

I have a sip trunk connecting to a huawei softx3000. At the moment, I can
register and dial in.

However, peer status shows not reachable

sip show peer as follow

  * Name   : cmphone
  Secret   : 
  MD5Secret: 
  Remote Secret: 
  Context  : from-cmphone
  Subscr.Cont. : device-hints
  Language :
  AMA flags: Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup:
  Pickupgroup  :
  MOH Suggest  :
  Mailbox  :
  VM Extension : asterisk
  LastMsgsSent : 32767/65535
  Call limit   : 0
  Max forwards : 0
  Dynamic  : No
  Callerid : "" <>
  MaxCallBR: 384 kbps
  Expire   : -1
  Insecure : port,invite
  Force rport  : Yes
  ACL  : No
  DirectMedACL : No
  T.38 support : No
  T.38 EC mode : Unknown
  T.38 MaxDtgrm: -1
  DirectMedia  : Yes
  PromiscRedir : No
  User=Phone   : No
  Video Support: No
  Text Support : No
  Ign SDP ver  : No
  Trust RPID   : No
  Send RPID: No
  Subscriptions: Yes
  Overlap dial : Yes
  Outb. proxy  : 202.0.179.3
  DTMFmode : rfc2833
  Timer T1 : 500
  Timer B  : 32000
  ToHost   : 202.0.179.3
  Addr->IP : 202.0.179.3:5060
  Defaddr->IP  : (null)
  Prim.Transp. : UDP
  Allowed.Trsp : UDP
  Def. Username: 852350xx
  SIP Options  : 100rel
  Codecs   : 0xe (gsm|ulaw|alaw)
  Codec Order  : (alaw:20,ulaw:20,gsm:20)
  Auto-Framing :  No
  100 on REG   : No
  Status   : UNREACHABLE
  Useragent:
  Reg. Contact :
  Qualify Freq : 6 ms
  Sess-Timers  : Accept
  Sess-Refresh : uas
  Sess-Expires : 1800 secs
  Min-Sess : 90 secs
  RTP Engine   : asterisk
  Parkinglot   :
  Use Reason   : No
  Encryption   : No

In sip.conf

I have

register = 852350x:secret@202.0.179.3

[cmphone]
type = friend
host = 202.0.179.3
secret = secret
username = 852350x
context = from-cmphone
dtmfmode = rfc2833
outboundproxy = 202.0.179.3
caninvite=no
insecure = port,invite
nat = yes

When debug is on, the error message is


<--- SIP read from UDP:202.0.179.3:5060 --->
SIP/2.0 504 Server Time-out
From: "asterisk" ;tag=as2d14b9ec
To: ;tag=6b0704d0
CSeq: 102 OPTIONS
Call-ID: 17e0315c21d7dbc10e8c185740e21...@sip.x.xxx
Via: SIP/2.0/UDP
14.xxx.xxx.xxx:5060;branch=z9hG4bK3646eaf2;received=14.xxx.xxx.xxx;rport=5060
Content-Length: 0

Any help is appreciate.

CK
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Re: [asterisk-users] Using voice modem as poor man's FXO in Asterisk 1.8

[asterisk asterisk]

HI,

My understanding is that the modem won't work. I believe asterisk does not
support.

I wonder why you do not have the built in ethernet in your motherboard. You
can spare your PCI slot for a proper FXO card and use USB-to-ethernet

For a PCI FXO card, the cheapest will be X100 but be aware of the quality
and compatibility. Or a better choice will be  TDM400

Other alternative:
Get a USB-FXO from Sangoma, expensive
Get a working SPA3000 as FXO --- cheapest I believe
Get a OBi100, out of stock at the moment. I also want to try

Hope this is of help to you

CK



On Mon, Feb 28, 2011 at 10:12 AM, Stuart Longland wrote:

> Hi all,
>
> I've tried researching this, and so far, have struggled to find any
> contemporary information on the issue, so I do apologise if asking this
> irritates people who have answered this before.
>
> I have managed to set up Asterisk 1.8 on the web server here.  I have
> two softphones (Ekiga) able to communicate with it.  So far so good.
> I'm now curious to see if I can link it with the PSTN phone line here.
>
> The web server in question is an Intel Atom system with a Mini-ITX
> motherboard.  Its one and only PCI slot is occupied by a PCI ethernet
> card.  So FXO card is not an option even if it were within budget.
>
> My options therefore look to be an external FXO device of some
> description (Ethernet or USB), or to use a voice modem.  I fear external
> FXOs are going to be even more expensive than internal FXO cards.
>
> Now, I have here an old Maestro JetStream 56k modem here that does
> amongst other things, voice comms, and I have used it in the past as a
> telephone by plugging a headset into the front of it (and it was full
> duplex too if I recall correctly).  I have also used it as an answering
> machine, with the audio being transmitted digitally over the RS232
> link.  So that to me suggests it is possible to get audio in to and out
> of the modem, either via a sound card or using the serial port.  The web
> server has a sound card too (hard not to buy a motherboard with one
> these days).
>
> Apart from the lack of any hardware signal processing, it seems all the
> components are there.  The server isn't particularly heavily loaded, and
> thus I see no reason why the machine wouldn't theoretically be able to
> handle the DSP in software … I've seen lesser hardware do quite
> sophisticated DSP in real-time.
>
> Now, I've hunted high and low for where this is configured.  Some
> mailing list threads point me to the nonexistant
> /etc/asterisk/modems.conf.  One points me to /etc/asterisk/phone.conf,
> but nothing there jumps out at me as being an obvious means for
> configuring a modem — nor can I find where it's documented on the
> Asterisk wiki.
>
> Where abouts should I look for documentation on configuring these modules?
>
> Regards,
> --
> Stuart Longland (aka Redhatter, VK4MSL)  .'''.
> Gentoo Linux/MIPS Cobalt and Docs Developer  '.'` :
> . . . . . . . . . . . . . . . . . . . . . .   .'.'
> http://dev.gentoo.org/~redhatter :.'
>
> I haven't lost my mind...
>  ...it's backed up on a tape somewhere.
>
>
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Re: [asterisk-users] Using voice modem as poor man's FXO in Asterisk 1.8

