date:20050724

Difficult to believe that someone worked on any remotely technical
project when he/she relies on AOL for email.  Just my 2c.  Never thought
anything could be worse than me using Outlook ;-)

> -Original Message-
> From: Terry H. Gilsenan [mailto:[EMAIL PROTECTED] 
> Sent: Sunday, July 24, 2005 11:01 PM
> Subject: RE: [Asterisk-Users] Did anyone else get spammed by GIZMO?
> 
> 
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of 
> > [EMAIL PROTECTED]
> > Sent: Monday, 25 July 2005 1:42 PM
> > Subject: Re: [Asterisk-Users] Did anyone else get spammed by GIZMO?
> > 
> > I simply answered a question dealing with a project that I my
> > self worked on for a year. I am not trying to profit. Only assist.
> > On this list, we are all in this together!
> >  
> > Brad
> > 
> 
> Would it be possible to leave at least part of the post you 
> are responding to in your response?
> 
> Otherwise the context and attribution, and indeed who you are 
> replying to and what you are talking about is completely lost
> 
> My .02
> 
> T

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Re: [Asterisk-Users] Zap PRI load testing

2005-07-24 Thread Niklas Larsson

On Sun, 24 Jul 2005 17:07:52 +0100, Julian Lyndon-Smith wrote:

> I was wanting to stress-test my new server, and as I have a TE410p
> card (but only using 2 ports), I was going to connect ports 3 & 4
> with a cross-over cable so that I could make a number of outbound
> calls on port 3 and receive them as inbound on port 4.
>
> Before I start on this, does anyone have something similar that
> they would be willing to share ? If not, we will put our scripts
> onto the wiki when they are done.

I have a simple script that make a number of calls:

www.gransbygdenbutik.com/call

It takes 3 args: channel, number of calls, and extension. To this i created an 
extension like this:

exten => 3033,1,Answer()
;exten => 3033,2,Playback(demo-congrats)
exten => 3033,2,Wait(60)
exten => 3033,3,Hangup()

And made some calls to the swedish number for time information (that's default 
values).

exten => 3034,1,Answer()
exten => 3034,2,Playback(demo-congrats)
;exten => 3034,2,Wait(60)
exten => 3034,3,Hangup()

call zap/g1/3034, 10, 3033

and you call between the spans. Depending on your context etc.

Have fun...

mvh,

Niklas Larsson

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RE: [Asterisk-Users] Asterisk users from Turkey?

2005-07-24 Thread Erdem HAKİ

Yes, i have some but what kind of experience?

Erdem HAKI

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ceyhun
KIRMIZITAS
Sent: Saturday, July 23, 2005 12:47 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asterisk users from Turkey?

Is there any1 who has some experience with Asterisk in Turkey?
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RE: [Asterisk-Users] Caller ID, Called ID and Forwarded ID

2005-07-24 Thread Terry H. Gilsenan

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Edwin Groothuis
> Sent: Monday, 25 July 2005 3:04 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Caller ID, Called ID and Forwarded ID
> 
> Last month I saw something funny which I can't reproduce anymore:
> 
> A 0500 number in .au is a service phone number and are 
> forwarded on exchange level to a real phonenumber. So if A 
> calls B it gets forwarded to C. Very simple.
> 
> Now the funny thing, on the phone of C, I saw both A and B as 
> the "caller id". I've been asking around and trying to get it 
> again with a private 0500 number, but haven't been able to 
> recreate the behaviour. It could be a special Q.931 option, 
> it could be a random fluke of the system, it just could be 
> standard practise.
> 
> Has anybody ever seen this behaviour or know more of it?
> 

I have seen stuff like this over the last 2-3 months.

I have my .au mobile forwarded to my .pg mobile when I travel to .pg, lately
I have seen the actuall caller number (about 50% of the time) on my .pg cell
phone display.

It used to be that I would see my .au cell phone number as the number that
was calling.

As I say, this happens about 50% of the time, and has only happened in the
last 2 to 3 months.

Seems like a Telstra thing with exchange level call forwarding?

Regrds,
T

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Re: [Asterisk-Users] Polycom 600 Ring-Answer (but not ring!)




No ring is at the phone level,
only from what I know
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[Asterisk-Users] Caller ID, Called ID and Forwarded ID

2005-07-24 Thread Edwin Groothuis

Last month I saw something funny which I can't reproduce anymore:

A 0500 number in .au is a service phone number and are forwarded
on exchange level to a real phonenumber. So if A calls B it gets
forwarded to C. Very simple.

Now the funny thing, on the phone of C, I saw both A and B as the
"caller id". I've been asking around and trying to get it again
with a private 0500 number, but haven't been able to recreate the
behaviour. It could be a special Q.931 option, it could be a random
fluke of the system, it just could be standard practise.

Has anybody ever seen this behaviour or know more of it?

Edwin

-- 
Edwin Groothuis  |Personal website: http://www.mavetju.org
[EMAIL PROTECTED]|  Weblog: http://weblog.barnet.com.au/edwin/
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Re: [Asterisk-Users] Polycom 600 Ring-Answer (but not ring!)


Kristian Kielhofner wrote:


Billy Dunn wrote:

I have a bunch of Polycom Soundpoint 600 phones and they are working 
great.  The only thing I can't seem to get them to do is to 
ring-answer without the ring.


This is what I have in my sip.cfg file on the boot server:

voIpProt.SIP.alertInfo.2.class="4"/>
voIpProt.SIP.alertInfo.3.class="5"/>


se.rt.4.timeout="2000" se.rt.

4.ringer="2" se.rt.4.callWait="6" se.rt.4.mod="1"/>
se.rt.5.timeout="1" se.rt.5.

ringer="1" se.rt.5.callWait="6" se.rt.5.mod="1"/>

This is what I have in my extensions.conf file:

; bdunn's Office Extension AUTO ANSWER - WITH RING
   exten => 83004,1,SetVar(ALERT_INFO="RA")
   exten => 83004,2,Macro(stdexten,3004,${BDUNNOFFICE})
; bdunn's Office Extension AUTO ANSWER - NO RING
   exten => 93004,1,SetVar(ALERT_INFO="RANR")
   exten => 93004,2,Macro(stdexten,3004,${BDUNNOFFICE})

This does most of what I need - 93004 answers to speakerphone 
automatically, but there is a ring (a very short ring).  Dialing 
83004 gives a moderate length ring and answers as expected.  I'd 
really like to do it without the ring if possible.


If there are any Polycom pros out there, I could use some help.  I 
have already checked out this: 
http://www.voip-info.org/tiki-index.php?page=Polycom%20auto-answer%20config#comments 



Thanks.



My Polycom configs at:

 http://www.krisk.org/asterisk/pcom/

will do what you are looking for.  Set your ALERT_INFO variable to 
equal "AA" (Auto Answer) and the phone will go right to speaker phone 
mode, no ring.


Oh yeah... I've been ripping off your work a lot.  It's been a big help 
to me.  Thanks.  I think the next message clears up where I went wrong.



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Re: [Asterisk-Users] Polycom 600 Ring-Answer (but not ring!)


dbruce wrote:


If you use the polycom provided config files, the default ring_answer class
is 4 and the auto_answer class is 3. So for your RANR alertinfo entry,
change the class to 3 and it will work as you expect. ie:

   


The correct ringtype entry should be:
   

Notice there is no timeout, ringer or callwait entry.

Regards,
Derek
 

I think that's where I went wrong.  I didn't see anything like that in 
the admin guide, but I'll look again.  Thanks very much!

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Re: [Asterisk-Users] Polycom 600 Ring-Answer (but not ring!)





I'll give you an example.  My staff does a lot of interviews with new
hires and salesmen.  I do not personally sit in on the whole thing
(they can really drag out!), but I would like to be able to listen in
to make notes as needed while doing other work.

I had another idea whereas I could dial a certain extension which would
connect me to all available Polycoms and allow me to announce
something.  The ring is less important here.

Mostly I'm at the point where I think it should be working, but it
isn't, and it's driving me a little nuts.  :-)

Thanks!


[EMAIL PROTECTED] wrote:

  
  
  
  Give me an idea of your application.
  I personally created a really cool asterisk system for the US
Army where the phones would ring on silent for about an hour before
people would pick up for a conference call.
   
  I Know this config like the back of my head.
   
  Keep in mind, Asterisk has many really cool back doors!
   
  Tell me your application
   
  Brad
  
  

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RE: [Asterisk-Users] super high bandwidth codec

2005-07-24 Thread Rusty Shackleford

Title: Message




  
  -Original Message-From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of 
  [EMAIL PROTECTED]Sent: Sunday, July 24, 2005 9:11 
  PMTo: asterisk-users@lists.digium.comSubject: Re: 
  [Asterisk-Users] super high bandwidth codec
  It has nothing to do with bandwidth.
  It has everything to do with your routing gear! 
   
