[asterisk-users] GSM Gateway promotion from £69GBP

[Sam Tam]

Hello All

 

This month we would like to offer our GSM Gateway range for less to clear up
some spaces.

 

CT-GSM-1000   Basic GSM Gateway (RJ11) Single Sim £69

CT-G01GSM Gateway with SMS Feature (RJ11) Single Sim £99

CT-G04GSM Gateway (RJ11) Quad Sims£400

CT-G08GSM Gateway (RJ11) 8 Sims   £800

CT-SC375  VoIP GSM Gateway (RJ45) Single Sim  £199

CT-SC375  VoIP GSM Gateway (RJ45) Single Sim SMS  £209

 

 

Delivery fee is £20 per item to anywhere in the world via normal post office
mail un-insurance. If you prefer insurance and express service, I can send
it via DHL for £40.

 

More info can be found on cyber-telcom.net

 

Sam 

 

 

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RE: [asterisk-users] Sending SMS from Asterisk

[Sam Tam]

Drop me an email
I know some GSM Gateway that has a direct serial port for SMS>
Sam

-Original Message-
From: Jon Pounder [mailto:[EMAIL PROTECTED] 
Sent: Wednesday, February 14, 2007 10:50 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Sending SMS from Asterisk

Quoting Patrick <[EMAIL PROTECTED]>:

> On Tue, 2007-02-13 at 18:31 -0700, Stephen Bosch wrote:
>> Singer Wang wrote:
>> > by your .ca address I assume your in Canada..
>> >
>> > both Telus and Rogers have a email-to-SMS gateway...
>>
>> Well, those are notoriously unreliable. I've had messages take hours to
>> arrive when sent by the email-to-SMS gateway. I was kinda hoping for
>> something more direct. Rogers prioritizes internal SMS messages over
>> e-mailed ones.

we do this with the vmobile.ca gateway (which is just using the actual bell
cellular network), and only a handful of times in several years hasn't it
been
instant. I get the sms before my desktop mail reader has even picked up the
same messages in most cases.




>>
>> What I'd like is some kind of SMSC -- or something that accomplishes the
>> same thing.
>
> Maybe http://www.kannel.org/ provides some useful info.
>
> Regards,
> Patrick
>
>
>
>
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Jon Pounder

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RE: [asterisk-users] PRI Call Start

[Michael Collins]

"At times I think the wiki has grown out of control."

 

I hear you.  I'd pay money to anyone willing to create and maintain a
master index!

 

-MC

 

  _  

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: Tuesday, February 13, 2007 7:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PRI Call Start

 

Thanks good info on that page.At times I think the wiki has grown
out of control.  There is almost too much info there... that even with a
search engine you can miss some.  Oh well.. Thanks for the pointer.

On 2/13/07, Michael Collins <[EMAIL PROTECTED]> wrote:

Yeah, it's hard to know what it would be filed under.  However, if you
use zap trunks then you'll want to know about this page:

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

 

BTW, see "Dialing a Group" for specifics on 'g' vs. 'G' as well as other
cool stuff.

 

-MC

 

  _  

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: Tuesday, February 13, 2007 1:45 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [asterisk-users] PRI Call Start

 

Oh interesting.   I don't recall seeing that documented anywhere.
Thanks!

On 2/13/07, John Novack < 
[EMAIL PROTECTED]> wrote:

g hunts low to high
G hunts high to low 

John Novack


Matt wrote:
> Hi,
> If I have a PRI with 23 channels on it.Can I setup Asterisk to
> start outbound calls at 23 and hunt back to 1?  I know I can
> individually do it with gX/23/5551212 (or something along those 
> lines).  But is there a way to make it hunt FROM 23 down to 1.  By
> default it starts at 1 and hunts up to 23.
>

>
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Re: [asterisk-users] E911 SIP or IAX providers?

[Dan Burwinkel]

Hi Kyle,

Vitelity.net does it for me... There are a few others too. I tried a 
half dozen, but none seem to have the elusive Customer Service, E911, 
and good Voice quality. I use multiple providers. Les.net is great for 
everything but E911. Origination-- Les.net . Termination-- Les.net, 
voipjet, and vitelity.net . E911-- Vitelity.net


I've been using VoIP exclusively at my home for about a year. I started 
with an ATA and moved to TrixBox and Polycom IP501s. I tried Linksys 
SPA942, Snom, GXP-2000, and Aastra. The only ones I really could get a 
totally natural sound out of was Aastra and Polycom. I'm finally happy 
with the sound. I really had a hard time finding a provider that 
supported smaller fish like me.


Dan

Kyle Sexton wrote:

Does anyone have any experience with any SIP or IAX providers that
support E911?  I'd love to convert entirely to Asterisk at my house,
but the lack of emergency dialing has been a major hold-up for me.
Thanks in advance for any suggestions!


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Re: [asterisk-users] Customisable In-band ringing?

[Jon Pounder]

Quoting Ray Jackson <[EMAIL PROTECTED]>:


hey, why not just play a wav or mp3 of the other person talking instead of
actually placing a call at all ? Wouldn't that be even cooler ? :)

reminds me of the old "dial 811 to hear a duck quack" - someone with too much
time on their hands.






Nic Bellamy wrote:

Ray Jackson wrote:
Using SIP with progressinband=yes I get Asterisk to generate the 
ringing sound for callers.  However, I was wondering if it is 
possible to configure what is 'played back' to the calling party?  
i.e. instead of just 'ring ring' could I potentially play back a 
song from an MP3, WAV or GSM file?  I'm thinking it would be quite 
cool to offer a customised 'ring' sound while the caller is waiting 
for you to pick up?  How can I do this with Asterisk or some 
external module perhaps?  Any advice welcome!

Hi Ray - LTNS - got the VoIP bug too now then? :-)


Been an addict for too long now... might have to seek rehab soon :-)

The 'm' option has been mentioned, however that requires a separate 
music-on-hold class for each different sound, which would quickly 
become unmanageable.

>
Another chap was working on a similar problem - have a look at the 
asterisk-dev thread titled "app_dial.c modification" started by 
Darren Nay on the 24th Jan 2007. I don't know if he's got it working 
yet, but it might be worth getting in touch with him.


Thanks for the pointer.  I am using RealTime/MySQL queries for 
everything to avoid having to ever reload.  As you pointed out the 
'm' flag relies on a class which must have already been created in 
musiconhold.conf first.  I was hoping there might be some way of 
dynamically telling Asterisk to play some inband Media without having 
to rely on static config.  I'll keep an eye on that thread and see if 
anything comes of it.


Cheers,
Ray


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Re: [asterisk-users] Customisable In-band ringing?

[Ray Jackson]

Nic Bellamy wrote:

Ray Jackson wrote:
Using SIP with progressinband=yes I get Asterisk to generate the 
ringing sound for callers.  However, I was wondering if it is possible 
to configure what is 'played back' to the calling party?  i.e. instead 
of just 'ring ring' could I potentially play back a song from an MP3, 
WAV or GSM file?  I'm thinking it would be quite cool to offer a 
customised 'ring' sound while the caller is waiting for you to pick 
up?  How can I do this with Asterisk or some external module perhaps?  
Any advice welcome!

Hi Ray - LTNS - got the VoIP bug too now then? :-)


Been an addict for too long now... might have to seek rehab soon :-)

The 'm' option has been mentioned, however that requires a separate 
music-on-hold class for each different sound, which would quickly become 
unmanageable.

>
Another chap was working on a similar problem - have a look at the 
asterisk-dev thread titled "app_dial.c modification" started by Darren 
Nay on the 24th Jan 2007. I don't know if he's got it working yet, but 
it might be worth getting in touch with him.


Thanks for the pointer.  I am using RealTime/MySQL queries for 
everything to avoid having to ever reload.  As you pointed out the 'm' 
flag relies on a class which must have already been created in 
musiconhold.conf first.  I was hoping there might be some way of 
dynamically telling Asterisk to play some inband Media without having to 
rely on static config.  I'll keep an eye on that thread and see if 
anything comes of it.


Cheers,
Ray


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[asterisk-users] E911 SIP or IAX providers?

[Kyle Sexton]

Does anyone have any experience with any SIP or IAX providers that
support E911?  I'd love to convert entirely to Asterisk at my house,
but the lack of emergency dialing has been a major hold-up for me.
Thanks in advance for any suggestions!

--
Kyle Sexton
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Re: [asterisk-users] PRI Call Start

[Matt]

Thanks good info on that page.At times I think the wiki has grown out of
control.  There is almost too much info there... that even with a search
engine you can miss some.  Oh well.. Thanks for the pointer.

On 2/13/07, Michael Collins <[EMAIL PROTECTED]> wrote:


 Yeah, it's hard to know what it would be filed under.  However, if you
use zap trunks then you'll want to know about this page:

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



BTW, see "Dialing a Group" for specifics on 'g' vs. 'G' as well as other
cool stuff.



-MC


  --

*From:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *On Behalf Of *Matt
*Sent:* Tuesday, February 13, 2007 1:45 PM
*To:* [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
*Subject:* Re: [asterisk-users] PRI Call Start



Oh interesting.   I don't recall seeing that documented anywhere.  Thanks!

On 2/13/07, *John Novack* < [EMAIL PROTECTED]> wrote:

g hunts low to high
G hunts high to low

John Novack


Matt wrote:
> Hi,
> If I have a PRI with 23 channels on it.Can I setup Asterisk to
> start outbound calls at 23 and hunt back to 1?  I know I can
> individually do it with gX/23/5551212 (or something along those
> lines).  But is there a way to make it hunt FROM 23 down to 1.  By
> default it starts at 1 and hunts up to 23.
> 
>
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Re: [asterisk-users] Sending SMS from Asterisk

[Jon Pounder]

Quoting Patrick <[EMAIL PROTECTED]>:


On Tue, 2007-02-13 at 18:31 -0700, Stephen Bosch wrote:

Singer Wang wrote:
> by your .ca address I assume your in Canada..
>
> both Telus and Rogers have a email-to-SMS gateway...

Well, those are notoriously unreliable. I've had messages take hours to
arrive when sent by the email-to-SMS gateway. I was kinda hoping for
something more direct. Rogers prioritizes internal SMS messages over
e-mailed ones.


we do this with the vmobile.ca gateway (which is just using the actual bell
cellular network), and only a handful of times in several years hasn't it been
instant. I get the sms before my desktop mail reader has even picked up the
same messages in most cases.






What I'd like is some kind of SMSC -- or something that accomplishes the
same thing.


Maybe http://www.kannel.org/ provides some useful info.

Regards,
Patrick




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Tools to Power Your e-Business Solutions
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www.opayc.com


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Re: [asterisk-users] Sending SMS from Asterisk

[Patrick]

On Tue, 2007-02-13 at 18:31 -0700, Stephen Bosch wrote:
> Singer Wang wrote:
> > by your .ca address I assume your in Canada..
> > 
> > both Telus and Rogers have a email-to-SMS gateway...
> 
> Well, those are notoriously unreliable. I've had messages take hours to
> arrive when sent by the email-to-SMS gateway. I was kinda hoping for
> something more direct. Rogers prioritizes internal SMS messages over
> e-mailed ones.
> 
> What I'd like is some kind of SMSC -- or something that accomplishes the
> same thing.

Maybe http://www.kannel.org/ provides some useful info.

Regards,
Patrick




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Re: [asterisk-users] How can I use "Asterisk Manager API" to hold and retrive an active call?

[Stefan Reuter]

James Zhang wrote:
> These are common functions. Why "Asterisk Manager"
> doesn't  provide commands to hold and retrive an active channel?
> If it must be implemented by AGI, could anyone give a direction or steps?

Sure the Manager API provides all thing to do that.
Maybe you are just using the wrong library on top of the Manager API ;)

Asterisk-Java as an example lets you retrieve active channels, iterate
over them, hangup, redirect, ... whatever.

Example to hangup all active channels:

for (AsteriskChannel channel : server.getChannels())
{
channel.hangup();
}

http://asterisk-java.org

I am sure other libraries provide similar abstraction.

=Stefan

-- 
reuter network consulting
Neusser Str. 110
50760 Koeln
Germany
Telefon: +49 221 1305699-0
Telefax: +49 221 1305699-90
E-Mail:  [EMAIL PROTECTED]
Jabber:  [EMAIL PROTECTED]



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Re: [asterisk-users] The High Performance Echo Canceller (HPEC)

[Nic Bellamy]

shadowym wrote:

Interesting,

Is this just a more advanced software echo canceller or software with
hardware hooks or software with hardware assisted processing?
  
A more advanced software canceller (there's no magical thing that makes 
"hardware" echo cancellers better, it's still software, but it's running 
on a DSP so it has more grunt available to it).


It's licensed from Adaptive Digital Technologies - G.168 compliant, and 
supports up to 1024 taps (128ms) of tail coverage. Comes as a binary 
blob, but such is life.

How would it compare to a true hardware echo canceller like the one Sangoma
uses.  Besides the extra CPU cycles required.
  
Quite comparable - not sure if Octasic (as used by Sangoma and the 
latest Digium cards) or ADT would win in a shootout, but they're both in 
the same quality class.