[asterisk asterisk]

I totally agreed with Leif Madsen that viable options are available and time
and effort spent on winmodem should be carefully considered.

My system also works with an ATA as PSTN gateway and VOIP SIP provider for
DID and inbound/outbound service. It will save time much more time and
effort while keep up the productivity.

CK

On Wed, Mar 2, 2011 at 8:53 AM, Leif Madsen wrote:

> On 11-02-27 09:12 PM, Stuart Longland wrote:
>
>> I've tried researching this, and so far, have struggled to find any
>> contemporary information on the issue, so I do apologise if asking this
>> irritates people who have answered this before.
>>
>> I have managed to set up Asterisk 1.8 on the web server here.  I have
>> two softphones (Ekiga) able to communicate with it.  So far so good.
>> I'm now curious to see if I can link it with the PSTN phone line here.
>>
>
> There are several very good answers in this thread, and I suggest reading
> them. However, if hardware costs are the issue, then my recommendation is
> always to look at a SIP connection from an ITSP as your connection to the
> PSTN. The costs are nearly trivial (at least in Canada here you can have a
> DID for inbound calls for something around $5 a month, with termination
> costs in the range of 1c/min -- in other commonwealth countries I presume
> the costs are similar?).
>
> My bill rarely rises above $20 a month, and I use my phone a lot.
> (Business, personal, and 3 DID numbers are included in that cost.)
>
> I highly suggest you spend your time and money elsewhere, rather than
> chasing the dragon that seems to be winmodem FXO connectivity.
>
> If you absolutely must have hardware, then I suggest you start with used
> ATA (analog telephony adapters) that can be found on eBay, kijiji,
> craigslist, or any other assorted websites.
>
> Leif.
>
>
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[asterisk-users] Help on incoming

[asterisk asterisk]

Hi,

I am using IAXmodem + hylafax to do outgoing and incoming fax with asterisk.
I wonder how to write a dialplan to differentiate incoming call or fax.
I am sharing a line for both voice and fax.

CK
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Re: [asterisk-users] [Opinion Request] SIP phones that work well with Asterisk

[asterisk asterisk]

Siemens IP A580 works fairly well.

2011/3/9 Sébastien BERGER 

> My personal experience :
> Corded : Snom 3xx and 8xx, Aastra 6731i, 6755i and 6757i and Polycom IP330,
> IP650.
> DECT : Siemens C470, Polycom Kirk KWS300 and 600v3
>
> Work well
>
> AB2L
> +33 (0)367100783
> sebast...@ab2l.eu
>
>
> Le 09/03/2011 13:09, John Kosmas a écrit :
>
>  Grandstream GXV3140 Multimedia IP Phones. Fully SIP capable. work well
>> with Asterisk.
>>
>>
>> On Wed, 2011-03-09 at 16:31 +0530, Raj Mathur wrote:
>>
>>> Hi,
>>>
>>> Would you recommend some standalone SIP phones that work well with
>>> Asterisk?  Personal experience preferred.
>>>
>>> Thanks,
>>>
>>> -- Raj
>>>
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>>
>>
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>
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[asterisk-users] wrong time retrieved from system command

[asterisk asterisk]

${STRFTIME(${EPOCH},GMT+8,%G%m%d-%H%M%S)}

I use the above command to get the system date and time

it returns 20110321-034329

but it is exactly 8 hours early than the system time when I type date in
linux terminal

Mon Mar 21 19:43:35 HKT 2011

I am looking for help.

CK
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Re: [asterisk-users] wrong time retrieved from system command

[asterisk asterisk]

With gmt+8, the result is

-Mon Mar 21 13:47:59 2011

For linux server timezone I set it via webmin and /etc/localtime is my
timezone file i.e. HK at GMT+8


On Mon, Mar 21, 2011 at 9:36 PM, Tzafrir Cohen wrote:

> On Mon, Mar 21, 2011 at 09:23:29PM +0800, aster...@ck-lee.com wrote:
> > The timezone is correct. I have double checked.
>
> How did you check it? What did you check, specifically?
>
> --
>   Tzafrir Cohen
> icq#16849755  jabber:tzafrir.co...@xorcom.com
> +972-50-7952406   mailto:tzafrir.co...@xorcom.com
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>
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Re: [asterisk-users] wrong time retrieved from system command

[asterisk asterisk]

Thanks,

You give me the right answer.