This is completely incorrect. Skype 
uses a codec that uses far more bandwidth than traditional telephony provides, 
which is why it's audio can have more range than even the best quality 
phone call. In theory, there is nothing preventing an all VOIP network from 
using such a codec, but as a practical matter, at least part of most phone calls 
are via traditional phone gear and/or networks, you don't see it widely 
deployed. 


--
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.338 / Virus Database: 267.9.4/57 - Release Date: 07/22/2005
 
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RE: [Asterisk-Users] super high bandwidth codec

2005-07-24 Thread Dean Collins









How do you figure?

 

How does skype sounds so damm good on the
same network/machine? I think you might be wrong.

 

Cheers,

Dean

 

 











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Monday, 25 July 2005 12:11
AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users]
super high bandwidth codec



 



It has nothing to do with bandwidth.





It has everything to do with your routing
gear!










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Re: [Asterisk-Users] Did anyone else get spammed by GIZMO?

2005-07-24 Thread Tzafrir Cohen

On Sun, Jul 24, 2005 at 11:45:07PM -0400, [EMAIL PROTECTED] wrote:
> you must be using Windows.

Maybe he does. Maybe you do. I don't. Your message still lack any
reference to the ones you respond to.

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend
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RE: [Asterisk-Users] DID + 800 Providers

2005-07-24 Thread Terry H. Gilsenan

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Nathan E. Pralle
> Sent: Monday, 25 July 2005 2:14 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] DID + 800 Providers
> 
> On Sunday 24 July 2005 22:36, Jay Milk scribbled:
> > Lousy pricing.  If you reply/advertise on-list, at least be 
> competitive.
> 
> Lousy attitude.  If you reply on-list, at least be courteous.
> 
> As far as pricing, listed prices are not always what we might 
> quote for quantity.
> 
> But this isn't an advertising forum so I won't go into 
> details any further.  I was simply replying to the 
> gentleman's question on-list for anyone else's 
> benefit.   My apologizes if it provoked someone's ire.

Nah, the info was clearly solicited by a list member. Its kewl.

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Re: [Asterisk-Users] Unlimited land line calls in Australia.




I personally have performed 'real time test" of plugging a Cisco 7940 phone 
behind a Netgear router and had is sound like "butt-crack".
and then plugged it in (within 2 min) behind a Cisco router, and it was 
Crystal Clear!
 
ROUTING GEAR IS ALWAYS THE KEY TO VOICE QUALITY ON VOIP.
 
That and a phone that has an embedded DSP chip vs. software driven.
 
Hence
 
GrandStream sucks
and Polycom and Cisco kick butt behind the proper routing gear!
 
Brad
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Re: [Asterisk-Users] DID + 800 Providers

2005-07-24 Thread Nathan E. Pralle

On Sunday 24 July 2005 22:36, Jay Milk scribbled:
> Lousy pricing.  If you reply/advertise on-list, at least be competitive.

Lousy attitude.  If you reply on-list, at least be courteous.

As far as pricing, listed prices are not always what we might quote for 
quantity.

But this isn't an advertising forum so I won't go into details any further.  I 
was simply replying to the gentleman's question on-list for anyone else's 
benefit.   My apologizes if it provoked someone's ire.

Nathan

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Re: [Asterisk-Users] super high bandwidth codec




It has nothing to do with bandwidth.
It has everything to do with your routing gear!
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[Asterisk-Users] asterisk with ser project, , , , here we go! ready or not!!!




Can anyone tell me if the acc_flag is only supported under the options 
request?
one of my telco pipes does not support the options field.
Yes, I know, non RFC.
 
Brad
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RE: [Asterisk-Users] Did anyone else get spammed by GIZMO?

2005-07-24 Thread Terry H. Gilsenan

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> [EMAIL PROTECTED]
> Sent: Monday, 25 July 2005 1:42 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] Did anyone else get spammed by GIZMO?
> 
> I simply answered a question dealing with a project that I my 
> self worked on for a year. I am not trying to profit. Only assist.
> On this list, we are all in this together!
>  
> Brad
> 

Would it be possible to leave at least part of the post you are responding
to in your response?

Otherwise the context and attribution, and indeed who you are replying to
and what you are talking about is completely lost

My .02

T

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[Asterisk-Users] Unlimited land line calls in Australia.

2005-07-24 Thread Dean Collins









Apart from Packet 8, is there anyone else who offers
unlimited calls in both the USA
and Australia?
I’m starting to get the shits with their call quality and looking at
alternatives.

 

I currently use the Packet 8 Asia unlimited service for $49
a month and it’s a great price point but starting to wear thin.

 

TIA,

Dean

 






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[Asterisk-Users] super high bandwidth codec

2005-07-24 Thread Dean Collins









I’ve just gotten off a skype conference call and it
pisses me off that the quality of skype is higher than my asterisk calls. 

 

Is there such a thing as a super high bandwidth codec?

 

In a situation that you have the bandwidth to share is there
something that I can use for important calls when the situation warrants it?

 

 

 

TIA,

Dean






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Re: [Asterisk-Users] Did anyone else get spammed by GIZMO?




you must be using Windows.
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Re: [Asterisk-Users] Did anyone else get spammed by GIZMO?




I simply answered a question dealing with a project that I my self worked 
on for a year. I am not trying to profit. Only assist.
On this list, we are all in this together!
 
Brad
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Re: [Asterisk-Users] DID + 800 Providers




unless you are willing to drop $650,000.00 for DID distribution 
segment.
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Re: [Asterisk-Users] DID + 800 Providers




PS, I never gave a price,,, just reality!
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Re: [Asterisk-Users] DID + 800 Providers




Dude, just delivering you reality. DIDs are like an ISP IP address, has to 
be hosted from carrier.
Just the way life is
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RE: [Asterisk-Users] Did anyone else get spammed by GIZMO?

Only softphone I ever downloaded (and promptly deleted) was xten.  And
no, I never "register" with this email address.

> -Original Message-
> From: C F [mailto:[EMAIL PROTECTED] 
> Sent: Sunday, July 24, 2005 5:13 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Did anyone else get spammed by GIZMO?
> 
> 
> Interesting, I've been on this list for almost a year now, 
> and I didn't recieve this spam. Are you sure you didn't 
> download any sip softphones and gave your email address? In 
> which case it is NOT spam, you gave your email address to them.
> 
> 
> On 7/22/05, Jay Milk <[EMAIL PROTECTED]> wrote:
> > > -Original Message-
> > > From: Jerry Glomph Black [mailto:[EMAIL PROTECTED]
> > > Sent: Friday, July 22, 2005 2:18 AM
> > > Subject: Re: [Asterisk-Users] Did anyone else get spammed 
> by GIZMO?
> > >
> > > It's not a spam.  They are not yokels.  Don't know about you
> > >
> > > Gizmo is basically a different front end offered by the 
> Sipphone.com 
> > > people, to offer an alternative to Skype which is not a 
> closed jail 
> > > (interoperates with all SIP devices, asterisk, etc.).
> > >
> > > I think they sent the mail to all registered sipphone.com users.
> > 
> > Dear Glomph,
> > 
> > Thanks, I know what the Gizmo project is.  I'm just not interested.
> > 
> > And yes, it's spam:
> > 1. It's unsolicited -- I'm not registered on Sipphone.com.  
> It came in 
> > through an email address I use exclusively for the 
> asterisk-users list 
> > and a small volume of personal mail. 2. It's commercial -- 
> #5 of "10 
> > more things to do" reads "Add $10 in CallOut credit to your account 
> > and talk for hours".  So, they're trying to sell me something.
> > 
> > Yokel: "A clumsy, unsophisticated person".  I think that would be 
> > their preferred label... Why?  Because if you send unsolicited 
> > commercial email in the US without offering a way to opt 
> out of future 
> > mailings, you're either clumsy and unsophisticated, or you're a 
> > criminal.  I gave them the benefit of the doubt and called them 
> > "yokels".  If that term offends you in any way, then let's use the 
> > more factual "criminals."
> > 
> > ___
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> UNSUBSCRIBE or update options visit:
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RE: [Asterisk-Users] Did anyone else get spammed by GIZMO?

Correct -- and, solicited or not, any commercial email message (that is,
any message soliciting business), must include unsubscribe instructions,
which the one in question did not.

> -Original Message-
> From: Steve Totaro [mailto:[EMAIL PROTECTED] 
> Sent: Sunday, July 24, 2005 8:45 PM
> Subject: Re: [Asterisk-Users] Did anyone else get spammed by GIZMO?
> 
> 
> Just because you give your email address does not allow for 
> unsolicited emails unless you agree in the EULA or terms and 
> conditions.
> 
> 
> - Original Message - 
> From: "C F" <[EMAIL PROTECTED]>
> Sent: Sunday, July 24, 2005 3:13 PM
> Subject: Re: [Asterisk-Users] Did anyone else get spammed by GIZMO?
> 
> 
> Interesting, I've been on this list for almost a year now, 
> and I didn't recieve this spam. Are you sure you didn't 
> download any sip softphones and gave your email address? In 
> which case it is NOT spam, you gave your email address to them.