The main issue is going to be CPU usage - getting this going at 1024 
taps on a full T1/E1 span would likely require two fast CPUs with the 
interrupts distributed evenly between them... and even then, *shrug*


Cheers,
   Nic.

-Original Message-
From: Nic Bellamy [mailto:[EMAIL PROTECTED] 
Sent: Tuesday, February 13, 2007 12:41 PM

To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] The High Performance Echo Canceller (HPEC)

Larry Shields wrote:
  
I recently read about the following new technologies from Digium.  Has 
anyone tried the new HPEC or knows when it will be available?


It's out now, and I've tried it - the difference between HPEC and MG2 from
trunk is stunning - in situations with bad echo where MG2 can take ten or
more seconds to converge to a reasonable degree, HPEC does it in perhaps
300ms - converging on my intake of breath before I say "hello", and
absolutely no echo after that unless I purposefully go out of my way to
screw it up (whistling/blowing into the handpiece for instance - even then,
the malfunction is minimal).

You can now buy it from the Digium website (US$10 per channel), or if you
have an in-warranty Digium card, email through the serial numbers to Digium
support and they'll give you a key (this is what I did).

You'll need Zaptel 1.2.13 to make it go.

It does take quite a bit of CPU though - perhaps 70% more compared to
MG2-trunk for the same number of taps from my rough measurements.

Cheers,
Nic.

--
Nic Bellamy,
Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/



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--
Nic Bellamy,
Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/

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Re: [asterisk-users] Sending SMS from Asterisk

[Stephen Bosch]

Singer Wang wrote:
> by your .ca address I assume your in Canada..
> 
> both Telus and Rogers have a email-to-SMS gateway...

Well, those are notoriously unreliable. I've had messages take hours to
arrive when sent by the email-to-SMS gateway. I was kinda hoping for
something more direct. Rogers prioritizes internal SMS messages over
e-mailed ones.

What I'd like is some kind of SMSC -- or something that accomplishes the
same thing.

Any ideas?

-Stephen-

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RE: [asterisk-users] PRI Call Start

[Michael Collins]

Yeah, it's hard to know what it would be filed under.  However, if you
use zap trunks then you'll want to know about this page:

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

 

BTW, see "Dialing a Group" for specifics on 'g' vs. 'G' as well as other
cool stuff.

 

-MC

 

  _  

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: Tuesday, February 13, 2007 1:45 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [asterisk-users] PRI Call Start

 

Oh interesting.   I don't recall seeing that documented anywhere.
Thanks!

On 2/13/07, John Novack < 
[EMAIL PROTECTED]> wrote:

g hunts low to high
G hunts high to low 

John Novack


Matt wrote:
> Hi,
> If I have a PRI with 23 channels on it.Can I setup Asterisk to
> start outbound calls at 23 and hunt back to 1?  I know I can
> individually do it with gX/23/5551212 (or something along those 
> lines).  But is there a way to make it hunt FROM 23 down to 1.  By
> default it starts at 1 and hunts up to 23.
>

>
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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] Sending SMS from Asterisk

[Singer Wang]

by your .ca address I assume your in Canada..

both Telus and Rogers have a email-to-SMS gateway...


Stephen Bosch wrote:

Hi:

Say I want to build an IVR application which sends an SMS message to a
mobile telephone when the caller responds to a prompt in certain way.

I think I can manage the part about generating the message and building
something to actually send it. The part I'm foggy about is: how would I
actually get the SMS message to the carrier? Discussions with the
carrier have led absolutely nowhere (they are not interested in helping
an individual customer and technical staff Tiers I and II have no idea
what I am talking about).

Are there SMS aggregators that I could use for sending messages to this
particular phone over the Internet?

Thanks for any input.

-Stephen-
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[asterisk-users] Sending SMS from Asterisk

[Stephen Bosch]

Hi:

Say I want to build an IVR application which sends an SMS message to a
mobile telephone when the caller responds to a prompt in certain way.

I think I can manage the part about generating the message and building
something to actually send it. The part I'm foggy about is: how would I
actually get the SMS message to the carrier? Discussions with the
carrier have led absolutely nowhere (they are not interested in helping
an individual customer and technical staff Tiers I and II have no idea
what I am talking about).

Are there SMS aggregators that I could use for sending messages to this
particular phone over the Internet?

Thanks for any input.

-Stephen-
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Re: [asterisk-users] Re: Auto Answer (Paging)

[C F]

http://lists.digium.com/pipermail/asterisk-users/2006-August/163798.html

On 2/12/07, Steve Davies <[EMAIL PROTECTED]> wrote:

On 2/12/07, Rob Schall <[EMAIL PROTECTED]> wrote:
> Steve,
>
> I posed a similar question to Shane, but maybe you'll know as well..
>
> I was able to get app_page to work. So when I call... **8050, it auto
> answers and the callee is muted. However, what if that person wants to
> "answer" the page and pickup to talk. They are already muted. Can you
> unmute if you are the callee?
>

:) Interesting question - I believe that this would require a
modification to app_meetme to allow a called-user to request to talk
if they are started muted. I certainly don't see such a feature
documented at the moment.

You may have some luck if you create a custom feature in features.conf
that executes the MeetMe UnMute command if a certain key sequence if
pressed. Not sure how that would work though, just dumping random
thoughts really.

http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe whould be a
good starting place for further reading and ideas.

Steve
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Re: [asterisk-users] Paging Followup

[C F]

Look in the list archives, I have posted a solution a while back. It
involves changing the source and recompling.

On 2/13/07, Rob Schall <[EMAIL PROTECTED]> wrote:

Hello All,

Hoping all of you might have an additional option for me to try at this
point. :)

My Goal:
To have a paging option that does the following When I press **_
it will send a ring-answer page to that person. The person on the other
end should be muted, so if they are in a conference, you can't hear what
is going on in the meeting. If that person hears me and decides they
want to talk to me, they can press *1 (or something similar).

My Problems:
Paging solves the first part of my issue (muted meetme) and I have the
Ring-Answer part working great. However, I can't get it to accept *1 or
anything like that to accept the call and talk.

I've Tried:
I've tried using Dial and Page commands to make this work. I noticed the
options that are sent with the page command (it generates a Meetme for
both ends of the call). I think I could just add a "s" to the meetme
options. I believe I should be able to do 2 meetme commands. One for me,
and one for the person I am paging, and we will connect to the same
conference.

But short of creating a call file, is this possible?

Any thoughts?
Rob

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RE: [asterisk-users] The High Performance Echo Canceller (HPEC)

[shadowym]

Interesting,

Is this just a more advanced software echo canceller or software with
hardware hooks or software with hardware assisted processing?

How would it compare to a true hardware echo canceller like the one Sangoma
uses.  Besides the extra CPU cycles required.  

-Original Message-
From: Nic Bellamy [mailto:[EMAIL PROTECTED] 
Sent: Tuesday, February 13, 2007 12:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] The High Performance Echo Canceller (HPEC)

Larry Shields wrote:
> I recently read about the following new technologies from Digium.  Has 
> anyone tried the new HPEC or knows when it will be available?
It's out now, and I've tried it - the difference between HPEC and MG2 from
trunk is stunning - in situations with bad echo where MG2 can take ten or
more seconds to converge to a reasonable degree, HPEC does it in perhaps
300ms - converging on my intake of breath before I say "hello", and
absolutely no echo after that unless I purposefully go out of my way to
screw it up (whistling/blowing into the handpiece for instance - even then,
the malfunction is minimal).

You can now buy it from the Digium website (US$10 per channel), or if you
have an in-warranty Digium card, email through the serial numbers to Digium
support and they'll give you a key (this is what I did).

You'll need Zaptel 1.2.13 to make it go.

It does take quite a bit of CPU though - perhaps 70% more compared to
MG2-trunk for the same number of taps from my rough measurements.

Cheers,
Nic.

--
Nic Bellamy,
Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/



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Re: [asterisk-users] Customisable In-band ringing?

[Nic Bellamy]

Ray Jackson wrote:
Using SIP with progressinband=yes I get Asterisk to generate the 
ringing sound for callers.  However, I was wondering if it is possible 
to configure what is 'played back' to the calling party?  i.e. instead 
of just 'ring ring' could I potentially play back a song from an MP3, 
WAV or GSM file?  I'm thinking it would be quite cool to offer a 
customised 'ring' sound while the caller is waiting for you to pick 
up?  How can I do this with Asterisk or some external module perhaps?  
Any advice welcome!

Hi Ray - LTNS - got the VoIP bug too now then? :-)

The 'm' option has been mentioned, however that requires a separate 
music-on-hold class for each different sound, which would quickly become 
unmanageable.


Another chap was working on a similar problem - have a look at the 
asterisk-dev thread titled "app_dial.c modification" started by Darren 
Nay on the 24th Jan 2007. I don't know if he's got it working yet, but 
it might be worth getting in touch with him.


Cheers,
   Nic.

--
Nic Bellamy,
Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/

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RE: [asterisk-users] Mini-ITX board + FXO PCI card?

[Michelle Dupuis]

We use a lot of mini-itx pc's, including the pCI slot.  I don't think any of
the systems have shared an irq with the PCI slot

MD 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Vincent
Delporte
Sent: Tuesday, February 13, 2007 5:29 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Mini-ITX board + FXO PCI card?

At 10:09 11/02/2007 -0500, Gordon Henderson wrote:
>Check the processor spec. carefully. [...] Also make sure you compile 
>asterisk for an i586

OK, I'll make sure it has enough cache and I'll recompile the code myself. 
I'm thinking of getting an ML 8000
http://via.com.tw/en/products/mainboards/motherboards.jsp?motherboard_id=301
.

At 10:09 11/02/2007 -0500, Manny A. Wise wrote:
>I did, and I was NOT happy with the results... Mini-itx have a serious 
>problems with IRQ sharing... I am happily using a embeded system now, 
>but the FXO and FXS have to be external.

Those boards only come with one PCI slot. Do you mean it could share an IRQ
with some embedded component like the video card?

BTW, in this age of big USB drive, I don't really nee a DVD/CDRW combo. 
Does someone know if the Via motherboards (at least the ML series) supports
booting off a USB drive, so I can use this to start Linux and fetch install
files from an FTP server?

Thank you.

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Re: [asterisk-users] SMS via VoIP and web

[Steve Kennedy]

On Wed, Feb 14, 2007 at 07:17:32AM +0800, Ronald Wiplinger wrote:

> Where can I get a starting point for setting up sms via VoIP and via web.
> I want to send SMS from VoIP or web  to VoIP phones and GSM phones.
> 1. how to set-up?
> 2. which smsc should I use? (what is the price?)
> 3. which phones can be used?

Some telcos support sending SMS down phone lines, it's reasonably common
in Europe and there's an ETSI spec for it.

However it's probably easier to use something like Kannel which has an
http interface and then either connect that to an SMSC or locally
through a GSM terminal (phone).

SMSC connections and pricing will vary depending on what country you're
in. As a small customer (in the UK at least) it's unlikely you'd get an
connection to an operator's SMSC and you'd have to go through an
aggregator.


Steve

-- 
NetTek Ltd  UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED]
Euro Tech News Blog http://eurotechnews.blogspot.com
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Re: [asterisk-users] The High Performance Echo Canceller (HPEC)

[Nic Bellamy]

Bill Gibbs wrote:

Will this work with SIP channels?  I get zero echo out the PRI but I do
get it occasionally on a LD provider (SIP) we use.  The stock * install
doesn't appear to be doing anything stopping echo on those channels.
  
Nope, it won't help - echo cancellation needs to be performed as close 
to the source of the echo as possible. When you've got SIP in the mix, 
you've got variable network delays, packet loss, jitter buffer 
interpolation and various other things to think about, and this would 
make an echo cancellers job orders of magnitude harder (and it's already 
a pretty hard problem).


While it would be technically possible to echo-cancel SIP channels, it'd 
be _extremely_ CPU intensive (you'd need massive tail coverage) and 
probably not do a very good job.


If you're getting echo from a VoIP->PSTN provider, they need to do 
something about it themselves - by the time it gets to you, it's too late.


Cheers,
   Nic.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nic
Bellamy
Sent: Tuesday, February 13, 2007 3:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] The High Performance Echo Canceller (HPEC)

Larry Shields wrote:
  

I recently read about the following new technologies from Digium.  Has



  

anyone tried the new HPEC or knows when it will be available?

It's out now, and I've tried it - the difference between HPEC and MG2 
from trunk is stunning - in situations with bad echo where MG2 can take 
ten or more seconds to converge to a reasonable degree, HPEC does it in 
perhaps 300ms - converging on my intake of breath before I say "hello", 
and absolutely no echo after that unless I purposefully go out of my way


to screw it up (whistling/blowing into the handpiece for instance - even

then, the malfunction is minimal).

You can now buy it from the Digium website (US$10 per channel), or if 
you have an in-warranty Digium card, email through the serial numbers to


Digium support and they'll give you a key (this is what I did).

You'll need Zaptel 1.2.13 to make it go.

It does take quite a bit of CPU though - perhaps 70% more compared to 
MG2-trunk for the same number of taps from my rough measurements.


Cheers,
Nic.