On Mon, Mar 21, 2011 at 10:19 PM, Barry Miller
wrote:

> On Mon, Mar 21, 2011 at 07:45:37PM +0800, asterisk asterisk wrote:
> > ${STRFTIME(${EPOCH},GMT+8,%G%m%d-%H%M%S)}
> >
> > I use the above command to get the system date and time
> >
> > it returns 20110321-034329
> >
> > but it is exactly 8 hours early than the system time when I type date in
> > linux terminal
> >
> > Mon Mar 21 19:43:35 HKT 2011
>
> Have you tried "${STRFTIME(${EPOCH},Hongkong,%G%m%d-%H%M%S)}" ?
>
> $ date ; TZ=UTC date ; TZ=Hongkong date
> Mon Mar 21 10:13:31 EDT 2011
> Mon Mar 21 14:13:31 UTC 2011
> Mon Mar 21 22:13:31 HKT 2011
>
> --
> Barry
>
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Re: [asterisk-users] Huawei K3765 + Internet + SMS + Telephone

[asterisk asterisk]

You need a separate  Huawei USB stick to do the connection with asterisk.

Your K3765 should work with asterisk via chan_datacard.

http://wiki.e1550.mobi/doku.php?id=requirements

I have just made my K3715 works very well with asterisk.

CK

On Fri, Apr 1, 2011 at 5:45 AM, Alejandro Kauffmann  wrote:

> On 3/31/2011 3:05 PM, Michelle Konzack wrote:
>
>> Hello Hans Witvliet,
>>
>> Am 2011-03-31 22:24:50, hacktest Du folgendes herunter:
>>
>>> Hi Michelle,
>>>
>>> Perhaps i'm not understanding your question correctly.
>>> > From what i read, i seems that you got your huawei working correctly as
>>> an umts/hspa-modem, But now you want to use sms/voip directly?
>>>
>> There are some devices created by "udev" and it seems I have to tty  and
>> a sound port or something like this...
>>
>>  afaict, you can only use the voice/text-services from asterisk over the
>>> IP-layer offered by your modem.
>>>
>> Do you mean with the "tty" and the sound port?
>>
>>  If you want to use the GSM-chip directly, you need (parts of) another
>>> project: not asterisk, but openbsc. But i don't think that they are yet
>>> capable of communicating to Huawei-hardware (i have one myself)
>>>
>> I think not
>>
>> I know with FreeSWITCH it is possibel, but FreeSWITCH is not  in  Debian
>> nor is it stabel enough.
>>
>> (I have Asterisk and FreeSWITCH installed to do testing)
>>
>>  You got a working IP-connecion ontop of the underlying gsm-stuff, and
>>> have access only on anything on the ip-level, not to the protocols
>>> underneath, i think
>>>
>> Hmmm...
>>
>> Thanks, Greetings and nice Day/Evening
>> Michelle Konzack
>>
>>
>>  Look into chan_datacard.
>
> http://forge.asterisk.org/gf/project/chan_datacard/
>
> Alex
>
>
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Re: [asterisk-users] AsteriskNow updated to Centos 5.6 and DAHDI doesn't work

[asterisk asterisk]

same here. Something seriously wrong after upgrade

Don't upgrade now.

On Sun, Apr 10, 2011 at 9:32 PM, Frank Tarczynski wrote:

> My AsteriskNow box was updated to Centos 5.6 (2.6.18-238.5.1.el5) and DAHDI
> doesn't want to load. I've tried building it from the sources, but get this
> error message:
> CC [M]
> /root/Desktop/dahdi-linux-complete-2.4.1.1+2.4.1/linux/drivers/dahdi/xpp/card_bri.o
> In file included from
> /root/Desktop/dahdi-linux-complete-2.4.1.1+2.4.1/linux/drivers/dahdi/xpp/xpd.h:31,
> from
> /root/Desktop/dahdi-linux-complete-2.4.1.1+2.4.1/linux/drivers/dahdi/xpp/card_bri.c:29:
> include/linux/device.h:408: error: expected identifier or ‘(’ before
> ‘const’
> make[4]: ***
> [/root/Desktop/dahdi-linux-complete-2.4.1.1+2.4.1/linux/drivers/dahdi/xpp/card_bri.o]
> Error 1
> make[3]: ***
> [/root/Desktop/dahdi-linux-complete-2.4.1.1+2.4.1/linux/drivers/dahdi/xpp]
> Error 2
> make[2]: ***
> [_module_/root/Desktop/dahdi-linux-complete-2.4.1.1+2.4.1/linux/drivers/dahdi]
> Error 2
> make[2]: Leaving directory `/usr/src/kernels/2.6.18-238.5.1.el5-x86_64'
> make[1]: *** [modules] Error 2
> make[1]: Leaving directory
> `/root/Desktop/dahdi-linux-complete-2.4.1.1+2.4.1/linux'
> make: *** [all] Error 2
>
> The code in question is:
> static inline const char *dev_name(const struct device *dev)
> {
> return kobject_name(&dev->kobj);
> }
>
> Anybody else seen this problem? Any resolutions?
>
> Thanks
>
>
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Re: [asterisk-users] Huawei K3765 + Internet + SMS + Telephone

[asterisk asterisk]

Look at this wiki for help.

http://wiki.e1550.mobi/doku.php

For asterisk, you can use your USB stick for voice/SMS but not internet at
the same time. A separate internet connect is required per my understanding.