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RE: [Asterisk-Users] DID + 800 Providers

Lousy pricing.  If you reply/advertise on-list, at least be competitive.

> -Original Message-
> From: Nathan E. Pralle [mailto:[EMAIL PROTECTED] 
> Sent: Sunday, July 24, 2005 10:25 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] DID + 800 Providers
> 
> 
> 
> > > So i'm looking for a good provider for DIDs and 800# from 
> the US and 
> > > CA, who offer online signup and ordering. The 
> provisioning should be 
> > > less than 12 hours, preferably instantly.
> 
> BinFone Telecom has instant provisioning and DIDs/800s for New York, 
> Washington, D.C., Baltimore, Indianapolis, St. Louis, Kansas 
> City, and 
> Phoenix.
> 
http://www.b0i0n0f0o0n0e.com

I am VP of Technology for it so if you have questions you may contact me

off-list.

Nathan

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Re: [Asterisk-Users] Polycom 600 Ring-Answer (but not ring!)




Give me an idea of your application.
I personally created a really cool asterisk system for the US Army where 
the phones would ring on silent for about an hour before people would pick up 
for a conference call.
 
I Know this config like the back of my head.
 
Keep in mind, Asterisk has many really cool back doors!
 
Tell me your application
 
Brad
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[Asterisk-Users] asterisk with ser project, , , , here we go! ready or not!!!




I every one, looking for suggestions, or even better yet, a how to 
guide.
I have read most of the wiki's on both, so I know this is completely 
possible
I have Asterisk and SER configured on the same server x6
yes, 6 servers.
If I can get this to work on one, we are golden!
 
Number one priority, re-invite = yes
number 2 priority = collection of CDR
number 3 priority = Asterisk functionality upon request.
 
Ok, I am sure you have all guessed where I am going with this.
 
Here is how it is supposed to work.
 
Ser replicates authentication as a proxy to all servers, to both Asterisk 
and SER. Harmonic authentication... ( I find many cases of people getting this 
to work.
 
A traditional outbound call to an Internet based (no Voice T-1 here), hits 
ser which redirects and collects the cdr and stateless.
All MySql databases are nicely mirrored right now..
 
end user request voice mail and/or conference. Ser forwards request to 
Asterisk. Asterisk ACKs and does it's job.
 
I am not a leech, I am and open participant in the mailing list 
already.
 
I am using a number of different systems, mainly FC4 for x64. I have a few 
Centos/Digium systems and the occasional Red Hat. It is up to me to solve the 
permissions stuff.
 
Suggestions?
Pointers?
 
Brad
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Re: [Asterisk-Users] DID + 800 Providers

2005-07-24 Thread Nathan E. Pralle


> > So i'm looking for a good provider for DIDs and 800# from the US and CA,
> > who offer online signup and ordering. The provisioning should be less
> > than 12 hours, preferably instantly.

BinFone Telecom has instant provisioning and DIDs/800s for New York, 
Washington, D.C., Baltimore, Indianapolis, St. Louis, Kansas City, and 
Phoenix.

http://www.binfone.com

I am VP of Technology for it so if you have questions you may contact me 
off-list.

Nathan
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Re: [Asterisk-Users] Polycom 600 Ring-Answer (but not ring!)

If you use the polycom provided config files, the default ring_answer class
is 4 and the auto_answer class is 3. So for your RANR alertinfo entry,
change the class to 3 and it will work as you expect. ie:




The correct ringtype entry should be:


Notice there is no timeout, ringer or callwait entry.

Regards,
Derek

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

Sent: Sunday, July 24, 2005 7:47 PM
Subject: [Asterisk-Users] Polycom 600 Ring-Answer (but not ring!)


> I have a bunch of Polycom Soundpoint 600 phones and they are working
> great.  The only thing I can't seem to get them to do is to ring-answer
> without the ring.
>
> This is what I have in my sip.cfg file on the boot server:
>
>  voIpProt.SIP.alertInfo.2.class="4"/>
>  voIpProt.SIP.alertInfo.3.class="5"/>
>
>  se.rt.4.timeout="2000" se.rt.
> 4.ringer="2" se.rt.4.callWait="6" se.rt.4.mod="1"/>
>  se.rt.5.timeout="1" se.rt.5.
> ringer="1" se.rt.5.callWait="6" se.rt.5.mod="1"/>
>
> This is what I have in my extensions.conf file:
>
> ; bdunn's Office Extension AUTO ANSWER - WITH RING
> exten => 83004,1,SetVar(ALERT_INFO="RA")
> exten => 83004,2,Macro(stdexten,3004,${BDUNNOFFICE})
> ; bdunn's Office Extension AUTO ANSWER - NO RING
> exten => 93004,1,SetVar(ALERT_INFO="RANR")
> exten => 93004,2,Macro(stdexten,3004,${BDUNNOFFICE})
>
> This does most of what I need - 93004 answers to speakerphone
> automatically, but there is a ring (a very short ring).  Dialing 83004
> gives a moderate length ring and answers as expected.  I'd really like
> to do it without the ring if possible.
>
> If there are any Polycom pros out there, I could use some help.  I have
> already checked out this:
>
http://www.voip-info.org/tiki-index.php?page=Polycom%20auto-answer%20config#
comments
>
> Thanks.
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Re: [Asterisk-Users] Polycom 600 Ring-Answer (but not ring!)




turn off the ringer. Put it in silent mode!
Brad
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Re: [Asterisk-Users] web managment

2005-07-24 Thread Time Bandit

> what is the best web based managment aplication for asterisk ???
This topic as been discussed many times. Search the archives : 
http://www.google.ca/search?hl=en&q=site%3Alists.digium.com+web+management
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[Asterisk-Users] HFC-S cards in Australia

2005-07-24 Thread Stephen Allan (External account)

Is anyone aware of a ISDN HFC-S card available in Australia that is
A-tick certified?

Thanks,
Stephen Allan.
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Re: [Asterisk-Users] DID + 800 Providers

2005-07-24 Thread Rich Adamson

> I'm looking for US DID and US50/CA 800# Providers.
> 
> I found voiceconduits.com 8 month ago, there interface looks good, but 
> there are still not live, I believe they won't be any time soon.
> 
> I found sixtel, but order take eternities, they probably won't get my 
> orders right any soon.
> 
> So i'm looking for a good provider for DIDs and 800# from the US and CA, 
> who offer online signup and ordering. The provisioning should be less 
> than 12 hours, preferably instantly.
> 
> If anybody knows or even uses such a provider, please leave me a note.

Take a look at www.teliax.com for DIDs, and NuFone for 800 numbers.

At least that way you have service split over two itsp's in case one
goes down (or bankrupt ;).


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Re: [Asterisk-Users] Polycom 600 Ring-Answer (but not ring!)

2005-07-24 Thread Kristian Kielhofner


Billy Dunn wrote:
I have a bunch of Polycom Soundpoint 600 phones and they are working 
great.  The only thing I can't seem to get them to do is to ring-answer 
without the ring.


This is what I have in my sip.cfg file on the boot server:

voIpProt.SIP.alertInfo.2.class="4"/>
voIpProt.SIP.alertInfo.3.class="5"/>


se.rt.4.timeout="2000" se.rt.

4.ringer="2" se.rt.4.callWait="6" se.rt.4.mod="1"/>
se.rt.5.timeout="1" se.rt.5.

ringer="1" se.rt.5.callWait="6" se.rt.5.mod="1"/>

This is what I have in my extensions.conf file:

; bdunn's Office Extension AUTO ANSWER - WITH RING
   exten => 83004,1,SetVar(ALERT_INFO="RA")
   exten => 83004,2,Macro(stdexten,3004,${BDUNNOFFICE})
; bdunn's Office Extension AUTO ANSWER - NO RING
   exten => 93004,1,SetVar(ALERT_INFO="RANR")
   exten => 93004,2,Macro(stdexten,3004,${BDUNNOFFICE})

This does most of what I need - 93004 answers to speakerphone 
automatically, but there is a ring (a very short ring).  Dialing 83004 
gives a moderate length ring and answers as expected.  I'd really like 
to do it without the ring if possible.


If there are any Polycom pros out there, I could use some help.  I have 
already checked out this: 
http://www.voip-info.org/tiki-index.php?page=Polycom%20auto-answer%20config#comments 



Thanks.


My Polycom configs at:

 http://www.krisk.org/asterisk/pcom/

will do what you are looking for.  Set your ALERT_INFO variable to equal 
"AA" (Auto Answer) and the phone will go right to speaker phone mode, no 
ring.