  



--
Nic Bellamy,
Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/

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Re: [asterisk-users] Zaptel Won't Compile In Slackware 11.0 Kernel 2.4.33

[demuel]

[EMAIL PROTECTED]:/usr/src/linux/include/linux$ pwd
/usr/src/linux/include/linux
[EMAIL PROTECTED]:/usr/src/linux/include/linux$ sudo make updatedb
make: *** No rule to make target `updatedb'.  Stop.
[EMAIL PROTECTED]:/usr/src/linux/include/linux$ ls -la page-flags.h
/bin/ls: page-flags.h: No such file or directory
[EMAIL PROTECTED]:/usr/src/linux/include/linux$

Did i missed something down here? Weird thing is, even a fresh install of 
slackware produced the
same kind of error. Actually, it used to be working about a week before I made 
a source upgrade.
Any thoughts?


Regards,
Demuel

> On Tue, Feb 13, 2007 at 02:23:21PM +, J. Espinal wrote:
>> make a 'updatedb' , and look for 'page-flags.h' , i think that you might
>> be missing that file,
>>
>
> under the include/ directory in the linux kernel source directory.
>
>>
>>
>>
>>
>> J. Espinal,
>>
>>
>>
>> [EMAIL PROTECTED] wrote:
>> >Anybody,
>> >
>> >
>> >I have download asterisk 1.4 via svn. whem I compiled it, I got the
>> >following error:
>> >
>> >
>> >/lib/modules/2.4.33.3/build/include/asm/system.h:190: warning:
>> >dereferencing type-punned pointer
>> >will break strict-aliasing rules
>> >zttranscode.c:37:30: linux/page-flags.h: No such file or directory
>> >make[1]: *** [zttranscode.o] Error 1
>> >make[1]: Leaving directory
>> >`/home/kingkong/code/projects/asterisk/source/zaptel-1.4'
>> >make: *** [all] Error 2
>>
>> make a 'updatedb' , and look for 'page-flags.h' , i think that you might
>> be missing that file,
>>
>
> under the include/ directory in the linux kernel source directory.
>
> Better yet: simply don't build zttranscode, unless you have a card that
> actually supports it...
>
> --
>Tzafrir Cohen
> icq#16849755jabber:[EMAIL PROTECTED]
> +972-50-7952406   mailto:[EMAIL PROTECTED]
> http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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[asterisk-users] SMS via VoIP and web

[Ronald Wiplinger]

Where can I get a starting point for setting up sms via VoIP and via web.

I want to send SMS from VoIP or web  to VoIP phones and GSM phones.

1. how to set-up?
2. which smsc should I use? (what is the price?)
3. which phones can be used?


bye

Ronald
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Re: [asterisk-users] Mini-ITX board + FXO PCI card?

[Vincent Delporte]

At 10:09 11/02/2007 -0500, Gordon Henderson wrote:
Check the processor spec. carefully. [...] Also make sure you compile 
asterisk for an i586


OK, I'll make sure it has enough cache and I'll recompile the code myself. 
I'm thinking of getting an ML 8000 
http://via.com.tw/en/products/mainboards/motherboards.jsp?motherboard_id=301 .


At 10:09 11/02/2007 -0500, Manny A. Wise wrote:
I did, and I was NOT happy with the results... Mini-itx have a serious 
problems with IRQ sharing... I am happily using a embeded system now, but 
the FXO and FXS have to be external.


Those boards only come with one PCI slot. Do you mean it could share an IRQ 
with some embedded component like the video card?


BTW, in this age of big USB drive, I don't really nee a DVD/CDRW combo. 
Does someone know if the Via motherboards (at least the ML series) supports 
booting off a USB drive, so I can use this to start Linux and fetch install 
files from an FTP server?


Thank you.

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Re: [asterisk-users] PRI Call Start

[Matt]

Oh interesting.   I don't recall seeing that documented anywhere.  Thanks!

On 2/13/07, John Novack <[EMAIL PROTECTED]> wrote:


g hunts low to high
G hunts high to low

John Novack


Matt wrote:
> Hi,
> If I have a PRI with 23 channels on it.Can I setup Asterisk to
> start outbound calls at 23 and hunt back to 1?  I know I can
> individually do it with gX/23/5551212 (or something along those
> lines).  But is there a way to make it hunt FROM 23 down to 1.  By
> default it starts at 1 and hunts up to 23.
> 
>
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[asterisk-users] FRITZ!Box Fon ata

[Razza]

Hi all, is it possible to to dumb down a "FRITZ!Box Fon ata" (

http://www.avm.de/en/Produkte/FRITZBox/FRITZ_Box_Fon_ata/index.html##)
and have the two FXS ports AND the ISDN interface register with
Asterisk. In much the same way a sipura SPA3K works?
 
Regards
Ray
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Re: [asterisk-users] PRI Call Start

[John Novack]

g hunts low to high
G hunts high to low

John Novack


Matt wrote:

Hi,
If I have a PRI with 23 channels on it.Can I setup Asterisk to 
start outbound calls at 23 and hunt back to 1?  I know I can 
individually do it with gX/23/5551212 (or something along those 
lines).  But is there a way to make it hunt FROM 23 down to 1.  By 
default it starts at 1 and hunts up to 23.



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[asterisk-users] PRI Call Start

[Matt]

Hi,
If I have a PRI with 23 channels on it.Can I setup Asterisk to start
outbound calls at 23 and hunt back to 1?  I know I can individually do it
with gX/23/5551212 (or something along those lines).  But is there a way to
make it hunt FROM 23 down to 1.  By default it starts at 1 and hunts up to
23.
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RE: [asterisk-users] The High Performance Echo Canceller (HPEC)

[Bill Gibbs]

Will this work with SIP channels?  I get zero echo out the PRI but I do
get it occasionally on a LD provider (SIP) we use.  The stock * install
doesn't appear to be doing anything stopping echo on those channels.

Bill

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nic
Bellamy
Sent: Tuesday, February 13, 2007 3:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] The High Performance Echo Canceller (HPEC)

Larry Shields wrote:
> I recently read about the following new technologies from Digium.  Has

> anyone tried the new HPEC or knows when it will be available?
It's out now, and I've tried it - the difference between HPEC and MG2 
from trunk is stunning - in situations with bad echo where MG2 can take 
ten or more seconds to converge to a reasonable degree, HPEC does it in 
perhaps 300ms - converging on my intake of breath before I say "hello", 
and absolutely no echo after that unless I purposefully go out of my way

to screw it up (whistling/blowing into the handpiece for instance - even

then, the malfunction is minimal).

You can now buy it from the Digium website (US$10 per channel), or if 
you have an in-warranty Digium card, email through the serial numbers to

Digium support and they'll give you a key (this is what I did).

You'll need Zaptel 1.2.13 to make it go.

It does take quite a bit of CPU though - perhaps 70% more compared to 
MG2-trunk for the same number of taps from my rough measurements.

Cheers,
Nic.

-- 
Nic Bellamy,
Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/

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[asterisk-users] AgentCallBackLogin vs AddQueueMember

[gc]

I am developing an ACD front end using Asterisk 1.2.14. I heard that 
AgentCallBackLogin will be deprecated in future version of *.
Is this true? If it is, how can I use AddQueueMember to replace 
AgentCallBackLogin? I mean to login an agent in multiple queues at once. I have 
multiple queues and a lot of agents defined in  queues.conf and agents.conf. 
Each agent may login more than one queue. It seem that AgentCallBackLogin  is 
much easier than AddQueueMember to manage this kind of situation. 

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Re: [asterisk-users] The High Performance Echo Canceller (HPEC)

[Nic Bellamy]

Larry Shields wrote:
I recently read about the following new technologies from Digium.  Has 
anyone tried the new HPEC or knows when it will be available?
It's out now, and I've tried it - the difference between HPEC and MG2 
from trunk is stunning - in situations with bad echo where MG2 can take 
ten or more seconds to converge to a reasonable degree, HPEC does it in 
perhaps 300ms - converging on my intake of breath before I say "hello", 
and absolutely no echo after that unless I purposefully go out of my way 
to screw it up (whistling/blowing into the handpiece for instance - even 
then, the malfunction is minimal).


You can now buy it from the Digium website (US$10 per channel), or if 
you have an in-warranty Digium card, email through the serial numbers to 
Digium support and they'll give you a key (this is what I did).


You'll need Zaptel 1.2.13 to make it go.

It does take quite a bit of CPU though - perhaps 70% more compared to 
MG2-trunk for the same number of taps from my rough measurements.


Cheers,
   Nic.

--
Nic Bellamy,
Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/

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Re: [asterisk-users] Originating calls: Astmanproxy vs Direct Connection vs Call files

[Tim Panton]

On 13 Feb 2007, at 12:56, Julian Lyndon-Smith wrote:

I've got around 45 people who need to place calls from our inhouse  
app. What is the considered "best practice" for placing these calls:


1) All clients connect to astmanproxy, and use AMI API Originate  
command
2) All clients connect directly to the astersik AMI and use the API  
Originate command
3) All clients create a db record, some process reads the record  
and writes out a call file
4) All clients connect to a web / socket service which then writes  
a call file ...


I'd advise a variation on 4)

All clients use a (SOAP/XML/http post?) web service.

Web service is implemented as a Java Servlet. Servlet uses the  
(excellent) asterisk-java package

to send commands to AMI

We've done something like it and are pretty happy.





We are currently using #1, but every few days or so, the "putting"  
of data seems to take longer and longer. If I kill astmanproxy and  
restart it, the system is lighting fast for another few days.


What settings would I need for astmanproxy in order simply to  
access the AMI API (I don't need to receive or process events from  
asterisk)


Julian



Tim Panton

www.mexuar.net
www.westhawk.co.uk/



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[asterisk-users] Compiling Asterisk With ZapTel?

[Charlie Grosvenor]

I have tried to compile asterisk with zaptel:

 

./configure --with-zaptel=/usr/src/zaptel-1.4.0

make

make install

 

however when I run asterisk it says that the zap command is missing.
What am I doing wrong? I have compiled and installed zaptel fine and it
is recognizing my card.

 

Thanks

 

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[asterisk-users] Your favorite switchboard application software ?

[Olivier]

Hi,

What's your favorite switchboard application software, for a 100-200 seats
company attendant ?

The setup is :
PSTN --- Asterisk - SIP Phones
  |
  - Attendant console

Has anyone used Icecom switchboard, FOP, Netwise, Hud ?
What do you think of those ?

Regards
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Re: [asterisk-users] WAIT FOR DIGIT not working

[Camilo Echeverry]

Hi .
I had the same problem but downloaded a test script and wait for digit
worked.
the only visible difference is that I wat nos using strict,

so I am rewriting the AGI with

use strict;

Hope this help.


On 9/14/06, Joel Lansden <[EMAIL PROTECTED]> wrote:


 Hello all,

I have been trying to solve this problem for days, with no luck.

When I run an AGI script from my extensions.conf, it seems no matter what
I do, the "WAIT FOR DIGIT" command will not work.  The system just flies
past it without waiting a single millisecond, and of course my script
crashes because it doesn't have the input it needs.  I have run 3 different
versions of Asterisk in the hopes of clearing this up, and presently am on
1.2.12.1.

My script is simple:


#!/usr/bin/perl

use POSIX;

$| = 1;

sub trim {
my @out = @_;
for (@out)
{
   s/^\s+//;
   s/\s+$//;
}
return wantarray ? @out : $out[0];
}

while() {
chomp;
last unless length($_);
if (/^agi_(\w+)\:\s+(.*)$/) {
$AGI{$1} = $2;
}
}

print "EXEC Ringing\n";
print "EXEC Wait 1\n";
print "EXEC Answer\n";
print "EXEC Festival 'Please enter the extension you want to call'\n";
$target = "";

print "WAIT FOR DIGIT 5000\n";
$target .= ;
print "WAIT FOR DIGIT 5000\n";
$target .= ;
print "WAIT FOR DIGIT 5000\n";
$target .= ;

print STDERR "Result was $target\n";


That's all there is to it, but it won't work.

Can anyone help?
Thanks!!!

~Joel


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--
Camilo Echeverry

Your life would be very empty if you had nothing to regret.
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Re: [asterisk-users] Recomended POE Phones

[George Pajari]

voip crazy wrote:

I am looking for phones witch support POE


Aastra 9133i and Aastra 480i

--
George Pajari, netVOICE communications604 484 VOIP (484 8647 x102)
Open Source VoIP/Telephony Specialists  1 877 NET VOIP (638 8647 x102)
 www.netvoice.ca  www.ip-centrex.ca
 www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca

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[asterisk-users] Asterisk 1.4.0 and callwaiting eventually drops call

[Jerry Geis]

I have asterisk 1.4.0 running.
I have a UIP200 that beeps when a second call is incoming.
I flip over with flash talk to that person then
hit flash to go back. The person is there for a short time
then the call is DROPPED.

This worked fine with this phone and 1.2.X

Is this a bug or anyone else have this issue with 1.4?

Jerry
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[asterisk-users] How can I use "Asterisk Manager API" to hold and retrive an active call?

[James Zhang]

These are common functions. Why "Asterisk Manager" doesn't  provide
commands to hold and retrive an active channel?
If it must be implemented by AGI, could anyone give a direction or
steps?
 