CK

On Sat, Apr 23, 2011 at 3:51 AM, Michelle Konzack <
linux4miche...@tamay-dogan.net> wrote:

> Hello asterisk asterisk,
>
> Am 2011-04-01 06:41:46, hacktest Du folgendes herunter:
> > You need a separate  Huawei USB stick to do the connection with asterisk.
> > Your K3765 should work with asterisk via chan_datacard.
>
> I have this now installed in my Kernel and tried to configure  asterisk,
> but if it connects, I  can not more use the UMTS IInternet connection.
>
> However, I can not get a voice connection...
>
> Is there a working config, which let me use
>
>1)  UMTS/HSPA Internet
>2)  Voice Calls
>3)  SMS
>
> at the same time?  Currently I use the  "Vodafone EasyBox 803 A"  and  I
> like to get rid of ths pig as fast as possibel!
>
> > http://wiki.e1550.mobi/doku.php?id=requirements
> > I have just made my K3715 works very well with asterisk.
> >
> > CK
>
> Thanks, Greetings and nice Weekend
>Michelle Konzack
>
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Re: [asterisk-users] Best modem for chan_datacard

[asterisk asterisk]

Huawei e180, K3715 are good to play around. Both voice and SMS are
supported.


On Fri, Apr 29, 2011 at 2:47 AM, Tiago Geada  wrote:

> I used succesfully huawei E1550
>
> On 24 April 2011 16:45, Dovid Bender  wrote:
>
>>  Hi List,
>>
>> I am looking to "play around" with chan_datacard. Any advice on the "best"
>> device to test with (that I can find on eBay) ?
>>
>> Regards,
>>
>> Dovid
>>
>>
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>>
>
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[asterisk-users] SMS callfile

[asterisk asterisk]

Hi,

I am looking for tutorial to generate a callfile so that after my program
executes, a callfile is generated and pass to asterisk to send to the
recipient.

Any suggestion?

Besides, do you know if there is a web-based GUI to send sms via asterisk?

Thanks.

CK
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Re: [asterisk-users] Soft phones.

[asterisk asterisk]

Hi,

Zoiper is a great software to have both SIP and IAX. As a beginner to
Asterisk, I find very well but to my understanding it does not have linux
version.

X-lite have both Windows and Linux but it is a bit clumsy to set up.

CK

On Fri, Jul 23, 2010 at 5:04 AM, Ronaldo Zacarias Afonso <
ronaldoafo...@gmail.com> wrote:

> Hi Ken,
>
> Can it be an IAX client?
> If so, I'd recommend KIAX. I used it once, both on Linux and Windows,
> and it worked for me.
>
> []s
> Ronaldo.
>
> On Thu, Jul 22, 2010 at 4:14 PM, Ken D'Ambrosio  wrote:
> > Hey, all.  I'm looking -- if possible -- for a decent, multi-platform
> > soft-phone.  Specifically, Linux and Windows; that way, I'll go through
> > the same issues my end users do.  I've noticed a couple (e.g., minisip,
> > which seems abandoned, and sip-communicator, which, honestly, is probably
> > a great IM client, but has a confusing interface for actual phone calls).
> > So I'm wondering if anyone has any favorites.  Failing multi-platform,
> > I'll stick with Twinkle on Linux, and gladly take suggestions for Windows
> > -- OSS if possible, but payware is acceptable.
> >
> > Thanks!
> >
> > -Ken
> >
> >
> > --
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> > dangerous content by MailScanner, and is
> > believed to be clean.
> >
> >
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Re: [asterisk-users] SIP response 500 "Server Internal Error"

[asterisk asterisk]

Hi,

I have problem in initiating an dial out call with  SIP response 500 "Server
Internal Error"

The sip debug as


  == Using SIP RTP CoS mark 5
Audio is at 113.253.226.92 port 18284
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 203.80.89.139:5060:
INVITE sip:27101...@s2hkbntel.net:5060 SIP/2.0
Via: SIP/2.0/UDP 113.253.226.92:5060;branch=z9hG4bK5b563aea;rport
Max-Forwards: 70
From: "IAX-cklee" 
>;tag=as4cffc48a
To: 
Contact: >
Call-ID: 051db26e59f7163b2458cb9e67ff5...@s2hkbntel.net
CSeq: 102 INVITE
User-Agent: Asterisk
Remote-Party-ID: "IAX-cklee" 
>;privacy=off;screen=yes
Date: Mon, 09 Aug 2010 15:03:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 241

v=0
o=root 1216883305 1216883305 IN IP4 113.253.226.92
s=Asterisk PBX 1.6.2.10
c=IN IP4 113.253.226.92
t=0 0
m=audio 18284 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
-- Called 27101...@hkbn2b

<--- SIP read from UDP:203.80.89.139:5060 --->
SIP/2.0 100 Trying
t: 
f: "IAX-cklee" 
>;tag=as4cffc48a
i: 051db26e59f7163b2458cb9e67ff5...@s2hkbntel.net
CSeq: 102 INVITE
v: SIP/2.0/UDP 113.253.226.92:5060;rport;branch=z9hG4bK5b563aea
Server: MCS5x00_3.0
k: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec
l: 0


<->
--- (9 headers 0 lines) ---

<--- SIP read from UDP:203.80.89.139:5060 --->
SIP/2.0 500 Server Internal Error
t: ;tag=301677433
f: "IAX-cklee" 
>;tag=as4cffc48a
i: 051db26e59f7163b2458cb9e67ff5...@s2hkbntel.net
CSeq: 102 INVITE
v: SIP/2.0/UDP 113.253.226.92:5060;rport;branch=z9hG4bK5b563aea
k: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec
l: 0


<->
--- (8 headers 0 lines) ---
-- Got SIP response 500 "Server Internal Error" back from 203.80.89.139
Transmitting (NAT) to 203.80.89.139:5060:
ACK sip:27101...@s2hkbntel.net:5060 SIP/2.0
Via: SIP/2.0/UDP 113.253.226.92:5060;branch=z9hG4bK5b563aea;rport
Max-Forwards: 70
From: "IAX-cklee" 
>;tag=as4cffc48a
To: ;tag=301677433
Contact: >
Call-ID: 051db26e59f7163b2458cb9e67ff5...@s2hkbntel.net
CSeq: 102 ACK
User-Agent: Asterisk
Remote-Party-ID: "IAX-cklee" 
>;privacy=off;screen=yes
Content-Length: 0

I have no idea how to make it work.