--
Kristian Kielhofner
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Re: [Asterisk-Users] Incoming call prob

2005-07-24 Thread Rich Adamson


> I am having a problem with your my nufone service. 
> I'm trying to setup incoming calls and I'm having no
> success. Outgoing works fine though.  The message I'm
> getting is "the person you are call is not currently
> reachable".  I'm going to give you as much info as I
> can.  I'm also an asterisk newb! Anyways, I installed
> [EMAIL PROTECTED]  Set up extensions which communicate to
> each other fine.  Everything is default except here is
> my iax.conf:
> 
> [general]
>
 
>
> register => meadmaker:@switch-2.nufone.net
> 
> [NuFone]
> type=peer
> host=switch-1.nufone.net
> secret=
> 
> and here is what I added to extension.conf:
> 
> ;added for nufone
> exten => _1NXXNXX,1,SetCallerID(8662521540)
> exten => _1NXXNXX,2,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}
> 
> 
> [inbound]
> exten => 8662521540,1,Answer()
> exten => 8662521540,2,Wait(2)
> exten => 8662521540,3,Dial(200)
> 
> I would appreciate some help.

Try this approach instead:

[nufoneout] ; for outbound calls via NuFone.net
type=peer  ; used for outgoing calls
host=switch-1.nufone.net
username=meadmaker
secret=mysecret
disallow=all
allow=gsm

[NuFone] ; for incoming calls from NuFone
type=user  ; used for Incoming calls
secret=mysecret
context=nufone888
disallow=all
allow=gsm

Note in the above that type=peer is a context for outgoing calls,
and type=user is for incoming calls. (You only had a context for
outgoing calls in your posted example. The above are working
examples from my system.)

Also, there are two basic forms of the register statement:
 register => meadmaker:@switch-2.nufone.net
or
 register => meadmaker:@switch-2.nufone.net/8662521540

The first form tells the Nufone equipment that you are ready to
accept calls, however their equipment will not send you any digits.
You will have to use an exten => s approach in extensions.conf.

The second form does the same thing, but also tells the Nufone
equipment to send 8662521540 to you for each call. Then, your
extensions.conf entry can use the exten => 8662521540 approach.

Also, your really don't want to "answer" the call first and then
wait. Just do:
 exten => 8662521540,3,Dial(200)

When "you" answer the incoming call at extension 200, your asterisk
system will "then" automatically send an answer to Nufone.

Using my example above, you can send calls to Nufone by doing:
 exten => _1NXXNXX,1,SetCallerID(8662521540)
 exten => _1NXXNXX,2,Dial,IAX2/nufoneout/${EXTEN}

Note that I've referenced the [nufoneout] context, where asterisk
picks up the host, username, secret, etc, from that context. No
need to bury your username, etc, within an exten => string.

Finally, depending upon whether you are using Stable or Head, the
SetCallerID keyword is in the process of changing to 
 Set(CallerIDnum=8662521540|a). Head already supports this; not
sure whether Stable has it or not right now. Be aware.

Rich


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RE: [Asterisk-Users] RE: Business Edition

2005-07-24 Thread Kevin Walsh

Brian West [EMAIL PROTECTED] wrote:
> Or better yet.. modify the disclaimer like I and a few others did to
> say that the only thing you will disclaim are things you post on the
> bug tracker!  NO UPDATES, NO CHANGES, NO NOTHING!  If its not posted
> under your user on mantis IT IS NOT DISCLAIMED!
> 
That would seem to be a reasonable suggestion.

-- 
   _/   _/  _/_/_/_/  _/_/  _/_/_/  _/_/
  _/_/_/   _/_/  _/_/_/_/_/  _/   K e v i n   W a l s h
 _/ _/_/  _/ _/ _/_/  _/_/[EMAIL PROTECTED]
_/   _/  _/_/_/_/  _/_/_/_/  _/_/

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Re: [Asterisk-Users] Busy Lamp Field SIP Phone

2005-07-24 Thread Doug Lytle


Thomas Christie wrote:


(no offense, your email is already deleted).

 


None taken.


Have you used this one or others?

What's your opinion of this gentleman using this application instead of a
hardware solution?
 

 



In the process of setting it up for a August 6th install. I won't know 
how well the employees take to it until it's been in production for a 
bit. I'm impressed with it so far.


I haven't found a hardware solution to date, so I can't comment on that 
portion of it.


Doug

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Re: [Asterisk-Users] why this macro works Only with a * StableVersion? ; -(

2005-07-24 Thread Kevin P. Fleming


xAD wrote:

Yes your are right, this is the modified version for CVS HEAD ..but don't
works  ;-D you have tried it?


No, because you haven't bothered to tell us what is wrong. "This doesn't 
work" is not much to go on, I'm sure nobody is going to try your macro 
without you first describing what it is supposed to do and what is going 
wrong.

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Re: [Asterisk-Users] [EMAIL PROTECTED]

Your BVRINGS variable is being evaluated to a null string... check your
spelling in the macro call.

Regards,
Derek

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

Sent: Sunday, July 24, 2005 6:00 PM
Subject: RE: [Asterisk-Users]
[EMAIL PROTECTED]


>
>  OK, I removed that line as you said below, and now when I call in on the
BV
> line, I see this as output:
>
>
> Jul 24 19:55:41 DEBUG[1078]: Setting NAT on RTP to 0
> Jul 24 19:55:41 DEBUG[1078]: Check for res for 240524
> Jul 24 19:55:41 DEBUG[1078]: 240524 is not a local user
> Jul 24 19:55:41 DEBUG[1078]: build_route: Contact hop:
> 
> Jul 24 19:55:41 VERBOSE[1078]: -- Executing
Macro("SIP/240524-7457",
> "exten-vm|@default|") in new stack
> Jul 24 19:55:41 VERBOSE[1078]: -- Executing
SetVar("SIP/240524-7457",
> "FROMCONTEXT=exten-vm") in new stack
> Jul 24 19:55:41 VERBOSE[1078]: -- Executing
Macro("SIP/240524-7457",
> "record-enable||IN") in new stack
> Jul 24 19:55:41 VERBOSE[1078]: -- Executing
GotoIf("SIP/240524-7457",
> "0 > 0?2:4") in new stack
> Jul 24 19:55:41 VERBOSE[1078]: -- Goto (macro-record-enable,s,4)
> Jul 24 19:55:41 DEBUG[1078]: Expression is '0'
> Jul 24 19:55:41 VERBOSE[1078]: -- Executing
GotoIf("SIP/240524-7457",
> "0?5:8") in new stack
> Jul 24 19:55:41 VERBOSE[1078]: -- Goto (macro-record-enable,s,8)
> Jul 24 19:55:41 DEBUG[1078]: Expression is '0'
> Jul 24 19:55:41 VERBOSE[1078]: -- Executing
GotoIf("SIP/240524-7457",
> "0?9:12") in new stack
> Jul 24 19:55:41 VERBOSE[1078]: -- Goto (macro-record-enable,s,12)
> Jul 24 19:55:41 VERBOSE[1078]: -- Executing
DBget("SIP/240524-7457",
> "RecEnable=RECORD-IN/") in new stack
> Jul 24 19:55:41 VERBOSE[1078]: -- DBget: varname=RecEnable,
> family=RECORD-IN, key=
> Jul 24 19:55:41 DEBUG[1078]: Unable to find key '' in family 'RECORD-IN'
> Jul 24 19:55:41 VERBOSE[1078]: -- DBget: Value not found in database.
> Jul 24 19:55:41 VERBOSE[1078]: -- Executing
SetVar("SIP/240524-7457",
> "CALLFILENAME=20050724-195541-1122249341.21") in new stack
> Jul 24 19:55:41 DEBUG[1078]: Expression is '0'
> Jul 24 19:55:41 VERBOSE[1078]: -- Executing
GotoIf("SIP/240524-7457",
> "0?15:99") in new stack
> Jul 24 19:55:41 VERBOSE[1078]: -- Goto (macro-record-enable,s,99)
> Jul 24 19:55:41 VERBOSE[1078]: -- Executing
NoOp("SIP/240524-7457",
> "NO RECORDING NEEDED") in new stack
> Jul 24 19:55:41 VERBOSE[1078]: -- Executing
Macro("SIP/240524-7457",
> "dial|||") in new stack
> Jul 24 19:55:41 DEBUG[1078]: Expression is '0'
> Jul 24 19:55:41 VERBOSE[1078]: -- Executing
GotoIf("SIP/240524-7457",
> "0?4:2") in new stack
> Jul 24 19:55:41 VERBOSE[1078]: -- Goto (macro-dial,s,2)
> Jul 24 19:55:41 WARNING[1078]: ast_yyerror(): syntax error: syntax error;
> Input:
>  !=
>
> ^
> Jul 24 19:55:41 DEBUG[1078]: Expression is '0'
> Jul 24 19:55:41 VERBOSE[1078]: -- Executing
GotoIf("SIP/240524-7457",
> "0?4:3") in new stack
> Jul 24 19:55:41 VERBOSE[1078]: -- Goto (macro-dial,s,3)
> Jul 24 19:55:41 VERBOSE[1078]: -- Executing
> SetCIDName("SIP/240524-7457", "Fork MD") in new stack
> Jul 24 19:55:41 VERBOSE[1078]: -- Executing AGI("SIP/240524-7457",
> "dialparties.agi") in new stack
> Jul 24 19:55:41 VERBOSE[1078]: -- Launched AGI Script
> /var/lib/asterisk/agi-bin/dialparties.agi
> Jul 24 19:55:41 VERBOSE[1078]: --  dialparties.agi: request =
> dialparties.agi
> Jul 24 19:55:41 VERBOSE[1078]: --  dialparties.agi: priority = 4
> Jul 24 19:55:41 VERBOSE[1078]: --  dialparties.agi: extension = s
> Jul 24 19:55:41 VERBOSE[1078]: --  dialparties.agi: language = en
> Jul 24 19:55:41 VERBOSE[1078]: --  dialparties.agi: accountcode =
> Jul 24 19:55:41 VERBOSE[1078]: --  dialparties.agi: uniqueid =
> 1122249341.21
> Jul 24 19:55:41 VERBOSE[1078]: --  dialparties.agi: channel =
> SIP/240524-7457
> Jul 24 19:55:41 VERBOSE[1078]:   dialparties.agi: callerid = Fork
> Jul 24 19:55:41 VERBOSE[1078]: --  dialparties.agi: context =
macro-dial
> Jul 24 19:55:41 VERBOSE[1078]: --  dialparties.agi: type = SIP
> Jul 24 19:55:41 VERBOSE[1078]: --  dialparties.agi: rdnis = unknown
> Jul 24 19:55:41 VERBOSE[1078]: --