Thanks in advance,
 
James
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[asterisk-users] RE: Nortel 81C MSDL Trunking to Asterisk TE110P, Nortel Resetting PRI Channels

[Marlon_Blair]

I have made the change as stated 
span=1,1,0,esf,b8zs
span=2,2,0,esf,b8zs

>Where is your primary clock? 
For the PRI, coming from the Asterisk PBX

>Do you have an atomic clock supplying timing to your Nortel system?  
No, timing picked up over the PRI from the provider  

>Does your Nortel get it's clock from the phone company?
 Yes

>Assuming one of these two is the case, and that both T1s are connected
to the Nortel system, Asterisk should derive it's >clock from the Nortel
system.  
Timing cannot come from the type of card we are using on the Nortel side

Marlon Blair
DOH, Network System Analyst
(850) 245-4400, Cell (850) 528-4244
Fax (850) 412-1148
Work Hours 7 AM to 3:30 PM
_ 

==
MISSION:  To promote and protect the health and safety of all
people in Florida through the delivery of quality public health services
and promotion of health care standards
*
PLEASE NOTE:  Florida has a very broad public records law.  Most
written communications to or from State officials regarding state
business are public records available to the public and media upon
request.  Your e-mail communications may therefore be subject to public
disclosure.
*
Please call the Customer Service Center for Service Requests at
850-922-7599 or SunCom 292-7599




Marlon_Blair at doh.state.fl.us
  wrote on
Tuesday, February 13, 2007 10:17
AM
> We are currently working to trunk from a Nortel 81C to an Asterisk 
> Server 1.4 running on Red Hat Linux.  We have two PRI trunks which 
> work with the exception of the clock slips, which is causing the 
> Nortel to reset the PRIs once a hour.  Thanks for any suggestions.

Assuming that 'clock slips' are the same thing as 'frame slips'

Where is your primary clock? Do you have an atomic clock supplying
timing to your Nortel system? Does your Nortel get it's clock from the
phone company? Assuming one of these two is the case, and that both T1s
are connected to the Nortel system, Asterisk should derive it's clock
from the Nortel system. 
To do that you would want to set up Zaptel.conf 
span=1,1,0,esf,b8zs
span=2,2,0,esf,b8zs

Asterisk would then use T1 number 1 to set it's internal t1 card clock.
If T1 1 was not available, then it would use T1 2. 

This line from your Nortel system makes me think it is using an external
clock. If this is the case then you must have an atomic clock or some
other 'external' clock source at your location. 

>  CLOK EXT

Don Pobanz



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Re: [asterisk-users] Zaptel Won't Compile In Slackware 11.0 Kernel 2.4.33

[Tzafrir Cohen]

On Tue, Feb 13, 2007 at 02:23:21PM +, J. Espinal wrote:
> make a 'updatedb' , and look for 'page-flags.h' , i think that you might 
> be missing that file,
> 

under the include/ directory in the linux kernel source directory.

> 
> 
> 
> 
> J. Espinal,
> 
> 
> 
> [EMAIL PROTECTED] wrote:
> >Anybody,
> >
> >
> >I have download asterisk 1.4 via svn. whem I compiled it, I got the 
> >following error:
> >
> >
> >/lib/modules/2.4.33.3/build/include/asm/system.h:190: warning: 
> >dereferencing type-punned pointer
> >will break strict-aliasing rules
> >zttranscode.c:37:30: linux/page-flags.h: No such file or directory
> >make[1]: *** [zttranscode.o] Error 1
> >make[1]: Leaving directory 
> >`/home/kingkong/code/projects/asterisk/source/zaptel-1.4'
> >make: *** [all] Error 2
>
> make a 'updatedb' , and look for 'page-flags.h' , i think that you might 
> be missing that file,
> 

under the include/ directory in the linux kernel source directory.

Better yet: simply don't build zttranscode, unless you have a card that
actually supports it...

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] Trixbox vs. Custom install

[Tom Rymes]

On Feb 13, 2007, at 11:53 AM, Tzafrir Cohen wrote:


On Tue, Feb 13, 2007 at 10:23:17AM -0500, Tom Rymes wrote:


[snip]


Not to start a flame-war, but I completely disagree. Troubleshooting
a GUI is much easier, given that you don't have to scout for typos,
transposed numbers, etc throughout the dialplan. With the GUI, you
have to double check the information that you input into the GUI, but
that's it. As for hardware, it should be no more difficult to get
Trixbox to play nicely with hardware than any other Asterisk install.
You may have to patch and/or recompile zaptel, asterisk, etc, but
that's no different than what you would have to do with a non-Trixbox
install.


Hmmm... I installed a trixbox system. 'yum update' failed to work, due
to funny games with yum's configuration. A default centos server
installation did not have the same issue.

This is just one example.


I have never run into this problem before, and the only change that I  
know of was to exclude the kernel from updates (to avoid having to  
recompile zaptel) Of course, if you want to update the kernel, change  
the yum settings and download and recompile zaptel. YMMV, so if it  
doesn't work for you, then act accordingly, I suppose. As a  
counterpoint to your example, I have installed Trixbox easily and  
successfuly many times with Sangoma hardware.



(and you really shouldn't have to in almost all cases)


A GUI does its absraction. By that it hides some information that it
deems irrelevant. In many cases this information is relevant.


My point that you quoted originally referred to the fact that you  
shouldn't normally have to recompile Zaptel, Asterisk, or anything  
else to get hardware working with Trixbox.  As for your comment about  
the GUI, I agree. My earlier e-mail tried to state that neither the  
GUI or the non-GUI method of installing and configuring Asterisk is  
better. The GUI is better for some, whereas the non-GUI is better for  
others. If the limitations imposed by the GUI are too much for your  
application, then the GUI isn't for you. If the relative difficulty  
of administering an Asterisk server without a GUI is too much for  
your application, then use the GUI.


One example: just figuring out if FreePBX actually dial, or not at  
all,

requires either a sufficiently-trained asterisk guy to review the
log/cli just to understand why a call did not go through.


I fail to see how this is different from a non-FreePBX setup? Don't  
you still need a sufficiently-trained Asterisk Guy to view the logs  
and CLI to determine why your custom dialplan didn't dial? Not to  
mention to create that custom dialplan in the first place? How does  
troubleshooting a non-GUI asterisk install require less technical  
know-how than troubleshooting a Free-PBX system?


Anyhow, I reiterate that I don't think that either solution is better  
than the other. Determine your requirements, weigh the pros and cons  
of the various GUIs and of running without a GUI and see which is the  
best fit for your requirements. I only object to those who say that  
"No one should use Trixbox/FreePBX, it's too restrictive" or "Running  
Asterisk with a GUI is always Better." Both statements are erroneous.


Tom
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[asterisk-users] Blocking collect calls in Brazil

[Kanelbullar]

Hello all,
   
  I have been unsucessfully looking for conclusive information regarding 
blocking collect calls in Brazil, using either MFC/R2 or ISDN E1 lines. So, I 
would like to ask the list a couple of questions.
   
  Regarding MFC/R2, there seems to be a patch for the chan_unicall module that 
allows analyzing the call category associated to collect calls and taking some 
action upon it. Embratel and Telemar appear to send "9" in the category field, 
in case of collect calls. But, what about the remaining brazilian operators? 
Does anyone know it any of them provides consistently the same number in the 
category field, whenever a collect call is present?
   
  Regarding ISDN E1s, I have found users saying they are using the dialplan 
"Flash" instruction for Zap. But, does this also work for digital lines? Or 
does it only work for analog lines?
   
  Many thanks in advance,
  Paulo


-

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RE: [asterisk-users] Nortel 81C MSDL Trunking to Asterisk TE110P, Nortel Resetting PRI Channels

[Watkins, Bradley]

You have the PRIs set up to recover clock from the Asterisk box, is that
what you want?  If so, you certainly do *not* want span=1,1,0 or 2,2,0
since that will make Asterisk think the 81C should be clock master.  Are
there any telco-timed PRIs somewhere?  If so, set up the PRIs on the 81C
to be CLOK INT and then use span=1,1,0,esf,b8zs and span=2,2,0,esf,b8zs
on the Asterisk box.
 
I'm going to assume that a big system like an existing 81C already has
the master clock set, but of course that will be a necessity if using
internal clocking.
 
- Brad
 
 
 




From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Tuesday, February 13, 2007 11:17 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Nortel 81C MSDL Trunking to Asterisk
TE110P,Nortel Resetting PRI Channels



We are currently working to trunk from a Nortel 81C to an
Asterisk Server 1.4 running on Red Hat Linux.  We have two PRI trunks
which work with the exception of the clock slips, which is causing the
Nortel to reset the PRIs once a hour.  Thanks for any suggestions.

81C   MSDL  Asterisk
Digium TE110P 
 REQ  prt 
 TYPE adan dch 10Zaptel.conf


loadzone= us 
 ADAN DCH 51
defaultzone=us 
   CTYP MSDL
span=1,1,0,esf,b8zs(also tried 1,0,0,esf,b8zs)  
   GRP  1
bchan=1-23 
   DNUM 7
dchan=24 
   PORT 1
span=2,1,0,esf,b8zs  (also tried 2,0,0,esf,b8zs) 
   DES  Asterisk VOIP bchan=25-47

   USR  PRI
dchan=48 
   DCHL 51 
   OTBF 32
zapata.conf 
   PARM RS422  DTE [channels] 
   DRAT 64KC
language=en 
   CLOK EXT
context=default 
   IFC  ESS5
switchtype=5ess 
   SIDE USR
signalling=pri_net 
   CNEG 1
group=1 
   RLS  ID  1
channel => 1-23 
   RCAP ND2   channel =>
25-47 
   MBGA NO
usecallerid=yes 
   OVLR NO
hidecallerid=no 
   OVLS NO
callwaiting=no 
   T200 3
threewaycalling=yes 
   T203 10
transfer=yes 
   N200 3
canpark=yes 
   N201 260
cancallforward=yes 
   K7
echocancancel=no 

echocancelwhenbridged=no 
 And I have immediate=no 
ADAN DCH 71   callreturn=yes 
Built the same asrxgain=0.0

ADAN DCH 31  txgain=0.0 

musiconhold=default 
  
ROUT 1 We start
up Asterisk in the following 
Type RDB   order: 
CUST 00 modprobe
zaptel 
Rout 30
modprobe wcte11xp 
DES  ASTERISK_VOIP_1   ztcfg 
TKTP TIE
safe_asterisk 
NPID_TBL_NUM   0 
ESN  NO  
CNVT NO  
SAT  NO  
RCLS INT 
VTRK NO  
DTRK YES 
BRIP NO  
DGTP PRI 
ISDN YES 
MODE PRA 
IFC  ESS5 
SBN  NO 
PNI  1 
SRVC NNSF 
NCNA YES 
NCRD YES 
CHTY BCH 
CTYP CDP 
INAC YES 
ISAR NO  
CPUB OFF 
DAPC NO  
BCOT 0 
DSEL VOD 
PTYP PRI 
AUTO NO  
DNIS NO  
DCDR NO  
ICOG IAO 
SRCH LIN 
TRMB YES 
STEP 
ACOD 7930 
TCPP NO  
PII NO  
TARG 
CLEN 1 
BILN NO 
OABS 
INST 
IDC  NO  
DCNO 0 * 
NDNO 0 
DEXT NO  
ANTK 
SIGO STD 
ICIS YES 
TIMR ICF  512 
 OGF  512 
 EOD  13952 
 NRD  10112 
 DDL  70 
 ODT  4096 
 RGV  640 
 GRD  896 
 SFB  3 
 NBS  2048 
 NBL  4096 
 IENB  5 
 TFD  0 
 VSS  0 
 VGD  6 
DRNG NO  
CDR  NO  
VRAT NO  
MUS  NO  
FRL  0 0 
FRL  1 0 
FRL  2 0 
FRL  3 0 
FRL  4 0 
FRL  5 0 
FRL  6 0 
FRL  7 0 
OHQ  NO  
OHQT 00 
CBQ  NO  
AUTH NO  
TDET NO  
TTBL 0 
ATAN NO  
PLEV 2 
ALRM NO  
ART  0 
SGRP 0 
AACR NO 


 DES  VERSA 
 TN   101 01 
 TYPE TIE 
 CDEN SD 
 CUST 0 
 TRK  PRI 
 PDCA 1 
 PCM

Re: [asterisk-users] problem with safe_asterisk

[Giorgio Incantalupo]

Ciao Andrea,
Tzafrir is right...safe_asterisk is not very good. I had a discussed 
with him time ago. It is better for you to develop it by yourself. I 
made a version checking PIDs which is what safe_asterisk is lacking.


Giorgio Incantalupo


Andrea De Vita wrote:

Hi all,

I have installed some Asterisk machine, all with the same problem.
My typical configuration is:
- Asterisk 1.2.14 (or 1.4.0beta3)
- CentOS 4.4 server.


The problem is this:
When I start Asterisk with the default init script 
(/etc/init.d/asterisk start) distributed with the source, and kill (or 
kill -9) Asterisk-pid,
then safe_asterisk doesn't correctly work (it dies and not restart 
Asterisk).
Instead, if I start Asterisk with safe_asterisk command from shell, 
after "kill Asterisk-pid", safe_asterisk restart Asterisk correctly.