CK
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Re: [asterisk-users] SIP response 500 "Server Internal Error"

[asterisk asterisk]

I try to disable firewall but no working. I use a softphone to connect on
the same lan segment, it works. Dial in is no problem but dial out always
have this error


On Mon, Aug 9, 2010 at 11:21 PM, Danny Nicholas  wrote:

>   *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *asterisk asterisk
> *Subject:* Re: [asterisk-users] SIP response 500 "Server Internal Error"
>
>
>
> >Hi,
> >I have problem in initiating an dial out call with  SIP response 500
> "Server Internal Error"
> >The sip debug as
> 
>
>
> ><--- SIP read from UDP:203.80.89.139:5060 --->
> SIP/2.0 500 Server Internal Error
> >t: ;tag=301677433
> >f: "IAX-cklee" 
> >;tag=as4cffc48a
> >i: 051db26e59f7163b2458cb9e67ff5...@s2hkbntel.net
> >CSeq: 102 INVITE
> >v: SIP/2.0/UDP 113.253.226.92:5060;rport;branch=z9hG4bK5b563aea
> >k: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec
> >l: 0
>
> >I have no idea how to make it work.
>
> >CK
>
>
>
> It looks like your firewall is blocking 203.80.89.139?
>
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Re: [asterisk-users] SIP response 500 "Server Internal Error"

[asterisk asterisk]

I fix the problem now because of the outbound CID issues.

On Tue, Aug 10, 2010 at 6:14 AM, asterisk asterisk wrote:

> I try to disable firewall but no working. I use a softphone to connect on
> the same lan segment, it works. Dial in is no problem but dial out always
> have this error
>
>
> On Mon, Aug 9, 2010 at 11:21 PM, Danny Nicholas  wrote:
>
>>   *From:* asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] *On Behalf Of *asterisk asterisk
>> *Subject:* Re: [asterisk-users] SIP response 500 "Server Internal Error"
>>
>>
>>
>> >Hi,
>> >I have problem in initiating an dial out call with  SIP response 500
>> "Server Internal Error"
>> >The sip debug as
>> 
>>
>>
>> ><--- SIP read from UDP:203.80.89.139:5060 --->
>> SIP/2.0 500 Server Internal Error
>> >t: ;tag=301677433
>> >f: "IAX-cklee" 
>> >;tag=as4cffc48a
>> >i: 051db26e59f7163b2458cb9e67ff5...@s2hkbntel.net
>> >CSeq: 102 INVITE
>> >v: SIP/2.0/UDP 113.253.226.92:5060;rport;branch=z9hG4bK5b563aea
>> >k: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec
>> >l: 0
>>
>> >I have no idea how to make it work.
>>
>> >CK
>>
>>
>>
>> It looks like your firewall is blocking 203.80.89.139?
>>
>> --
>>
>> _
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>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
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[asterisk-users] 603 error

[asterisk asterisk]

Hi,

I have an interesting problem that the dial out via sip always generates 603
error

The following is the sip debug


Your help is appreciated.

CK
  == Using SIP RTP CoS mark 5
-- Executing [998560...@dlpn_dp1:1] Dial("SIP/6100-005b",
"SIP/13398560...@hkbn2b") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 113.253.230.26 port 11316
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 203.80.89.139:5060:
INVITE sip:13398560...@s2hkbntel.net:5060 SIP/2.0
Via: SIP/2.0/UDP 113.253.230.26:5060;branch=z9hG4bK575022bd;rport
Max-Forwards: 70
From: "ck...@mobile"

>;tag=as1d554c43
To: 
Contact: >
Call-ID: 34c9241622c72c7d26b13fdc22d95...@s2hkbntel.net
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.10
Date: Sun, 15 Aug 2010 13:47:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 241

v=0
o=root 2083113394 2083113394 IN IP4 113.253.230.26
s=Asterisk PBX 1.6.2.10
c=IN IP4 113.253.230.26
t=0 0
m=audio 11316 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
-- Called 13398560...@hkbn2b

<--- SIP read from UDP:203.80.89.139:5060 --->
SIP/2.0 100 Trying
t: 
f: "ck...@mobile" 
>;tag=as1d554c43
i: 34c9241622c72c7d26b13fdc22d95...@s2hkbntel.net
CSeq: 102 INVITE
v: SIP/2.0/UDP 113.253.230.26:5060
;received=113.253.230.70;rport;branch=z9hG4bK575022bd
Server: MCS5x00_3.0
k: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec
l: 0


<->
--- (9 headers 0 lines) ---

<--- SIP read from UDP:203.80.89.139:5060 --->
SIP/2.0 487 Request Terminated
t: ;tag=1652716799
f: "ck...@mobile" 
>;tag=as1d554c43
i: 34c9241622c72c7d26b13fdc22d95...@s2hkbntel.net
CSeq: 102 INVITE
v: SIP/2.0/UDP 113.253.230.26:5060
;received=113.253.230.70;rport;branch=z9hG4bK575022bd
k: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec
l: 0