Re: [Asterisk-Users] Stupid hold music

2005-07-24 Thread Tom Tune

How come nobody has mentioned "The Girl From Ipanema" as performed by
Herb Alpert and Co.?
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[Asterisk-Users] Polycom 600 Ring-Answer (but not ring!)

I have a bunch of Polycom Soundpoint 600 phones and they are working 
great.  The only thing I can't seem to get them to do is to ring-answer 
without the ring.


This is what I have in my sip.cfg file on the boot server:

voIpProt.SIP.alertInfo.2.class="4"/>
voIpProt.SIP.alertInfo.3.class="5"/>


se.rt.4.timeout="2000" se.rt.

4.ringer="2" se.rt.4.callWait="6" se.rt.4.mod="1"/>
se.rt.5.timeout="1" se.rt.5.

ringer="1" se.rt.5.callWait="6" se.rt.5.mod="1"/>

This is what I have in my extensions.conf file:

; bdunn's Office Extension AUTO ANSWER - WITH RING
   exten => 83004,1,SetVar(ALERT_INFO="RA")
   exten => 83004,2,Macro(stdexten,3004,${BDUNNOFFICE})
; bdunn's Office Extension AUTO ANSWER - NO RING
   exten => 93004,1,SetVar(ALERT_INFO="RANR")
   exten => 93004,2,Macro(stdexten,3004,${BDUNNOFFICE})

This does most of what I need - 93004 answers to speakerphone 
automatically, but there is a ring (a very short ring).  Dialing 83004 
gives a moderate length ring and answers as expected.  I'd really like 
to do it without the ring if possible.


If there are any Polycom pros out there, I could use some help.  I have 
already checked out this: 
http://www.voip-info.org/tiki-index.php?page=Polycom%20auto-answer%20config#comments


Thanks.
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Re: [Asterisk-Users] why this macro works Only with a * StableVersion? ; -(

2005-07-24 Thread xAD

Yes your are right, this is the modified version for CVS HEAD ..but don't
works  ;-D you have tried it?

this version works only on STABLE:

[macro-tono_simulato]
exten => s,1,SetVar(DIALED=${ARG1})
exten => s,2,SetVar(TOCONTEXT=${ARG2})
exten => s,3,SetVar(STRIP=${ARG3})
exten => s,4,SetVar(TONES=425/600,0/1000,425/200,0/200 )
exten => s,5,Goto(tono_simulato_pausa|s|1)

[tono_simulato_pausa]
exten => s,1,DigitTimeout(0)
exten => s,2,Wait(0.5)
exten => s,3,Playtones(${TONES})
exten => s,4,WaitExten(3)

exten => _X,1,GotoIF(${DIALED}${EXTEN} > 0?100:2)
exten => _X,2,SetVar(DIALED=${DIALED}${EXTEN})
exten => _X,3,DigitTimeout(0)

exten => _X,100,SetVar(DIALED=${DIALED}${EXTEN})
exten => _X,101,StopPlaytones
exten => _X,102,DigitTimeout(3)

exten => _X.,1,Goto(${TOCONTEXT}|$[${DIALED:${STRIP}}]${EXTEN}|1)

exten => t,1,Hangup

you know why on CVS don't works?

thanks.

- Original Message - 
From: "Kevin P. Fleming" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Monday, July 25, 2005 12:16 AM
Subject: Re: [Asterisk-Users] why this macro works Only with a *
StableVersion? ; -(


> xAD wrote:
> > Hi all, i have a weird problem, i need to know why this 'simple' macro
works Only with a * STABLE version (1.0.0.x to 1.0.0.9) , if i try it with a
CVS HEAD version.. don't works ;-(
> >
> > Thanks in advance.
> >
> > [local]
> >
> > include => default
> >
> > [macro-tono_simulato]
> > exten => s,1,Set(DIALED=${ARG1})
> > exten => s,2,Set(TOCONTEXT=${ARG2})
> > exten => s,3,Set(STRIP=${ARG3})
> > exten => s,4,Set(TONES=dial)
> > exten => s,5,Goto(tono_simulato_pausa|s|1)
>
> This macro _cannot_ work in version 1.0.x, because Set() is not
> supported there (only SetVar()).
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[Asterisk-Users] Disconnecting a call on asterisk

2005-07-24 Thread peiyin

Dear all,

I want to create a php web front end to disconnect a SIP call (from a
particular sip phone) which is in progress. Any ideas how to do so?


Thanks in advance.
peiyin

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RE: [Asterisk-Users] Busy Lamp Field SIP Phone

2005-07-24 Thread Thomas Christie

I thought there would already be one or eight hundred of them out there
already.  I was joking about doing the work for ... Um ... What's-his-name
(no offense, your email is already deleted).

Have you used this one or others?

What's your opinion of this gentleman using this application instead of a
hardware solution?


Thomas Christie

There are 10 types of people in the world:  those who understand binary and
those who don't.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle
Sent: Sunday, July 24, 2005 21:21
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Busy Lamp Field SIP Phone

Thomas Christie wrote:

> This may not be the answer you're looking for ... but why not whip up 
> a little program or web application for the operator's PC that shows 
> the extension, name, busy status, new voice-mail count, etc?  Or you 
> could have it done by a consultant.  In the long run, it will probably 
> be less hassle, cheaper, and much more functional than what you're 
> considering.
>  
>
>  
>
Such an applicatioin already exists, it's called Flash Operator.

http://www.asternic.org

Doug

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Re: [Asterisk-Users] Busy Lamp Field SIP Phone

2005-07-24 Thread Doug Lytle


Thomas Christie wrote:

This may not be the answer you're looking for ... but why not whip up 
a little program or web application for the operator's PC that shows 
the extension, name, busy status, new voice-mail count, etc?  Or you 
could have it done by a consultant.  In the long run, it will probably 
be less hassle, cheaper, and much more functional than what you're 
considering.
 

 


Such an applicatioin already exists, it's called Flash Operator.

http://www.asternic.org

Doug

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Re: [Asterisk-Users] Need to ztcfg every time I reboot *

2005-07-24 Thread Andrew Latham

Add your module to your module startup method for your distro. See
http://voip-info.org and search for startup or boot, then read.



On 7/24/05, Angus Comber <[EMAIL PROTECTED]> wrote:
>  
> Hello 
>   
> I am sure this is a very basic Linux question. 
>   
> But every time I reboot my * I need to 
>   
> modprobe  
>   
> and then 
>   
> ztcfg 
>   
> After doing this I can then run * without it complaining about not loading a
> channel.  The module being loaded is qozap - a ISDN card. 
>   
> What do I need to do to make the ztcfg configuration persistent? 
>   
> Angus 
>   
> ___
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-- 

Andrew Latham - AKA: LATHAMA (lay-th-ham-eh)
WWW: http://lathama.com
Email: [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED]
If any of the above are down we have bigger problems than my email!

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RE: [Asterisk-Users] Busy Lamp Field SIP Phone

2005-07-24 Thread Thomas Christie




This may not be the answer you're looking for ... but why 
not whip up a little program or web application for the operator's PC that shows 
the extension, name, busy status, new voice-mail count, etc?  Or you could 
have it done by a consultant.  In the long run, it will probably be less 
hassle, cheaper, and much more functional than what you're 
considering.
 