I would use the init script because I like to use Linux-HA that 
require this.


Someone can help me?

Thanks

Andrea De Vita

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Re: [asterisk-users] Recomended POE Phones

[Dave Fullerton]

Stephen Bosch wrote:

voip crazy wrote:

Hi all,

I am looking for phones witch support POE, with a good relation between
quality and price to work with asterisk. I just see the Thompson st2030
and the Linksys SPA 922 an SPA 942. Witch of this phones or another ones
gave you the best results in a productivity enviroment?

Thanks in advance.

VoipCrazy.


Aren't the Polycom phones PoE?

-Stephen-


Yes and mostly yes. The Polycom 601, 430 and 650 have built-in 802.3af 
PoE capability. Meaning, you can plug the phone into a PoE switch with 
about any patch cable and it will power on. The 301 and 501 require a 
special cable that has a small (maybe 2"x1"x.5") box on the far end that 
converts either 802.3af or cisco (depending on the cable) PoE into what 
the phone uses. This special cable does have a jack on it so you can 
extend the overall length of the cable if need be. It works, but is a 
bit cumbersome to have to deal with.


-Dave
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Re: [asterisk-users] Zaptel Won't Compile In Slackware 11.0 Kernel 2.4.33

[J. Espinal]

make a 'updatedb' , and look for 'page-flags.h' , i think that you might 
be missing that file,






J. Espinal,



[EMAIL PROTECTED] wrote:

Anybody,


I have download asterisk 1.4 via svn. whem I compiled it, I got the following 
error:


/lib/modules/2.4.33.3/build/include/asm/system.h:190: warning: dereferencing 
type-punned pointer
will break strict-aliasing rules
zttranscode.c:37:30: linux/page-flags.h: No such file or directory
make[1]: *** [zttranscode.o] Error 1
make[1]: Leaving directory 
`/home/kingkong/code/projects/asterisk/source/zaptel-1.4'
make: *** [all] Error 2


Any comments?


Regards,
Demuel



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Re: [asterisk-users] AGI "GET DATA" and "WAIT FOR DIGIT" don't work

[J. Espinal]

I have experienced similar problems with AGI some time ago... sometimes, 
the script just get to the 'WAIT FOR DIGIT' function and the streamed 
audio file before it is a little 'long' and the timeout runs while you 
are still listening the audio... Try testing with a very large timeout 
number, let us say... 10 seconds... , that will get u some clue of the 
problem, If the timeout still gets too fast then the function is not 
working as spected (i have experienced that problem too),


test and give us ur result, :)


J. Espinal,




Camilo Echeverry wrote:

Hi.
I'm trying to get digits form the user via agi
something like this: this only should print result=asciicode

but none of the functions even wait until timeout ..
they just pass .. (after a nanosecond)

the las print is always timeout.

Any clue ..?


my $callerid = $AGI{'callerid'} ;
if($callerid !~ /[0-9]{7,20}/){
   #way numbre one
   print "EXEC PLAYBACK  please_enter_your_number \"\"\n";  my $result 
= ;

   print "WAIT FOR DIGIT 3000\n"; my $result = ;

   # Way number two
   # print "GET DATA   please_enter_your_number \"-1\" \"10\""; my 
$result = ;
  
}

print STDERR "$result";



--
Camilo Echeverry

Your life would be very empty if you had nothing to regret.


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Re: [asterisk-users] Trixbox vs. Custom install

[Tzafrir Cohen]

On Tue, Feb 13, 2007 at 09:53:17AM -0700, Stephen Bosch wrote:
> Tom Rymes wrote:
> > On Feb 12, 2007, at 7:13 PM, Stephen Bosch wrote:
> > 
> >> Lee Jenkins wrote:
> >>> Stefano Corsi wrote:
> > 
> >>> The nice things about GUI's in my opinion is that routine chores such as
> >>> setting up extensions, dialing extensions, hunt groups, etc. are less
> >>> likely to contain scripting bugs or typos.  The downside from what I
> >>> gather with many GUI's is that the friendly abstraction that insulates
> >>> you from the nuts and bolts of scripting and configuration also makes it
> >>> difficult to customize the dialplan in some cases.
> >>
> >> It also makes troubleshooting problems a handful-and-a-half. And woe is
> >> you if you need kernel customizations to make your hardware work.
> > 
> > Not to start a flame-war, but I completely disagree. Troubleshooting a
> > GUI is much easier, given that you don't have to scout for typos,
> > transposed numbers, etc throughout the dialplan. With the GUI, you have
> > to double check the information that you input into the GUI, but that's
> > it. As for hardware, it should be no more difficult to get Trixbox to
> > play nicely with hardware than any other Asterisk install. You may have
> > to patch and/or recompile zaptel, asterisk, etc, but that's no different
> > than what you would have to do with a non-Trixbox install. (and you
> > really shouldn't have to in almost all cases)
> 
> I come from the practice of compiling everything from sources because
> binary distributions -- be they of Asterisk or any other Linux or Linux
> application -- are unreliable. Nobody knows what hardware you're running
> but you; compiling from sources gives you a better chance of ending up
> with a result that works. I used to use binary distributions; that's
> when I had the most trouble getting stuff working. I did one source
> installation and never looked back.

You can take those binary packages and rebuild them when you need so.
rpm, deb and similar provide a very strong method of reproducable
builds. 

Well-built packages also tend to work better than a simple 'make
install' because they are better debugged.

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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[asterisk-users] problems with trunks IAX2 and queues

[Nuria Fernandez]

Hi for all

I'm making some test and I can see an incorrect behaviour.

I have two asterisk with an IAX2 trunk. In asterisk 1 I have a queue and an
agent and, in Asterisk 2 I have three clients. When the clients make calls
to an asterisk 1, its calls entry in the queue. While they are waiting, an
agent login into the queue.

I'm waiting and waiting and waiting but the clients never contact with the
agent.

If I change the scenary and the clients call when the agent is previously
login, only the first client can contact with the agent.

Do you know something about it?

I'm using asterisk 1.2.15, zaptel 1.2.13 and libpri 1.2.4.
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Re: [asterisk-users] Recomended POE Phones

[Stephen Bosch]

voip crazy wrote:
> Hi all,
> 
> I am looking for phones witch support POE, with a good relation between
> quality and price to work with asterisk. I just see the Thompson st2030
> and the Linksys SPA 922 an SPA 942. Witch of this phones or another ones
> gave you the best results in a productivity enviroment?
> 
> Thanks in advance.
> 
> VoipCrazy.

Aren't the Polycom phones PoE?

-Stephen-
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Re: [asterisk-users] Digium Card ?

[Stephen Bosch]

Paul Hales wrote:
> On Mon, 2007-02-12 at 22:14 -0800, George Pajari wrote:
>> On Tue, 2007-02-13 at 16:24 +1100, Dennis Kavadas wrote:
>>
>>> Hi all
>>>
>>> I'm after a Digium card that will allow me to connect an Asterisk box to..
>>>
>>>  2 x sip providers
>>>  1 x company PBX
>>>  1 x POTS provider.
>>>
>>> Can anyone recomment a card that can do the job.
>>>   
>> To which Paul Hales wrote:
>>> TE210P?
>>>   
>> Pray tell which port on the TE210P you would suggest using for the 
>> (analog) POTS lines.
>>
> 
> The second one - with a channel bank attached to it.

I hope you're paying for it!

-stephen-
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RE: [asterisk-users] Nortel 81C MSDL Trunking to Asterisk TE110P, Nortel Resetting PRI Channels

[Don Pobanz]

[EMAIL PROTECTED] wrote on Tuesday, February 13, 2007 10:17
AM
> We are currently working to trunk from a Nortel 81C to an Asterisk 
> Server 1.4 running on Red Hat Linux.  We have two PRI trunks which 
> work with the exception of the clock slips, which is causing the 
> Nortel to reset the PRIs once a hour.  Thanks for any suggestions.

Assuming that 'clock slips' are the same thing as 'frame slips'

Where is your primary clock? Do you have an atomic clock supplying
timing to your Nortel system? Does your Nortel get it's clock from the
phone company? Assuming one of these two is the case, and that both T1s
are connected to the Nortel system, Asterisk should derive it's clock
from the Nortel system. 
To do that you would want to set up Zaptel.conf 
span=1,1,0,esf,b8zs
span=2,2,0,esf,b8zs

Asterisk would then use T1 number 1 to set it's internal t1 card clock.
If T1 1 was not available, then it would use T1 2. 

This line from your Nortel system makes me think it is using an external
clock. If this is the case then you must have an atomic clock or some
other 'external' clock source at your location. 

>  CLOK EXT

Don Pobanz
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Re: [asterisk-users] Trixbox vs. Custom install

[Stephen Bosch]

Tom Rymes wrote:
> On Feb 12, 2007, at 7:13 PM, Stephen Bosch wrote:
> 
>> Lee Jenkins wrote:
>>> Stefano Corsi wrote:
> 
>>> The nice things about GUI's in my opinion is that routine chores such as
>>> setting up extensions, dialing extensions, hunt groups, etc. are less
>>> likely to contain scripting bugs or typos.  The downside from what I
>>> gather with many GUI's is that the friendly abstraction that insulates
>>> you from the nuts and bolts of scripting and configuration also makes it
>>> difficult to customize the dialplan in some cases.
>>
>> It also makes troubleshooting problems a handful-and-a-half. And woe is
>> you if you need kernel customizations to make your hardware work.
> 
> Not to start a flame-war, but I completely disagree. Troubleshooting a
> GUI is much easier, given that you don't have to scout for typos,
> transposed numbers, etc throughout the dialplan. With the GUI, you have
> to double check the information that you input into the GUI, but that's
> it. As for hardware, it should be no more difficult to get Trixbox to
> play nicely with hardware than any other Asterisk install. You may have
> to patch and/or recompile zaptel, asterisk, etc, but that's no different
> than what you would have to do with a non-Trixbox install. (and you
> really shouldn't have to in almost all cases)

I come from the practice of compiling everything from sources because
binary distributions -- be they of Asterisk or any other Linux or Linux
application -- are unreliable. Nobody knows what hardware you're running
but you; compiling from sources gives you a better chance of ending up
with a result that works. I used to use binary distributions; that's
when I had the most trouble getting stuff working. I did one source
installation and never looked back.

Not for everybody, sure -- but I find I waste less time if I just build
the damn thing from scratch. There are distros that let you do this more
easily (Gentoo comes to mind).

And troubleshooting a GUI is *not* easier if there is something wrong
with the GUI. Now you're not troubleshooting anymore -- you're
debugging. How painful that is for me is a question of depth of
documentation. Trixbox' documentation is not great.

I'm not just shooting my mouth off. I speak from experience here.

>> I would say this -- if all you're ever going to use is VOIP trunks, by
>> all means use Trixbox. It's great for that. But if you're using any kind
>> of PSTN hardware (TDM cards, Sangoma) just stick with straight Asterisk.
> 
> Are you kidding? Sangoma actually has a version of Trixbox on their site
> that comes bundled with their drivers already installed (see
> http://wiki.sangoma.com/Trixbox-1xx ). All you have to do is configure
> the card(s) in the same way as you would with any Asterisk install.

Having to hunt around for packages and drivers in multiple locations
cancels the benefit of a "1 hour and you're up" install of anything. (I
respectfully challenge that assertion, anyway -- it was never in danger
of being anywhere near that for me, because things didn't work "out of
the box".)

>> Here's another reason to seriously consider generic: the userbase is
>> larger, AND they're more likely to know what they're talking about when
>> a problem does arise. Trixbox attracts a lot of amateurs who are
>> themselves new to IP telephony; that's why they choose it.
> 
> Valid point, but FreePBX (the program Trixbox uses for GUI Asteirsk
> config) also has a large userbase, and a number of Trixbox problems are
> not Trixbox specific, and can be addressed by the Asterisk community as
> a whole.

Have a look at the list archives and see how Trixbox questions are
handled by the list membership.

It doesn't build confidence.

>>> Of course, you should take this with a grain of salt since I tried [EMAIL 
>>> PROTECTED]
>>> (now TrixBox) for a total of 2 weeks before gutting it.
>>
>> There is a good reason people don't stick with it for long.
> 
> Many people do not stick with Trixbox for long, and many others do. The
> crux of the issue is this: FreePBX/Trixbox, and most other GUIs will
> make it easier to get your system up and running, and they make it
> easier to maintain it, make changes, etc. (I am defining "easier" as
> "requiring less technical familiarity with the underpinnings of exactly
> what is going on" as well as "less intimidating and error prone since no
> manual editing of configuration files is required.")

Fair enough -- and this would be fine for me if things "just worked".
They often don't. Then I'm back to

> On the other hand,
> emacs/vi/pico/whatevereditoryouprefer and the text config files without
> a GUI are more difficult, but offer greater flexibility

with all of the disadvantages and none of the advantages.

Anyway, that was my input; your mileage may vary.