<->
--- (8 headers 0 lines) ---
Transmitting (NAT) to 203.80.89.139:5060:
ACK sip:13398560...@s2hkbntel.net:5060 SIP/2.0
Via: SIP/2.0/UDP 113.253.230.26:5060;branch=z9hG4bK575022bd;rport
Max-Forwards: 70
From: "ck...@mobile"

>;tag=as1d554c43
To: ;tag=1652716799
Contact: >
Call-ID: 34c9241622c72c7d26b13fdc22d95...@s2hkbntel.net
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.10
Content-Length: 0


---
Scheduling destruction of SIP dialog '
34c9241622c72c7d26b13fdc22d95...@s2hkbntel.net' in 6400 ms (Method: INVITE)
  == Everyone is busy/congested at this time (1:0/0/1)
-- Executing [998560...@dlpn_dp1:2] Hangup("SIP/6100-005b", "") in
new stack
  == Spawn extension (DLPN_DP1, 998560848, 2) exited non-zero on
'SIP/6100-005b'
Really destroying SIP dialog '34c9241622c72c7d26b13fdc22d95...@s2hkbntel.net'
Method: INVITE
ns*CLI> sip set debug off
SIP Debugging Disabled
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[asterisk-users] Fwd: 603 error

[asterisk asterisk]

Hi,
I have an interesting problem that the dial out via sip always generates 603
error

The following is the sip debug


Your help is appreciated.

CK
  == Using SIP RTP CoS mark 5
-- Executing [998560...@dlpn_dp1:1] Dial("SIP/6100-005b",
"SIP/13398560...@hkbn2b") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 113.253.230.26 port 11316
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 203.80.89.139:5060:
INVITE sip:13398560...@s2hkbntel.net:5060 SIP/2.0
Via: SIP/2.0/UDP 113.253.230.26:5060;branch=z9hG4bK575022bd;rport
Max-Forwards: 70
From: "ck...@mobile"

>;tag=as1d554c43
To: 
Contact: >
Call-ID: 34c9241622c72c7d26b13fdc22d95...@s2hkbntel.net
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.10
Date: Sun, 15 Aug 2010 13:47:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 241

v=0
o=root 2083113394 2083113394 IN IP4 113.253.230.26
s=Asterisk PBX 1.6.2.10
c=IN IP4 113.253.230.26
t=0 0
m=audio 11316 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
-- Called 13398560...@hkbn2b

<--- SIP read from UDP:203.80.89.139:5060 --->
SIP/2.0 100 Trying
t: 
f: "ck...@mobile" 
>;tag=as1d554c43
i: 34c9241622c72c7d26b13fdc22d95...@s2hkbntel.net
CSeq: 102 INVITE
v: SIP/2.0/UDP 113.253.230.26:5060
;received=113.253.230.70;rport;branch=z9hG4bK575022bd
Server: MCS5x00_3.0
k: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec
l: 0


<->
--- (9 headers 0 lines) ---

<--- SIP read from UDP:203.80.89.139:5060 --->
SIP/2.0 487 Request Terminated
t: ;tag=1652716799
f: "ck...@mobile" 
>;tag=as1d554c43
i: 34c9241622c72c7d26b13fdc22d95...@s2hkbntel.net
CSeq: 102 INVITE
v: SIP/2.0/UDP 113.253.230.26:5060
;received=113.253.230.70;rport;branch=z9hG4bK575022bd
k: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec
l: 0


<->
--- (8 headers 0 lines) ---
Transmitting (NAT) to 203.80.89.139:5060:
ACK sip:13398560...@s2hkbntel.net:5060 SIP/2.0
Via: SIP/2.0/UDP 113.253.230.26:5060;branch=z9hG4bK575022bd;rport
Max-Forwards: 70
From: "ck...@mobile"

>;tag=as1d554c43
To: ;tag=1652716799
Contact: >
Call-ID: 34c9241622c72c7d26b13fdc22d95...@s2hkbntel.net
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.10
Content-Length: 0


---
Scheduling destruction of SIP dialog '
34c9241622c72c7d26b13fdc22d95...@s2hkbntel.net' in 6400 ms (Method: INVITE)
  == Everyone is busy/congested at this time (1:0/0/1)
-- Executing [998560...@dlpn_dp1:2] Hangup("SIP/6100-005b", "") in
new stack
  == Spawn extension (DLPN_DP1, 998560848, 2) exited non-zero on
'SIP/6100-005b'
Really destroying SIP dialog '34c9241622c72c7d26b13fdc22d95...@s2hkbntel.net'
Method: INVITE
ns*CLI> sip set debug off
SIP Debugging Disabled
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Re: [asterisk-users] sending sms from Asterisk server

[asterisk asterisk]

Could you share your AGI script?

CK

On Wed, Aug 18, 2010 at 5:43 AM, Johann Hoehn wrote:

> On 08/17/2010 09:00 AM, Tino wrote:
> > Hello,
> >
> > I would like to send sms to some external phone numbers from my
> > asterisk server. Is it possible to send sms via softphones like X-Lite
> > ? . Any tips regarding this will be helpful
> >
> > thanks
> >
> >
> This is easy to do by using email to SMS gateways.  A list of them is on
> wikipedia (http://en.wikipedia.org/wiki/List_of_SMS_gateways).  For the
> Asterisk side, you have an extension that sends the email.  I personally
> use an AGI script for this part, but you could use a System() call as well.
>
>
> --johann
>
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Re: [asterisk-users] OT - Gigaset C470IP - How to access SMS settings

[asterisk asterisk]

Olivier,

You should find out the SMS tab in the handset but not in the web service.
Did you IP pone work?