And as luck would have it, I'm a consultant programmer 
...
 
(Why do I suddenly feel like Mr Haney from Green 
Acres?)
 
Thomas Christie
There are 10 types of people in the 
world:  those who understand binary and those who don't.
 


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Craig 
BruendermanSent: Sunday, July 24, 2005 19:31To: 
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Busy Lamp 
Field SIP Phone


Does anyone have a recommendation 
for a good SIP phone with a busy lamp field? I need my operator to be able to 
see extension status for about 20 extensions and transfer via HOLD + extension 
button. I’ve got a pair of SNOM 360s with the sidecar, but I’m very disappointed 
with them. The buttons are cheap and rubbery like a Sipura 841, the handset cord 
is short and cheap, the audio quality is not good, the speaker is bad, the 
button layout is too busy, and I’ve found it very hard to train my users how to 
use.
 
Has anyone else had the same 
experience with the 360s? I’m ready to replace them with anything. I put Polycom 
501’s in for everyone else and they are wonderful.
 
Does the Cisco 7960 + 7914 work well 
for such an application? Keep in mind it has to show free/busy status of 20 
although I guess we could get by with 14 if the 7960 + 7914 is a solid option. 
I’m in Louisville, 
KY if anyone knows of a distributor 
or retailer that I could demo a unit from.
 
Thanks
 
Craig 
Bruenderman
Network Advocates, 
Inc.9001 Shelbyville 
Road, 260 BurhansLouisville, KY  
40222Main:  502-412-1050Free:  
877-412-1050DID:   502-992-5929Fax:   
502-412-1058Mobile:  
502-548-1100
 
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Re: [Asterisk-Users] Busy Lamp Field SIP Phone

2005-07-24 Thread Joseph


dbruce wrote:
The Cisco 7960 SIP firmware does not support the 7914 "sidecar"... To 
use the "sidecar" you need to use the SCCP (Call Manager) Firmware. 
There is chan_skinny in Asterisk but I don't recall seing support in it 
for the 7914. You may want to look at the chan_sccp driver.. it is 
supposed to have support for the "sidecar".
 


It supports up to 2 side cars.

However the Busy Lamp option is only now being developed, however should 
be forth coming very soon. I have tested the speed dials without the BLF 
with 1 side car and it works fine.


You can test the chan_sccp with this:
http://www.ipblue.com/solutions_vtgopcadv.htm
It does emulate a cisco 7960+7914 (old firmware version)

--

respectfully, Joseph

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Re: [Asterisk-Users] ClueCon in 2 Weeks!

2005-07-24 Thread Tom Hayden

Mine had no problem sending me. I can't wait!

--
Tom

On 7/24/05, Brian West <[EMAIL PROTECTED]> wrote:
> I'll talk to your boss if he has a problem! ;)
> 
> /b
> 
> On Jul 23, 2005, at 11:03 PM, Terry Moore-Read wrote:
> 
> >
> > Mine did.
> >
> >
> >
>  [EMAIL PROTECTED] 7/21/2005 2:54 PM >>>
> 
> > Brian West wrote:
> >
> >
> >> ClueCon is coming in 2 weeks so we urge everyone who plans on
> >> attending to register today so we get a proper headcount!
> >>
> >> 
> >>
> >> Thanks,
> >> Brian West
> >> Asterlink.com
> >> 
> >>
> >
> > Anyone else think that was a joke at first impression? Good luck
> > convincing the boss to pay for your way to "ClueCon" ;-)
> > ___
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> >
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> >
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> > you have received this in error, please notify
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-- 
Tom
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RE: [Asterisk-Users] [EMAIL PROTECTED]


 OK, I removed that line as you said below, and now when I call in on the BV
line, I see this as output:


Jul 24 19:55:41 DEBUG[1078]: Setting NAT on RTP to 0
Jul 24 19:55:41 DEBUG[1078]: Check for res for 240524
Jul 24 19:55:41 DEBUG[1078]: 240524 is not a local user
Jul 24 19:55:41 DEBUG[1078]: build_route: Contact hop:

Jul 24 19:55:41 VERBOSE[1078]: -- Executing Macro("SIP/240524-7457",
"exten-vm|@default|") in new stack
Jul 24 19:55:41 VERBOSE[1078]: -- Executing SetVar("SIP/240524-7457",
"FROMCONTEXT=exten-vm") in new stack
Jul 24 19:55:41 VERBOSE[1078]: -- Executing Macro("SIP/240524-7457",
"record-enable||IN") in new stack
Jul 24 19:55:41 VERBOSE[1078]: -- Executing GotoIf("SIP/240524-7457",
"0 > 0?2:4") in new stack
Jul 24 19:55:41 VERBOSE[1078]: -- Goto (macro-record-enable,s,4)
Jul 24 19:55:41 DEBUG[1078]: Expression is '0'
Jul 24 19:55:41 VERBOSE[1078]: -- Executing GotoIf("SIP/240524-7457",
"0?5:8") in new stack
Jul 24 19:55:41 VERBOSE[1078]: -- Goto (macro-record-enable,s,8)
Jul 24 19:55:41 DEBUG[1078]: Expression is '0'
Jul 24 19:55:41 VERBOSE[1078]: -- Executing GotoIf("SIP/240524-7457",
"0?9:12") in new stack
Jul 24 19:55:41 VERBOSE[1078]: -- Goto (macro-record-enable,s,12)
Jul 24 19:55:41 VERBOSE[1078]: -- Executing DBget("SIP/240524-7457",
"RecEnable=RECORD-IN/") in new stack
Jul 24 19:55:41 VERBOSE[1078]: -- DBget: varname=RecEnable,
family=RECORD-IN, key=
Jul 24 19:55:41 DEBUG[1078]: Unable to find key '' in family 'RECORD-IN'
Jul 24 19:55:41 VERBOSE[1078]: -- DBget: Value not found in database.
Jul 24 19:55:41 VERBOSE[1078]: -- Executing SetVar("SIP/240524-7457",
"CALLFILENAME=20050724-195541-1122249341.21") in new stack
Jul 24 19:55:41 DEBUG[1078]: Expression is '0'
Jul 24 19:55:41 VERBOSE[1078]: -- Executing GotoIf("SIP/240524-7457",
"0?15:99") in new stack
Jul 24 19:55:41 VERBOSE[1078]: -- Goto (macro-record-enable,s,99)
Jul 24 19:55:41 VERBOSE[1078]: -- Executing NoOp("SIP/240524-7457",
"NO RECORDING NEEDED") in new stack
Jul 24 19:55:41 VERBOSE[1078]: -- Executing Macro("SIP/240524-7457",
"dial|||") in new stack
Jul 24 19:55:41 DEBUG[1078]: Expression is '0'
Jul 24 19:55:41 VERBOSE[1078]: -- Executing GotoIf("SIP/240524-7457",
"0?4:2") in new stack
Jul 24 19:55:41 VERBOSE[1078]: -- Goto (macro-dial,s,2)
Jul 24 19:55:41 WARNING[1078]: ast_yyerror(): syntax error: syntax error;
Input:
 != 

^
Jul 24 19:55:41 DEBUG[1078]: Expression is '0'
Jul 24 19:55:41 VERBOSE[1078]: -- Executing GotoIf("SIP/240524-7457",
"0?4:3") in new stack
Jul 24 19:55:41 VERBOSE[1078]: -- Goto (macro-dial,s,3)
Jul 24 19:55:41 VERBOSE[1078]: -- Executing
SetCIDName("SIP/240524-7457", "Fork MD") in new stack
Jul 24 19:55:41 VERBOSE[1078]: -- Executing AGI("SIP/240524-7457",
"dialparties.agi") in new stack
Jul 24 19:55:41 VERBOSE[1078]: -- Launched AGI Script
/var/lib/asterisk/agi-bin/dialparties.agi
Jul 24 19:55:41 VERBOSE[1078]: --  dialparties.agi: request =
dialparties.agi
Jul 24 19:55:41 VERBOSE[1078]: --  dialparties.agi: priority = 4
Jul 24 19:55:41 VERBOSE[1078]: --  dialparties.agi: extension = s
Jul 24 19:55:41 VERBOSE[1078]: --  dialparties.agi: language = en
Jul 24 19:55:41 VERBOSE[1078]: --  dialparties.agi: accountcode = 
Jul 24 19:55:41 VERBOSE[1078]: --  dialparties.agi: uniqueid =
1122249341.21
Jul 24 19:55:41 VERBOSE[1078]: --  dialparties.agi: channel =
SIP/240524-7457
Jul 24 19:55:41 VERBOSE[1078]:   dialparties.agi: callerid = Fork
Jul 24 19:55:41 VERBOSE[1078]: --  dialparties.agi: context = macro-dial
Jul 24 19:55:41 VERBOSE[1078]: --  dialparties.agi: type = SIP
Jul 24 19:55:41 VERBOSE[1078]: --  dialparties.agi: rdnis = unknown
Jul 24 19:55:41 VERBOSE[1078]: --  dialparties.agi: enhanced = 0.0
Jul 24 19:55:41 VERBOSE[1078]: --  dialparties.agi: dnid = 201
Jul 24 19:55:41 VERBOSE[1078]:   dialparties.agi: Caller ID name is 'Fork MD'
number is '410515'
Jul 24 19:55:41 DEBUG[1078]: Manager received command 'Login'
Jul 24 19:55:41 VERBOSE[1078]:   == Parsing '/etc/asterisk/manager.conf': Jul
24 19:55:41 VERBOSE[1078]:   == Parsing '/etc/asterisk/manager.conf': Found
Jul 24 19:55:41 VERBOSE[1078]:   == Parsing
'/etc/asterisk/manager_custom.conf': Jul 24 19:55:41 VERBOSE[1078]:   ==
Parsing '/etc/asterisk/manager_custom.conf': Found
Jul 24 19:55:41 DEBUG[1078]: 0.0.0.0/0.0.0.0/0.0.0.0 appended to acl for peer
Jul 24 19:55:41 DEBUG[1078