-Stephen-
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Re: [asterisk-users] Trixbox vs. Custom install

[Tzafrir Cohen]

On Tue, Feb 13, 2007 at 10:23:17AM -0500, Tom Rymes wrote:
> On Feb 12, 2007, at 7:13 PM, Stephen Bosch wrote:
> 
> >Lee Jenkins wrote:
> >>Stefano Corsi wrote:
> 
> [snip]
> 
> >>The nice things about GUI's in my opinion is that routine chores  
> >>such as
> >>setting up extensions, dialing extensions, hunt groups, etc. are less
> >>likely to contain scripting bugs or typos.  The downside from what I
> >>gather with many GUI's is that the friendly abstraction that  
> >>insulates
> >>you from the nuts and bolts of scripting and configuration also  
> >>makes it
> >>difficult to customize the dialplan in some cases.
> >
> >It also makes troubleshooting problems a handful-and-a-half. And  
> >woe is
> >you if you need kernel customizations to make your hardware work.
> 
> Not to start a flame-war, but I completely disagree. Troubleshooting  
> a GUI is much easier, given that you don't have to scout for typos,  
> transposed numbers, etc throughout the dialplan. With the GUI, you  
> have to double check the information that you input into the GUI, but  
> that's it. As for hardware, it should be no more difficult to get  
> Trixbox to play nicely with hardware than any other Asterisk install.  
> You may have to patch and/or recompile zaptel, asterisk, etc, but  
> that's no different than what you would have to do with a non-Trixbox  
> install. 

Hmmm... I installed a trixbox system. 'yum update' failed to work, due
to funny games with yum's configuration. A default centos server
installation did not have the same issue.

This is just one example.

> (and you really shouldn't have to in almost all cases)

A GUI does its absraction. By that it hides some information that it
deems irrelevant. In many cases this information is relevant.

One example: just figuring out if FreePBX actually dial, or not at all,
requires either a sufficiently-trained asterisk guy to review the
log/cli just to understand why a call did not go through. 

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] error when compiling asterisk-1.4

[younss azzayani]

this is the solution
apt-get install g++
it's work know
thnank you (all of you) :)
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FW: [asterisk-users] After upgrade to 1.4 transfers don't workproperly

[Savoy, Kevin - Williston, ND]

No one knows what the Notify answer on an owned channel is?

Anyone?

-Original Message-
From: Savoy, Kevin - Williston, ND 
Sent: Monday, February 12, 2007 11:14 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: FW: [asterisk-users] After upgrade to 1.4 transfers don't workproperly

Sorry if this is a repeat but I didn't receive a copy of it so I'm not sure it 
actually posted.


The below worked for normal transfers. Now here is another situation. When we 
try to transfer a call directly to voicemail it plays the voicemail message but 
we can't transfer the call. The only way I could get it to work was to do a 
conference and then drop out of that conference.

My dial plan for direct dialing is:

exten=>_*40XX,n,Voicemail(${EXTEN:1},u)

When this is attempted the following message shows up on the CLI of Asterisk:

[Feb  9 11:52:03] WARNING[5054]: chan_sip.c:12310 handle_response: Notify 
answer on an owned channel?

Can anyone tell me what this means and what I can do to fix this?

Thanks

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Chavez
Sent: Wednesday, February 07, 2007 5:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] After upgrade to 1.4 transfers don't workproperly

On Wed, 2007-02-07 at 14:12 -0600, Savoy, Kevin - Williston, ND wrote:
> I have discovered an issue on my system after upgrading from 1.2.13 to
> 1.4. A call comes in on a T1 line and goes to a Polycom 501 SIP phone.
> I have confirmed this on multiple phones. When the called person
> answers and tries to transfer the call to another extension, the call
> successfully transfers, however the person answering the transfer
> cannot hear the person that called in, the caller. My dial command
> simply is 
> 
>  
> 
I had exactly the same problem when upgrading to 1.4 and I solved by
making sure canreinvite=no is in sip.conf for every phone.

> 
-- 
Telecomunicaciones Abiertas de Mexico S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001
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[asterisk-users] End Wrap-up Time?

[James Fromm]

Does anyone have a solution to allow an agent to selectively end his
wrap-up time?  We define a wrap-up time of 60 seconds to allow our
agents to finish their notes from a call.  In some cases, the full 60
seconds is not needed and our agents would like to end their wrap-up time.

Thanks,
Jay


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Re: [asterisk-users] error when compiling asterisk-1.4

[younss azzayani]

but i can't work with yum am using debian not RH

2007/2/13, Savoy, Kevin - Williston, ND <[EMAIL PROTECTED]>:

Try "yum install gcc-c++"

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of younss
azzayani
Sent: Tuesday, February 13, 2007 8:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] error when compiling asterisk-1.4

i cant make "apt-get install gcc-c++" it s result nothing
so i typed "apt-cache search gcc-c++" no thing also
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[asterisk-users] Nortel 81C MSDL Trunking to Asterisk TE110P, Nortel Resetting PRI Channels

[Marlon_Blair]

We are currently working to trunk from a Nortel 81C to an Asterisk
Server 1.4 running on Red Hat Linux.  We have two PRI trunks which work
with the exception of the clock slips, which is causing the Nortel to
reset the PRIs once a hour.  Thanks for any suggestions.

81C   MSDL  Asterisk Digium
TE110P
 REQ  prt
 TYPE adan dch 10Zaptel.conf
loadzone
= us
 ADAN DCH 51 defaultzone=us
   CTYP MSDL   span=1,1,0,esf,b8zs
(also tried 1,0,0,esf,b8zs)  
   GRP  1 bchan=1-23
   DNUM 7 dchan=24
   PORT 1   span=2,1,0,esf,b8zs
(also tried 2,0,0,esf,b8zs)
   DES  Asterisk VOIP bchan=25-47
   USR  PRI  dchan=48
   DCHL 51
   OTBF 32 zapata.conf
   PARM RS422  DTE [channels]
   DRAT 64KC  language=en
   CLOK EXT  context=default
   IFC  ESS5   switchtype=5ess
   SIDE USR
signalling=pri_net
   CNEG 1group=1
   RLS  ID  1   channel =>
1-23
   RCAP ND2   channel => 25-47
   MBGA NOusecallerid=yes
   OVLR NO   hidecallerid=no
   OVLS NO   callwaiting=no
   T200 3
threewaycalling=yes
   T203 10  transfer=yes
   N200 3canpark=yes
   N201 260  cancallforward=yes
   K7
echocancancel=no
 
echocancelwhenbridged=no
 And I have immediate=no
ADAN DCH 71   callreturn=yes
Built the same asrxgain=0.0

ADAN DCH 31  txgain=0.0
 
musiconhold=default
 
ROUT 1 We start up
Asterisk in the following 
Type RDB   order:
CUST 00 modprobe zaptel
Rout 30 modprobe
wcte11xp
DES  ASTERISK_VOIP_1   ztcfg
TKTP TIE safe_asterisk

NPID_TBL_NUM   0
ESN  NO  
CNVT NO  
SAT  NO  
RCLS INT 
VTRK NO  
DTRK YES 
BRIP NO  
DGTP PRI 
ISDN YES 
MODE PRA 
IFC  ESS5
SBN  NO
PNI  1 
SRVC NNSF
NCNA YES 
NCRD YES 
CHTY BCH 
CTYP CDP 
INAC YES 
ISAR NO  
CPUB OFF 
DAPC NO  
BCOT 0 
DSEL VOD 
PTYP PRI 
AUTO NO  
DNIS NO  
DCDR NO  
ICOG IAO 
SRCH LIN 
TRMB YES 
STEP 
ACOD 7930
TCPP NO  
PII NO  
TARG 
CLEN 1 
BILN NO
OABS 
INST 
IDC  NO  
DCNO 0 *
NDNO 0 
DEXT NO  
ANTK 
SIGO STD 
ICIS YES
TIMR ICF  512 
 OGF  512 
 EOD  13952 
 NRD  10112 
 DDL  70 
 ODT  4096 
 RGV  640 
 GRD  896 
 SFB  3 
 NBS  2048 
 NBL  4096 
 IENB  5 
 TFD  0 
 VSS  0 
 VGD  6 
DRNG NO  
CDR  NO  
VRAT NO  
MUS  NO  
FRL  0 0 
FRL  1 0 
FRL  2 0 
FRL  3 0 
FRL  4 0 
FRL  5 0 
FRL  6 0 
FRL  7 0 
OHQ  NO  
OHQT 00 
CBQ  NO  
AUTH NO  
TDET NO  
TTBL 0 
ATAN NO  
PLEV 2 
ALRM NO  
ART  0 
SGRP 0 
AACR NO


 DES  VERSA
 TN   101 01
 TYPE TIE
 CDEN SD
 CUST 0
 TRK  PRI
 PDCA 1
 PCML MU
 NCOS 0
 RTMB 1 73
 B-CHANNEL SIGNALING
 TGAR 1
 AST  NO
 IAPG 0
 CLS  UNR DTN WTA LPR APN THFD HKD
  P10 VNL
 TKID

>ld 22
PT2000 
MARP NOT ACTIVATED
CEQU 
  DLOP  NUM DCH FRM TMDI LCMT YALM TRSH
   PRI  012 24  ESF NO   B8S  FDL  00 
051 24  ESF NO   B8S  FDL  00 
073 24  ESF NO   B8S  FDL  00 


RLI  30 
ENTR 0 
LTER NO
ROUT 30 
TOD  0 ON  1 ON  2 ON  3 ON  
 4 ON  5 ON  6 ON  7 ON  
VNS  NO
CNV  NO
EXP  NO
FRL  0 
DMI  30 
FCI  0 
FSNI 0 
SBOC NRR 
IDBB DBD 
IOHQ NO
OHQ  NO
CBQ  NO
ISET 0 
NALT 5 
MFRL 2 
OVLL 0

Marlon Blair   e-mail  [EMAIL PROTECTED]
> DOH, Network System Analyst
> (850) 245-4400, S/C 205-4400, Cell (850) 528-4244
> Fax (850) 412-1148
> Work Hours 7 AM to 3:30 PM
> 
> ==
> MISSION:  To promote and protect the health and safety of all people
> in Florida through the delivery of quality public health services and
> promotion of health care standards
> *
> PLEASE NOTE:  Florida has a very broad public records law.  Most
> written communications to or from State officials regarding state
> business a

[asterisk-users] AGI "GET DATA" and "WAIT FOR DIGIT" don't work

[Camilo Echeverry]

Hi.
I'm trying to get digits form the user via agi
something like this: this only should print result=asciicode

but none of the functions even wait until timeout ..
they just pass .. (after a nanosecond)

the las print is always timeout.

Any clue ..?


my $callerid = $AGI{'callerid'} ;
if($callerid !~ /[0-9]{7,20}/){
  #way numbre one
  print "EXEC PLAYBACK  please_enter_your_number \"\"\n";  my $result =
;
  print "WAIT FOR DIGIT 3000\n"; my $result = ;

  # Way number two
  # print "GET DATA   please_enter_your_number \"-1\" \"10\""; my $result =
;

}
print STDERR "$result";



--
Camilo Echeverry

Your life would be very empty if you had nothing to regret.
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RE: [asterisk-users] error when compiling asterisk-1.4

[Savoy, Kevin - Williston, ND]

Try "yum install gcc-c++"

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of younss
azzayani
Sent: Tuesday, February 13, 2007 8:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] error when compiling asterisk-1.4

i cant make "apt-get install gcc-c++" it s result nothing
so i typed "apt-cache search gcc-c++" no thing also
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Re: [asterisk-users] Customisable In-band ringing?

[Yuan LIU]

From: Paul Hales <[EMAIL PROTECTED]>
Date: Tue, 13 Feb 2007 22:15:57 +1100

The 'm' option in the dial command, from memory.

PaulH


Also search for early media - I'm under the impression that you may not need 
progressinband, as it is often undesirable.


Yuan Liu


On Tue, 2007-02-13 at 23:25 +1300, Ray Jackson wrote:
> All,
>
> Using SIP with progressinband=yes I get Asterisk to generate the ringing
> sound for callers.  However, I was wondering if it is possible to
> configure what is 'played back' to the calling party?  i.e. instead of
> just 'ring ring' could I potentially play back a song from an MP3, WAV
> or GSM file?  I'm thinking it would be quite cool to offer a customised
> 'ring' sound while the caller is waiting for you to pick up?  How can I
> do this with Asterisk or some external module perhaps?  Any advice 
welcome!

>
> Cheers,
> Ray



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RE: [asterisk-users] question about regex

[Yuan LIU]

From: "Rilawich Ango" <[EMAIL PROTECTED]>
Date: Tue, 13 Feb 2007 17:43:05 +0800

Hi,  I have tried the regex function below with MACRO_EXTEN=5000*.
However, both of them return 0 instead 1 to me.  How can I search the 
character in the end of line?


${REGEX("[*]$" ${MACRO_EXTEN})
returns 0


You must be using Asterisk 1.2.10 or something.  Take a look at 
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+regex.  The 
workaround is funky: using a variable to represent special characters like 
$.