CK

On Tue, Sep 14, 2010 at 2:27 PM, Olivier  wrote:

> Hi,
>
> With my Gigaset C470IP (with latest 02223 firmware), I can't find a way to
> access SMS settings from web configuration app or using a handset.
>
> Has someone been more successful without using auto-configuration mode ?
>
> (For instance, manual says an SMS entry is showing on handset screen but as
> I plugged my base station into a private LAN, I skipped the whole
> auto-configuration process ).
>
> Regards
>
>
>
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Re: [asterisk-users] OT - Gigaset C470IP - How to access SMS settings

[asterisk asterisk]

Yes, only on the handset. My line does not support SMS so sending out is
failed.

On Wed, Sep 15, 2010 at 9:28 PM, Randy R  wrote:

> On Wed, Sep 15, 2010 at 1:43 PM, Olivier  wrote:
>
>>
>> On the S675IP SMS is here:
>>>
>>> Messaging -> SMS -> Settings
>>>
>>
>> No SMS entry is showing  on Settings/Messaging page, here.
>> How did you set your S675IP ?
>> Did you use any autoconfiguration or country menu ?
>>
>>
>> We don't use SMS on fixed. There is nothing on the web menu, only the
> handset menus.
>
> /r
>
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[asterisk-users] Dual WAN with load balancing

[asterisk asterisk]

I encounter problem in using Dual WAN with load balancing on asterisk
1.6.2.11.

My problem is registration of one VOIP provider. I can dial out but not
probably answer. It drops. One of the error message is
SIP/2.0 404 not found.

I am not sure about the problem but note that it may be related to incorrect
IP being used. Sometimes, WAN 1 and sometimes WAN 2

Could someone help to point to fix?

Thanks.

CK
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Re: [asterisk-users] Dual WAN with load balancing

[asterisk asterisk]

Apart from that, any other tricks that I can manipulate within asterisk.
??sip.conf parameter or other??

On Thu, Sep 16, 2010 at 12:07 AM, Luki  wrote:

> > I am not sure about the problem but note that it may be related to
> incorrect
> > IP being used. Sometimes, WAN 1 and sometimes WAN 2
>
> Most likely. Get a provider that uses IP authentication rather than
> registrations, and enable access from both of your WAN IPs. All set.
>
> Luki
>
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[asterisk-users] Some give 603 Declined

[asterisk asterisk]

Hi,

I have some problem with my provider. While the sip registration is
successful, i intermittently encounter problem in dialing out. I receive 603
Declined error in my Sjphone client. The asterisk log shows line is
busy/congestion.

Appreciate if help or direction can be provided.

Thanks.

CK
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Re: [asterisk-users] Some give 603 Declined

[asterisk asterisk]

Here is the sip log

ns*CLI> sip set debug peer hkbn2b
SIP Debugging Enabled for IP: 203.80.89.139:5060
[Oct 15 06:35:19] NOTICE[2462]: chan_sip.c:18334 handle_response_register:
Outbound Registration: Expiry for sip.voipuser.org is 120 sec (Scheduling
reregistration in 105 s)
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
-- Executing [8935944...@dlpn_dp1:1] Dial("SIP/6100-0006",
"SIP/35944...@hkbn2b,,r") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Audio is at 113.253.226.153 port 10650
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 203.80.89.139:5060:
INVITE sip:35944...@s2hkbntel.net:5060 SIP/2.0
Via: SIP/2.0/UDP 113.253.226.153:5060;branch=z9hG4bK1880eaca;rport
Max-Forwards: 70
From: "ck...@mobile"

>;tag=as12eb85f9
To: 
Contact: >
Call-ID: 3f603bea2560e9b836ea250932486...@s2hkbntel.net
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.12
Date: Thu, 14 Oct 2010 22:35:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 241

v=0
o=root 316173620 316173620 IN IP4 113.253.226.153
s=Asterisk PBX 1.6.2.12
c=IN IP4 113.253.226.153
t=0 0
m=audio 10650 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
-- Called 35944...@hkbn2b

<--- SIP read from UDP:203.80.89.139:5060 --->
SIP/2.0 100 Trying
t: 
f: "ck...@mobile" 
>;tag=as12eb85f9
i: 3f603bea2560e9b836ea250932486...@s2hkbntel.net
CSeq: 102 INVITE
v: SIP/2.0/UDP 113.253.226.153:5060
;received=113.253.226.174;rport;branch=z9hG4bK1880eaca
Server: MCS5x00_3.0
k: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec
l: 0


<->
--- (9 headers 0 lines) ---

<--- SIP read from UDP:203.80.89.139:5060 --->
SIP/2.0 487 Request Terminated
t: ;tag=781480306
f: "ck...@mobile" 
>;tag=as12eb85f9
i: 3f603bea2560e9b836ea250932486...@s2hkbntel.net
CSeq: 102 INVITE
v: SIP/2.0/UDP 113.253.226.153:5060
;received=113.253.226.174;rport;branch=z9hG4bK1880eaca
k: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec
l: 0