Re: [Asterisk-Users] Busy Lamp Field SIP Phone

The Cisco 7960 SIP firmware does not support the
7914 "sidecar"... To use the "sidecar" you need to use the SCCP (Call Manager)
Firmware. There is chan_skinny in Asterisk but I don't recall seing support in
it for the 7914. You may want to look at the chan_sccp driver.. it is supposed
to have support for the "sidecar".
As far as I know there are no phones that will
support monitoring 20 extensions with BLF.
Regards,
Derek

- Original Message -
From:
Craig Bruenderman
To: asterisk-users@lists.digium.com

Sent: Sunday, July 24, 2005 5:30 PM
Subject: [Asterisk-Users] Busy Lamp Field
SIP Phone

Does anyone have a recommendation
for a good SIP phone with a busy lamp field? I need my operator to be able to
see extension status for about 20 extensions and transfer via HOLD + extension
button. Ive got a pair of SNOM 360s with the sidecar, but Im very
disappointed with them. The buttons are cheap and rubbery like a Sipura 841,
the handset cord is short and cheap, the audio quality is not good, the
speaker is bad, the button layout is too busy, and Ive found it very hard to
train my users how to use.
Has anyone else had the same
experience with the 360s? Im ready to replace them with anything. I put
Polycom 501s in for everyone else and they are
wonderful.
Does the Cisco 7960 + 7914 work
well for such an application? Keep in mind it has to show free/busy status of
20 although I guess we could get by with 14 if the 7960 + 7914 is a solid
option. Im in Louisville,
KY if anyone knows of a
distributor or retailer that I could demo a unit
from.
Thanks
Craig
Bruenderman
Network Advocates,
Inc.9001 Shelbyville
Road, 260 BurhansLouisville, KY
40222Main: 502-412-1050Free:
877-412-1050DID: 502-992-5929Fax:
502-412-1058Mobile:
502-548-1100

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Re: [Asterisk-Users] Help [EMAIL PROTECTED]

What is happening is that the _X. extension is catching the call... you need
to take it out... it was only meant as a test to make sure which extension
it was actually being sent to...

It seems that it is then executing your vm macro without any parameters...
don't know why that would happen...

Try again after removing the catchall...

Regards,
Derek

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

Sent: Sunday, July 24, 2005 4:35 PM
Subject: RE: [Asterisk-Users] Help
[EMAIL PROTECTED]


>
>  OK, I added in the rule you gave me below, and yep it said 201, so your
right
> on with that one.
>
> So with the following extension rules:
>
> ;setup SIP extension for BroadVoice
> [globals]
> BVNUMBER=240524 ; your calling number
> BVRINGS=201 ; the phone to ring
> BVVMBOX=201 ; the VM box for this user
>
>
> [outrt-003-BroadVoice]
> include => outrt-003-BroadVoice-custom
> exten => _8.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30)
> exten => _8.,2,Congestion()
> exten => _8.,102,Busy()
>
> [frombroadvoice]
> exten => _X.,1,Noop(Incoming call for extension ${EXTEN}in context
> frombroadvoice)
> exten => 201,1,Macro(exten-vm,[EMAIL PROTECTED],${BVRINGS})
> exten => ${VM_PREFIX}${BVVMBOX},1,Macro(vm,${BVVMBOX})
>
>
>
> I now see:
>
>
> Jul 24 18:25:37 DEBUG[1078]: Setting NAT on RTP to 0
> Jul 24 18:25:37 DEBUG[1078]: Check for res for 240524
> Jul 24 18:25:37 DEBUG[1078]: 240524 is not a local user
> Jul 24 18:25:37 DEBUG[1078]: build_route: Contact hop:
> 
> Jul 24 18:25:37 VERBOSE[1078]: -- Executing
NoOp("SIP/240524-13be",
> "Incoming call for extension 701in context frombroadvoice") in new stack
> Jul 24 18:25:47 VERBOSE[1078]:   == CDR updated on SIP/240524-13be
> Jul 24 18:25:47 VERBOSE[1078]: -- Executing
Macro("SIP/240524-13be",
> "vm|") in new stack
> Jul 24 18:25:47 VERBOSE[1078]: -- Executing
Goto("SIP/240524-13be",
> "s-|1") in new stack
> Jul 24 18:25:47 VERBOSE[1078]: -- Goto (macro-vm,s-,1)
> Jul 24 18:25:47 VERBOSE[1078]: -- Executing
> VoiceMail("SIP/240524-13be", "u") in new stack
> Jul 24 18:25:47 WARNING[1078]: No entry in voicemail config file for ''
> Jul 24 18:25:47 VERBOSE[1078]: -- Executing
Hangup("SIP/240524-13be",
> "") in new stack
> Jul 24 18:25:47 VERBOSE[1078]:   == Spawn extension (macro-vm, s-, 2)
exited
> non-zero on 'SIP/240524-13be' in macro 'vm'
> Jul 24 18:25:47 VERBOSE[1078]:   == Spawn extension (frombroadvoice, , 1)
> exited non-zero on 'SIP/240524-13be'
> Jul 24 18:25:47 DEBUG[1078]: cdr_mysql: inserting a CDR record.
> Jul 24 18:25:47 DEBUG[1078]: cdr_mysql: SQL command as follows:  INSERT
INTO
> cdr
>
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration
,b
> illsec,disposition,amaflags,accountcode) VALUES ('2005-07-24
> 18:25:37','\"Leadmon H \" <410515>','410515','','frombroadvoice',
> 'SIP/240524-13be','','Hangup','',10,0,'ANSWERED',3,'')
> Jul 24 18:25:47 DEBUG[1078]: update_user_counter(240524) - decrement
inUse
> counter
> Jul 24 18:25:47 DEBUG[1078]: 240524 is not a local user
> Jul 24 18:25:47 DEBUG[1078]: Stopping retransmission on
> 'SD2o9vc01-cd0c7ebdf9d4cc726c74f1e511173834-js11002' of Response
628736564:
> Found
> Jul 24 18:25:47 DEBUG[1078]: Stopping retransmission on
> 'SD2o9vc01-cd0c7ebdf9d4cc726c74f1e511173834-js11002' of Request 102: Found
>
>
>
> On a side note, and I want to mention it as maybe relevant, something is
now
> screwy with my vMail, which is strange as I haven't done anything but the
> tests here for you.   I can access the vmail, and if an extension is busy
it
> goes to the vmail.   What is messed up is that if I dial an extension and
just
> let it ring, it just keeps on ringing and doesn't transfer out.  For the
life
> of me I can't figure that one out right now, and don't see how just
changing
> this broadvoice context now has it so extension to extension (say 200
calling
> 202) is no longer getting a mailbox.   Never easy..  :(
>
>
> Still when I call the broadvoice number, something changed, as now I get a
> moment of silence, and then it hangs up.  No ringing, no vmail, actually I
> think I got a fraction of a ring one time, then it hung up.
>
> I am for sure still confused on this one, so any help much appreciated...
>
>
> ---
> Howard Leadmon - howard at leadmon.net
> http://www.leadmon.net
>
>
> > -Original Message-
> > From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > [EMAIL PROTECTED] On Behalf Of dbruce
> > Sent: Sunday, July 24, 2005 5:04 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] Help with
> > [EMAIL PROTECTED]
> >
> > Ok.. I screwed up...
> >
> > You have a register statement:
> >
[EMAIL PROTECTED]:123abc:[EMAIL PROTECTED]/
> > 201
> >
> > so, the incoming call from broadvoice will be sent to extension 201 in
the
> > frombroadvoice context.
> >
> > To ensure what is

[Asterisk-Users] Busy Lamp Field SIP Phone

2005-07-24 Thread Craig Bruenderman









Does anyone have a recommendation for a good SIP phone with
a busy lamp field? I need my operator to be able to see extension status for
about 20 extensions and transfer via HOLD + extension button. I’ve got a
pair of SNOM 360s with the sidecar, but I’m very disappointed with them.
The buttons are cheap and rubbery like a Sipura 841, the handset cord is short
and cheap, the audio quality is not good, the speaker is bad, the button layout
is too busy, and I’ve found it very hard to train my users how to use.