Yuan Liu


${REGEX("*$" ${MACRO_EXTEN})
returns 0 with error

ango



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[asterisk-users] Paging Followup

[Rob Schall]

Hello All,

Hoping all of you might have an additional option for me to try at this
point. :)

My Goal:
To have a paging option that does the following When I press **_
it will send a ring-answer page to that person. The person on the other
end should be muted, so if they are in a conference, you can't hear what
is going on in the meeting. If that person hears me and decides they
want to talk to me, they can press *1 (or something similar).

My Problems:
Paging solves the first part of my issue (muted meetme) and I have the
Ring-Answer part working great. However, I can't get it to accept *1 or
anything like that to accept the call and talk.

I've Tried:
I've tried using Dial and Page commands to make this work. I noticed the
options that are sent with the page command (it generates a Meetme for
both ends of the call). I think I could just add a "s" to the meetme
options. I believe I should be able to do 2 meetme commands. One for me,
and one for the person I am paging, and we will connect to the same
conference.

But short of creating a call file, is this possible?

Any thoughts?
Rob

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Re: [asterisk-users] Trixbox vs. Custom install

[Tom Rymes]

On Feb 12, 2007, at 7:13 PM, Stephen Bosch wrote:


Lee Jenkins wrote:

Stefano Corsi wrote:


[snip]

The nice things about GUI's in my opinion is that routine chores  
such as

setting up extensions, dialing extensions, hunt groups, etc. are less
likely to contain scripting bugs or typos.  The downside from what I
gather with many GUI's is that the friendly abstraction that  
insulates
you from the nuts and bolts of scripting and configuration also  
makes it

difficult to customize the dialplan in some cases.


It also makes troubleshooting problems a handful-and-a-half. And  
woe is

you if you need kernel customizations to make your hardware work.


Not to start a flame-war, but I completely disagree. Troubleshooting  
a GUI is much easier, given that you don't have to scout for typos,  
transposed numbers, etc throughout the dialplan. With the GUI, you  
have to double check the information that you input into the GUI, but  
that's it. As for hardware, it should be no more difficult to get  
Trixbox to play nicely with hardware than any other Asterisk install.  
You may have to patch and/or recompile zaptel, asterisk, etc, but  
that's no different than what you would have to do with a non-Trixbox  
install. (and you really shouldn't have to in almost all cases)



I would say this -- if all you're ever going to use is VOIP trunks, by
all means use Trixbox. It's great for that. But if you're using any  
kind
of PSTN hardware (TDM cards, Sangoma) just stick with straight  
Asterisk.


Are you kidding? Sangoma actually has a version of Trixbox on their  
site that comes bundled with their drivers already installed (see  
http://wiki.sangoma.com/Trixbox-1xx ). All you have to do is  
configure the card(s) in the same way as you would with any Asterisk  
install.



I've just had my second go at Trixbox (version 2.0 now) and after
wasting a bunch of time with hardware problems, I'm going to  
replace it

with a generic install.


I would suggest (hopefully politely) that you not blame your lack of  
experience and ability on Trixbox. If you can get the Sangoma  
wanrouter software downloaded and compiled, along with Zaptel,  
Asterisk, libpri, etc, then you can certainly do the same on Trixbox,  
because all you have to do is "yum search wanpipe"  and then "yum  
install" the modules and utils packages. Once installed, follow the  
instructions on Sangoma's website to configure the card. If all else  
fails, you can easily call for support from Sangoma. Even if you  
choose not to use yum, it's just as easy to get a Sangoma board  
working under Trixbox as it is for any other Asterisk install.



Here's another reason to seriously consider generic: the userbase is
larger, AND they're more likely to know what they're talking about  
when

a problem does arise. Trixbox attracts a lot of amateurs who are
themselves new to IP telephony; that's why they choose it.


Valid point, but FreePBX (the program Trixbox uses for GUI Asteirsk  
config) also has a large userbase, and a number of Trixbox problems  
are not Trixbox specific, and can be addressed by the Asterisk  
community as a whole.


Of course, you should take this with a grain of salt since I tried  
[EMAIL PROTECTED]

(now TrixBox) for a total of 2 weeks before gutting it.


There is a good reason people don't stick with it for long.


Many people do not stick with Trixbox for long, and many others do.  
The crux of the issue is this: FreePBX/Trixbox, and most other GUIs  
will make it easier to get your system up and running, and they make  
it easier to maintain it, make changes, etc. (I am defining "easier"  
as "requiring less technical familiarity with the underpinnings of  
exactly what is going on" as well as "less intimidating and error  
prone since no manual editing of configuration files is required.")  
On the other hand, emacs/vi/pico/whatevereditoryouprefer and the text  
config files without a GUI are more difficult, but offer greater  
flexibility.


S it comes down to "Which is more important to you? Ease of  
use for you and/or your clients (who may want to control adds/moves,  
etc.) or greater flexibility and control?" Once you answer that  
question, you can answer the question "Which is better for me?" The  
correct answer to that question may very well be different for you  
than it is for me. (and it may be different for you six months from  
now than it is today.)


Tom

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Re: [asterisk-users] error when compiling asterisk-1.4

[younss azzayani]

sorry,
#cat /etc/apt/sources.list
#deb file:///cdrom/ sarge main

deb ftp://ftp2.fr.debian.org/debian/ stable main
deb-src ftp://ftp2.fr.debian.org.debian/ stable main
deb http://security.debian.org/ stable/updates main
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Re: [asterisk-users] error when compiling asterisk-1.4

[younss azzayani]

when i type " cat /etc/apt/sources.list | grep gcc-c++" i got nothing
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Re: [asterisk-users] Recomended POE Phones

[Tom]

I have installed both the Cisco 79xx and Linksys SPA942.  I 
personally have Cisco on my desk because there is quite a bit of 
difference in the look and feel.  Also the sound quality is better on 
the Cisco phones.


The SPA942 is a nice phone for the price but is lighter and smaller 
than the Cisco 79xx phones.  We offer both to our business 
customers.  We have a number of choices in our demo room.


The ST2030 looks nice in pictures but my demo hasn't arrived yet.

Tom

At 06:41 AM 2/13/2007, you wrote:

I've just setup about 10 SPA942s. Great phones. Has the look and feel
of a Cisco phone. Documentation for configuring the phone remotely is
not easily accessible but (with the help of the all knowing Google)
found on the Internet. Have them connected to a POE switch also from
Linksys.

On 13 Feb 2007, at 3:24 AM, voip crazy wrote:


Hi all,

I am looking for phones witch support POE, with a good relation
between quality and price to work with asterisk. I just see the
Thompson st2030 and the Linksys SPA 922 an SPA 942. Witch of this
phones or another ones gave you the best results in a productivity
enviroment?

Thanks in advance.

VoipCrazy.
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Johan Martin
Catenare LLC
534 Pacific Ave
San Francisco, CA. 94133

Phone: (415) 834-9802
Fax: (415) 294-4495
http://www.catenare.com

AOL: catenarellc
Yahoo: martin_johan
GTalk: [EMAIL PROTECTED]
MSN: [EMAIL PROTECTED]

FreeWorldDialup  :716798  - http://www.freeworlddialup.com/
Gizmo Project: 747-627-9132 - http://www.gizmoproject.com/

Skype: catenare
http://www.skype.com

http://www.linkedin.com/in/catenare



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Re: [asterisk-users] error when compiling asterisk-1.4

[Tzafrir Cohen]

On Tue, Feb 13, 2007 at 02:37:05PM +, younss azzayani wrote:
> i cant make "apt-get install gcc-c++" it s result nothing
> so i typed "apt-cache search gcc-c++" no thing also

What do you have on /etc/apt/sources.list ?

-- 
   Tzafrir Cohen   
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RE: [asterisk-users] Using Asterisk/callerid with "pay as you go"

[Doug Crompton]

Yes, thank you. I found one, "callwithus" which has excellant Asterisk
support, IAX/SIP and the support actually answered in minutes! So far good
connects (usig IAX) and good prices. Lets hope it stays that way.

I wonder why more companies can't be like that. This callerID thing is
stupid. If you can go to many companies and can set it then why don't all
companies offer that feature? It certainly would be a customer draw.

Doug

On Tue, 13 Feb 2007, Dovid B wrote:

> If you asked this question on the biz list you would get a lot of people
> that will tell you that they offer services where you can set the caller ID
> to what ever you want. To name a few::
> Nufone
> Teliax
> Voipjet
>
> - Original Message -
> From: "Doug Crompton" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> 
> Sent: Monday, February 12, 2007 10:33 PM
> Subject: [asterisk-users] Using Asterisk/callerid with "pay as you go"
> VOIPproviders
>
>
> >I am curious how others handle "call out" VOIP and callerid. I have found
> > numerous providers that are cheap and seem to have good voice quality but
> > offer no provisions for callerid.  I find it almost useless to use call
> > out when the receiving party gets some bogus callerid number that has no
> > relation to my call.
> >
> > I understand the big thing is spoofing callerid but are there any
> > companies that offer a means of qualifying callerid so it works right?
> >
> > Like it or not callerid is used heavily and without a proper return ID
> > many callee's don't answer and if they tried to return the call they get
> > no where. Seems like a big problem to me.
> >
> > Very aggrevating.
> >
> > Doug
> >
> > ___
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> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>


"Those that sacrifice essential liberty to obtain a little temporary safety
 deserve neither liberty nor safety."  -- Ben Franklin (1759)


*  Doug Crompton   *
*  Richboro, PA 18954  *
*  215-431-6307*
*  *
* [EMAIL PROTECTED]*
* http://www.crompton.com  *



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Re: [asterisk-users] error when compiling asterisk-1.4

[younss azzayani]

i cant make "apt-get install gcc-c++" it s result nothing
so i typed "apt-cache search gcc-c++" no thing also
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[asterisk-users] Using Dynamic Groups instead of AgentCallbackLogin - how to log which agent took the call?

[James FitzGibbon]

Hello all.

I'm setting up a new call center PBX using * v1.4, and figure it's better to
go with AddQueueMember over AgentCallbackLogin.  The functionality of
AgentCallbackLogin still works, but without a firm idea of how long it will
be in the codebase, I'm wary of building a system on top of it.

The basic mechanics work, but I'm having some trouble on logging.  With
AgentCallbackLogin, ${MEMBERINTERFACE} and the entry in the queue log refer
to the agent that the call was sent to.  With AddQueueMember, the log entry
refers to the channel (in my case, SIP/).  To tie call disposition back
to a person, I need to maintain a map of channels to agents and then
populate that in the CDR somehow.

This isn't a huge problem, as I already have to set up an agent login
context in the dialplan, so when I authenticate the agent and call
AddQueueMember, I can easily stick the mapping into AstDB, then pull it out
before the call to Queue() and stick it in the CDR userfield.  I can't think
of a way to put the agent in the queue log however, unless I'm missing a way
to customize what shows up in queue.log.

I could do the mapping in post-processing by having my agent login
application store the login / logout times, the channel and the agent, but
any missing data found in post would not easily be correctable in a billing
program.

Am I missing something obvious, or is this just part and parcel of using
dynamic queues instead of AgentCallbackLogin?

Thanks

--
j.
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RE: [asterisk-users] error when compiling asterisk-1.4

[Savoy, Kevin - Williston, ND]

Sounds like you don't have the gcc-c++ package installed. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
Sent: Tuesday, February 13, 2007 6:53 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] error when compiling asterisk-1.4

On Tue, Feb 13, 2007 at 12:30:06PM +, younss azzayani wrote:
> hi,
> when i type
> 
> asterisk-1.4# ./configure
> **
> i got this error
> 
> configure: error: C++ preprocessor "/lib/cpp" fails sanity check
> See `config.log' for more details.
> *
> # vi config.log
> ***
> ...;
> cpp: installation problem, cannot exec 'cc1plus' : No such file or
directory
> ..
> 
> ./configure : line 1: g++: command not found

g++ ?

apt-get install gcc-c++

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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RE: [asterisk-users] Using Asterisk/callerid with "pay as you go"

[Dovid B]

If you asked this question on the biz list you would get a lot of people 
that will tell you that they offer services where you can set the caller ID 
to what ever you want. To name a few::
Nufone
Teliax
Voipjet

- Original Message - 
From: "Doug Crompton" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Monday, February 12, 2007 10:33 PM
Subject: [asterisk-users] Using Asterisk/callerid with "pay as you go" 
VOIPproviders


>I am curious how others handle "call out" VOIP and callerid. I have found
> numerous providers that are cheap and seem to have good voice quality but
> offer no provisions for callerid.  I find it almost useless to use call
> out when the receiving party gets some bogus callerid number that has no
> relation to my call.
>
> I understand the big thing is spoofing callerid but are there any
> companies that offer a means of qualifying callerid so it works right?
>
> Like it or not callerid is used heavily and without a proper return ID
> many callee's don't answer and if they tried to return the call they get
> no where. Seems like a big problem to me.
>
> Very aggrevating.
>
> Doug
>
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> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
> 
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[asterisk-users] Zaptel Won't Compile In Slackware 11.0 Kernel 2.4.33

[demuel]

Anybody,


I have download asterisk 1.4 via svn. whem I compiled it, I got the following 
error:


/lib/modules/2.4.33.3/build/include/asm/system.h:190: warning: dereferencing 
type-punned pointer
will break strict-aliasing rules
zttranscode.c:37:30: linux/page-flags.h: No such file or directory
make[1]: *** [zttranscode.o] Error 1
make[1]: Leaving directory 
`/home/kingkong/code/projects/asterisk/source/zaptel-1.4'
make: *** [all] Error 2


Any comments?