<->
--- (8 headers 0 lines) ---
Transmitting (NAT) to 203.80.89.139:5060:
ACK sip:35944...@s2hkbntel.net:5060 SIP/2.0
Via: SIP/2.0/UDP 113.253.226.153:5060;branch=z9hG4bK1880eaca;rport
Max-Forwards: 70
From: "ck...@mobile"

>;tag=as12eb85f9
To: ;tag=781480306
Contact: >
Call-ID: 3f603bea2560e9b836ea250932486...@s2hkbntel.net
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.12
Content-Length: 0


---
Scheduling destruction of SIP dialog '
3f603bea2560e9b836ea250932486...@s2hkbntel.net' in 6400 ms (Method: INVITE)
  == Everyone is busy/congested at this time (1:0/0/1)
-- Executing [8935944...@dlpn_dp1:2] Hangup("SIP/6100-0006", "") in
new stack
  == Spawn extension (DLPN_DP1, 8935944101, 2) exited non-zero on
'SIP/6100-0006'
[Oct 15 06:35:23] NOTICE[2462]: chan_sip.c:11601 sip_reregister:--
Re-registration for  8887109...@sip.pennytel.com
Reliably Transmitting (NAT) to 203.80.89.139:5060:
OPTIONS sip:s2hkbntel.net SIP/2.0
Via: SIP/2.0/UDP 113.253.226.153:5060;branch=z9hG4bK703ea06a;rport
Max-Forwards: 70
From: "asterisk"

>;tag=as1d0ccbd8
To: 
Contact: >
Call-ID: 67f6129e02db3377276c62f209913...@sip.etransmed.net
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.12
Date: Thu, 14 Oct 2010 22:35:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:203.80.89.139:5060 --->
SIP/2.0 100 Trying
t: 
f: "asterisk" 
>;tag=as1d0ccbd8
i: 67f6129e02db3377276c62f209913...@sip.etransmed.net
CSeq: 102 OPTIONS
v: SIP/2.0/UDP 113.253.226.153:5060
;received=113.253.226.174;rport;branch=z9hG4bK703ea06a
Server: MCS5x00_3.0
k: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec
l: 0


<->
--- (9 headers 0 lines) ---

<--- SIP read from UDP:203.80.89.139:5060 --->
SIP/2.0 404 Not Found
t: ;tag=820879923
f: "asterisk" 
>;tag=as1d0ccbd8
i: 67f6129e02db3377276c62f209913...@sip.etransmed.net
CSeq: 102 OPTIONS
v: SIP/2.0/UDP 113.253.226.153:5060
;received=113.253.226.174;rport;branch=z9hG4bK703ea06a
k: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec
l: 0


<->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '
67f6129e02db3377276c62f209913...@sip.etransmed.net' Method: OPTIONS



On Thu, Oct 14, 2010 at 7:55 AM, Paul Belanger  wrote:

> On Wed, Oct 13, 2010 at 6:48 PM, asterisk asterisk 
> wrote:
> > Appreciate if help or direction can be provided.
> >
> 21.6.2 603 Decline
>
>   The callee's machine was successfully contacted but the user
>   explicitly does not wish to or cannot participate.  The resp

Re: [asterisk-users] particular sip registry and outbound proxy

[asterisk asterisk]

Put the outboundproxy=192.0.2.1 under individual sip context not under the
[general], it should work.

CK

On Mon, Oct 25, 2010 at 11:43 PM, sipbeast  wrote:

> Hi,
>
>   My asterisk's version is 1.6.0.26.   I've couple sip providers and  I've
> for new SIP provider I need define outbound proxy. Everything is ok in peer
> section (outboundproxy=192.0.2.1). But what about SIP REGISTER messages? I
> need send SIP register messages also via outbound proxy. How to write SIP
> OUTBOUND call register statement and send this to proxy?
> If I define in general section this:
> outboundproxy=192.0.2.1
>
>   Works OK , but now Asterisk sends all SIP messages  via this 192.0.2.1
> proxy! So after changes other carrier stopped to work.
> Is it possible to write SIP registration statement to one provider and send
> these messages via outbound proxy? I mean to have multiple registration line
> and have different outboundproxy for each line.
> Thanks
>
> //Dante
>
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[asterisk-users] Gtalk and asterisk 1.6

[asterisk asterisk]

I have been using rpm version of asterisk 1.6. However, I notice the support
for gtalk is absent from rpm. I tried to compile source code and then moved
to the /usr/lib/asterisk/modules. But the modules cannot be loaded.

Anyone has successful experience.

Mine is using 1.6.2.12.

I also tried in asterisk 1.8. It works well but only the GUI is not working.

CK
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Re: [asterisk-users] asterisk-gui-2.1.0-rc1

[asterisk asterisk]

Try to use firefox instead of IE. Besides, you may check if there is any
problem in the extensions.conf. My recent experiment of installing gui into
asterisk 11.x is that there is problem in some of the macro script within
extensions.conf.

I delete the sample macro scripts in extensions.conf and use the attached
for my asterisk.


On Sat, May 25, 2013 at 2:31 AM, aristidis tsitras wrote:

>  Hi, how did you installed it?
> if it is svn, thry to install it again.
> if it is through source then delete it and try through svn
>
>
> Hi
>
>
>  I have installed asterisk-gui-2.1.0-rc1 . After I logged in to the GUI ,
> it was continuously refreshing the web browser and trying to load the
> configurations.
>
>  Can I know where is gone wrong ?
>
>  Thanks in advance
> Luke
>
>
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>
>
>
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extensions_macro.conf
Description: Binary data
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