Regards,
Demuel



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[asterisk-users] Originating calls: Astmanproxy vs Direct Connection vs Call files

[Julian Lyndon-Smith]

I've got around 45 people who need to place calls from our inhouse app. 
What is the considered "best practice" for placing these calls:


1) All clients connect to astmanproxy, and use AMI API Originate command
2) All clients connect directly to the astersik AMI and use the API 
Originate command
3) All clients create a db record, some process reads the record and 
writes out a call file
4) All clients connect to a web / socket service which then writes a 
call file ...


We are currently using #1, but every few days or so, the "putting" of 
data seems to take longer and longer. If I kill astmanproxy and restart 
it, the system is lighting fast for another few days.


What settings would I need for astmanproxy in order simply to access the 
AMI API (I don't need to receive or process events from asterisk)


Julian
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Re: [asterisk-users] error when compiling asterisk-1.4

[Tzafrir Cohen]

On Tue, Feb 13, 2007 at 12:30:06PM +, younss azzayani wrote:
> hi,
> when i type
> 
> asterisk-1.4# ./configure
> **
> i got this error
> 
> configure: error: C++ preprocessor "/lib/cpp" fails sanity check
> See `config.log' for more details.
> *
> # vi config.log
> ***
> ...;
> cpp: installation problem, cannot exec 'cc1plus' : No such file or directory
> ..
> 
> ./configure : line 1: g++: command not found

g++ ?

apt-get install gcc-c++

-- 
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Re: [asterisk-users] problem with safe_asterisk

[Tzafrir Cohen]

On Tue, Feb 13, 2007 at 01:35:50PM +0100, Andrea De Vita wrote:
> Hi all,
> 
> I have installed some Asterisk machine, all with the same problem.
> My typical configuration is:
> - Asterisk 1.2.14 (or 1.4.0beta3)
> - CentOS 4.4 server.
> 
> 
> The problem is this:
> When I start Asterisk with the default init script (/etc/init.d/asterisk 
> start) distributed with the source, and kill (or kill -9) Asterisk-pid,
> then safe_asterisk doesn't correctly work (it dies and not restart 
> Asterisk).
> Instead, if I start Asterisk with safe_asterisk command from shell, 
> after "kill Asterisk-pid", safe_asterisk restart Asterisk correctly.
> 
> I would use the init script because I like to use Linux-HA that require 
> this.

Edit that init.d script lightly not to use safe_asterisk.
safe_asterisk is not close to robust anyway, and thus will only
complicate things.

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

[Catenare LLC]

I've just setup about 10 SPA942s. Great phones. Has the look and feel  
of a Cisco phone. Documentation for configuring the phone remotely is  
not easily accessible but (with the help of the all knowing Google)  
found on the Internet. Have them connected to a POE switch also from  
Linksys.


On 13 Feb 2007, at 3:24 AM, voip crazy wrote:


Hi all,

I am looking for phones witch support POE, with a good relation  
between quality and price to work with asterisk. I just see the  
Thompson st2030 and the Linksys SPA 922 an SPA 942. Witch of this  
phones or another ones gave you the best results in a productivity  
enviroment?


Thanks in advance.

VoipCrazy.
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Johan Martin
Catenare LLC
534 Pacific Ave
San Francisco, CA. 94133

Phone: (415) 834-9802
Fax: (415) 294-4495
http://www.catenare.com

AOL: catenarellc
Yahoo: martin_johan
GTalk: [EMAIL PROTECTED]
MSN: [EMAIL PROTECTED]

FreeWorldDialup  :716798  - http://www.freeworlddialup.com/
Gizmo Project: 747-627-9132 - http://www.gizmoproject.com/

Skype: catenare
http://www.skype.com

http://www.linkedin.com/in/catenare



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[asterisk-users] problem with safe_asterisk

[Andrea De Vita]

Hi all,

I have installed some Asterisk machine, all with the same problem.
My typical configuration is:
- Asterisk 1.2.14 (or 1.4.0beta3)
- CentOS 4.4 server.


The problem is this:
When I start Asterisk with the default init script (/etc/init.d/asterisk 
start) distributed with the source, and kill (or kill -9) Asterisk-pid,
then safe_asterisk doesn't correctly work (it dies and not restart 
Asterisk).
Instead, if I start Asterisk with safe_asterisk command from shell, 
after "kill Asterisk-pid", safe_asterisk restart Asterisk correctly.


I would use the init script because I like to use Linux-HA that require 
this.


Someone can help me?

Thanks

Andrea De Vita

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Re: [asterisk-users] Recomended POE Phones

[Tijl Van den Broeck]

Whatever you take:
Stay away from cisco poe phones unless you're using cisco poe
switches.. and even then. Cisco doesn't always apply the POE standard,
older models are totally not conform the POE standard (they switched
the + and - poles at the socket).
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[asterisk-users] error when compiling asterisk-1.4

[younss azzayani]

hi,
when i type

asterisk-1.4# ./configure
**
i got this error

configure: error: C++ preprocessor "/lib/cpp" fails sanity check
See `config.log' for more details.
*
# vi config.log
***
...;
cpp: installation problem, cannot exec 'cc1plus' : No such file or directory
..

./configure : line 1: g++: command not found
*

Can you Help Me Please
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Re: [asterisk-users] Recomended POE Phones

[Dave Cotton]

On Tue, 2007-02-13 at 12:24 +0100, voip crazy wrote:
> Hi all,
> 
> I am looking for phones witch support POE, with a good relation
> between quality and price to work with asterisk. I just see the
> Thompson st2030 and the Linksys SPA 922 an SPA 942. Witch of this
> phones or another ones gave you the best results in a productivity
> enviroment? 

Aastra 9133i and 480i


-- 
Dave Cotton <[EMAIL PROTECTED]>

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RE: [asterisk-users] Recomended POE Phones

[Ahsan Masood]

Hi,

 

Following are the commonly used POE enabled phones

 

Sipura 942 

Snom 320 and 360

GXP2000

Aastra 9133i and 480i

 

Ahsan

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of voip crazy
Sent: 13 February 2007 11:24
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Recomended POE Phones

 

Hi all,

I am looking for phones witch support POE, with a good relation between
quality and price to work with asterisk. I just see the Thompson st2030
and the Linksys SPA 922 an SPA 942. Witch of this phones or another ones
gave you the best results in a productivity enviroment? 

Thanks in advance.

VoipCrazy.

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Re: [asterisk-users] Recomended POE Phones

[Paul Hales]

Snom 320.

PaulH

On Tue, 2007-02-13 at 12:24 +0100, voip crazy wrote:
> Hi all,
> 
> I am looking for phones witch support POE, with a good relation
> between quality and price to work with asterisk. I just see the
> Thompson st2030 and the Linksys SPA 922 an SPA 942. Witch of this
> phones or another ones gave you the best results in a productivity
> enviroment? 
> 
> Thanks in advance.
> 
> VoipCrazy.
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> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

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[asterisk-users] Recomended POE Phones

[voip crazy]

Hi all,

I am looking for phones witch support POE, with a good relation between
quality and price to work with asterisk. I just see the Thompson st2030 and
the Linksys SPA 922 an SPA 942. Witch of this phones or another ones gave
you the best results in a productivity enviroment?

Thanks in advance.

VoipCrazy.
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Re: [asterisk-users] Customisable In-band ringing?

[Paul Hales]

The 'm' option in the dial command, from memory.

PaulH

On Tue, 2007-02-13 at 23:25 +1300, Ray Jackson wrote:
> All,
> 
> Using SIP with progressinband=yes I get Asterisk to generate the ringing 
> sound for callers.  However, I was wondering if it is possible to 
> configure what is 'played back' to the calling party?  i.e. instead of 
> just 'ring ring' could I potentially play back a song from an MP3, WAV 
> or GSM file?  I'm thinking it would be quite cool to offer a customised 
> 'ring' sound while the caller is waiting for you to pick up?  How can I 
> do this with Asterisk or some external module perhaps?  Any advice welcome!
> 
> Cheers,
> Ray
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Re: [asterisk-users] Can Asterisk handle 7000 SIP users?

[Marnus van Niekerk]

Copy and paste from my reply to a similar question a couple of weeks ago:

5000 sip registrations is quite a lot, but the more important thing is 
the number of simultaneous calls.
If most of your calls is going to be SIP 2 SIP then I would suggest you 
use openSER for the SIP registrations and most SIP call routing and use 
asterisk only for calls to/from PSTN and media such as voicemail and 
announcements.


openSER is made for this and is a lot faster at doing SIP call setup.  
Then use asterisk where it is good.
There are good examples of setting up openSER with asterisk on the net 
sharing a MySQL DB for users, auth etc.


Have a look at 
http://openser.org/dokuwiki/doku.php/asterisk:realtime-integration and 
other Asterisk related info at http://openser.org/dokuwiki/doku.php



M

Dominik Zalewski wrote:

Hi All,

One of my customer asked me if Asterisk can handle 7000 SIP users. They want  
anyone that have access to wireless hotspot to make voice calls to the office 
using software phone or SIP cordless phone.


 Does anybody did such a setup? What are hardware requirements for server and 
how much bandwidth I will need using comercial codec?



Thank you in advance,

Dominik
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[asterisk-users] Can Asterisk handle 7000 SIP users?

[Dominik Zalewski]

Hi All,

One of my customer asked me if Asterisk can handle 7000 SIP users. They want  
anyone that have access to wireless hotspot to make voice calls to the office 
using software phone or SIP cordless phone.

 Does anybody did such a setup? What are hardware requirements for server and 
how much bandwidth I will need using comercial codec?


Thank you in advance,

Dominik
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[asterisk-users] Customisable In-band ringing?

[Ray Jackson]

All,

Using SIP with progressinband=yes I get Asterisk to generate the ringing 
sound for callers.  However, I was wondering if it is possible to 
configure what is 'played back' to the calling party?  i.e. instead of 
just 'ring ring' could I potentially play back a song from an MP3, WAV 
or GSM file?  I'm thinking it would be quite cool to offer a customised 
'ring' sound while the caller is waiting for you to pick up?  How can I 
do this with Asterisk or some external module perhaps?  Any advice welcome!


Cheers,
Ray
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[asterisk-users] Error when compiling zaptel 1.2

[younss azzayani]

Hello
when i try to compile zaptel i get this error code line
any Help & explain please :)
*
ipbx:/usr/src/zaptel-1.2#uname -r
2.6.8-3-686
ipbx:/usr/src/zaptel-1.2#make linux26
make: *** No rule to make target `linux26` . Stop.
***

Kind Regards

Younss AZ
KASTERISK.COM
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Re: [asterisk-users] asterisk-sounds doesn't exist in the sources, how can i get it?

[younss azzayani]

when i check this link
"http://ftp.digium.com/pub/asterisk/old-releases/"; you'll find a lot
of sounnds package realises, can i use one of  them
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Re: [asterisk-users] asterisk-sounds doesn't exist in the sources, how can i get it?

[younss azzayani]

i found someone that is called "asterisk-sounds-main"
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[asterisk-users] question about regex

[Rilawich Ango]

Hi,  I have tried the regex function below with MACRO_EXTEN=5000*.
However, both of them return 0 instead 1 to me.  How can I search the
character in the end of line?

${REGEX("[*]$" ${MACRO_EXTEN})
returns 0

${REGEX("*$" ${MACRO_EXTEN})
returns 0 with error

ango
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Re: [asterisk-users] asterisk-sounds doesn't exist in the sources, how can i get it?

[Tzafrir Cohen]

On Tue, Feb 13, 2007 at 09:29:26AM +, younss azzayani wrote:
> Hello,
> I can't find asterisk-sounds in the svn.digium server, i ve been got 
> asterisk,
> zaptel,libpri,asterisk-addons (1.2 stable version)
> Thank You

apt-cache search -n asterisk-sounds

It should be called 'asterisk-sounds-extra'

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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[asterisk-users] asterisk-sounds doesn't exist in the sources, how can i get it?

[younss azzayani]

Hello,
I can't find asterisk-sounds in the svn.digium server, i ve been got asterisk,
zaptel,libpri,asterisk-addons (1.2 stable version)
Thank You

Younss AZ
KASTERISK.COM
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Re: [asterisk-users] Problems Asterisk with Digium TDM400 card => he don't see the disconnect

[younss azzayani]

can you show us you zaptel.conf & zapta?

Younss AZ
KASTERISK.COM
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[asterisk-users] Local channel characteristics

[Yuan LIU]

I'm reading about Local channel and applications.  One fancy idea came up: 
if my generic dial plan uses Dial() with no timeout, can I assign it a 
timeout for special purposes by

 Dial(Local/[EMAIL PROTECTED],,20)
or even add other Dial() options.

Well, I can't. (Maybe a feature request?)  So this pseudo channel seems to 
lack some characteristics of a "real" one.  Now the question is: what 
characteristics Local does have?  I can imagine that many Dial() options 
won't be effective, either.  Where are these stuff documented?


Yuan Liu


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