 

Has anyone else had the same experience with the 360s? I’m
ready to replace them with anything. I put Polycom 501’s in for everyone
else and they are wonderful.

 

Does the Cisco 7960 + 7914 work well for such an
application? Keep in mind it has to show free/busy status of 20 although I
guess we could get by with 14 if the 7960 + 7914 is a solid option. I’m
in Louisville, KY if anyone knows of a distributor or
retailer that I could demo a unit from.

 

Thanks

 

Craig Bruenderman

Network Advocates, Inc.
9001 Shelbyville Road,
260 Burhans
Louisville, KY  40222
Main:  502-412-1050
Free:  877-412-1050
DID:   502-992-5929
Fax:   502-412-1058
Mobile: 
502-548-1100

 






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[Asterisk-Users] Need to ztcfg every time I reboot *




Hello
 
I am sure this is a very basic Linux 
question.
 
But every time I reboot my * I need to 

 
modprobe 
 
and then
 
ztcfg
 
After doing this I can then run * without it 
complaining about not loading a channel.  The module being loaded is qozap 
- a ISDN card.
 
What do I need to do to make the ztcfg 
configuration persistent?
 
Angus
 
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Re: [Asterisk-Users] Why can't sip/200 call sip/202

That was another problem - now fixed.

Thanks for all your help on extensions.conf

Angus

- Original Message - 
From: "Angus Comber" <[EMAIL PROTECTED]>
To: "Mark Edwards" <[EMAIL PROTECTED]>; "Asterisk Users Mailing 
List - Non-Commercial Discussion" 

Sent: Sunday, July 24, 2005 11:30 PM
Subject: Re: [Asterisk-Users] Why can't sip/200 call sip/202

Sorry to be a pain... but

I restarted my * and now when I launch * get this:

 == Parsing '/etc/asterisk/zapata.conf': Found
Jul 24 18:52:45 WARNING[6817]: chan_zap.c:932 zt_open: Unable to specify 
channel 1: No such device or address
Jul 24 18:52:45 ERROR[6817]: chan_zap.c:6473 mkintf: Unable to open 
channel 1: No such device or address

here = 0, tmp->channel = 1, channel = 1
Jul 24 18:52:45 ERROR[6817]: chan_zap.c:10303 setup_zap: Unable to 
register channel '1-2'
Jul 24 18:52:45 WARNING[6817]: loader.c:345 ast_load_resource: 
chan_zap.so: load_module failed, returning -1

 == Unregistered channel type 'Tor'
 == Unregistered channel type 'Zap'
Jul 24 18:52:45 WARNING[6817]: loader.c:440 load_modules: Loading module 
chan_zap.so failed!

linux:~ # Ouch ... error while writing audio data: : Broken pipe

I have a Junghanns quadBRI card installed.  I have modprobe qozap - so it 
is loaded and seems to be working OK.  I assume there was something in 
extensions.conf which was somehow required.  something to do with 
[channels] ?

The error in chan_zap.c seems to be saying that channel 1 cannot be 
opened.

Angus

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

Sent: Sunday, July 24, 2005 10:13 PM
Subject: Re: [Asterisk-Users] Why can't sip/200 call sip/202

OK Angus

just start here

mv extensions.conf extensions.conf.old

and create a new extensions.conf

[default]
exten => _2XX,1,Dial(SIP/${EXTEN},20,Ttm)
exten => _2XX,2,Hangup

just those 3 lines

do an 'extensions reload' in the CLI or just restart Asterisk

and see if it works

regards,

Mark.
On 7/25/05, Angus Comber <[EMAIL PROTECTED]> wrote:
I think the 777 may be a bit of a Red Herring.  I dialed 777 as a test. 
I

can't dial 202 from 200 if I actually dial 202!

My extensions.conf file:

;
; Static extension configuration file, used by
; the pbx_config module. This is where you configure all your
; inbound and outbound calls in Asterisk.
;
; This configuration file is reloaded
; - With the "extensions reload" command in the CLI
; - With the "reload" command (that reloads everything) in the CLI

;
; The "General" category is for certain variables.
;
[general]
;
; If static is set to no, or omitted, then the pbx_config will rewrite
; this file when extensions are modified.  Remember that all comments
; made in the file will be lost when that happens.
;
; XXX Not yet implemented XXX
;
static=yes
;
; if static=yes and writeprotect=no, you can save dialplan by
; CLI command 'save dialplan' too
;
writeprotect=no

; You can include other config files, use the #include command (without 
the

';')
; Note that this is different from the "include" command that includes
contexts within
; other contexts. The #include command works in all asterisk 
configuration

files.
;#include "filename.conf"

; The "Globals" category contains global variables that can be referenced
; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental
variable
; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid
;
[globals]
CONSOLE=Console/dsp; Console interface for demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
IAXINFO=guest ; IAXtel username/password
;IAXINFO=myuser:mypass
TRUNK=Zap/g2 ; Trunk interface
;
; Note the 'g2' in the TRUNK variable above. It specifies which group
(defined
; in zapata.conf) to dial, i.e. group 2, and how to choose a channel to 
use

in
; the specified group. The four possible options are:
;
; g: select the lowest-numbered non-busy Zap channel (aka. ascending
sequential hunt group).
; G: select the highest-numbered non-busy Zap channel (aka. descending
sequential hunt group).
; r: use a round-robin search, starting at the next highest channel than
last time (aka. ascending rotary hunt group).
; R: use a round-robin search, starting at the next lowest channel than 
last

time (aka. descending rotary hunt group).
;
TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
;TRUNK=IAX2/user:[EMAIL PROTECTED]

;
; Any category other than "General" and "Globals" represent
; extension contexts, which are collections of extensions.
;
; Extension names may be numbers, letters, or combinations
; thereof. If an extension name is prefixed by a '_'
; character, it is interpreted as a pattern rather than a
; literal.  In patterns, some characters have special meanings:
;
;   X - any digit from 0-9
;   Z - any digit from 1-9
;   N - any digit from 2-9
;   [1235-9] - any digit in the brackets (in this example, 
1,2,3,5,6,7,8,9)

;   . - wildcard, matches anything remaining (e.g. _9011. matches
; an

Re: [Asterisk-Users] Did anyone else get spammed by GIZMO?

2005-07-24 Thread Steve Totaro

Just because you give your email address does not allow for unsolicited
emails unless you agree in the EULA or terms and conditions.


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

Sent: Sunday, July 24, 2005 3:13 PM
Subject: Re: [Asterisk-Users] Did anyone else get spammed by GIZMO?


Interesting, I've been on this list for almost a year now, and I
didn't recieve this spam. Are you sure you didn't download any sip
softphones and gave your email address?
In which case it is NOT spam, you gave your email address to them.


On 7/22/05, Jay Milk <[EMAIL PROTECTED]> wrote:
> > -Original Message-
> > From: Jerry Glomph Black [mailto:[EMAIL PROTECTED]
> > Sent: Friday, July 22, 2005 2:18 AM
> > Subject: Re: [Asterisk-Users] Did anyone else get spammed by GIZMO?
> >
> > It's not a spam.  They are not yokels.  Don't know about you
> >
> > Gizmo is basically a different front end offered by the
> > Sipphone.com people, to offer an alternative to Skype which
> > is not a closed jail (interoperates with all
> > SIP devices, asterisk, etc.).
> >
> > I think they sent the mail to all registered sipphone.com users.
>
> Dear Glomph,
>
> Thanks, I know what the Gizmo project is.  I'm just not interested.
>
> And yes, it's spam:
> 1. It's unsolicited -- I'm not registered on Sipphone.com.  It came in
> through an email address I use exclusively for the asterisk-users list
> and a small volume of personal mail.
> 2. It's commercial -- #5 of "10 more things to do" reads "Add $10 in
> CallOut credit to your account and talk for hours".  So, they're trying
> to sell me something.
>
> Yokel: "A clumsy, unsophisticated person".  I think that would be their
> preferred label... Why?  Because if you send unsolicited commercial
> email in the US without offering a way to opt out of future mailings,
> you're either clumsy and unsophisticated, or you're a criminal.  I gave
> them the benefit of the doubt and called them "yokels".  If that term
> offends you in any way, then let's use the more factual "criminals."
>
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Re: [Asterisk-Users] Asterisk Crashes after update

2005-07-24 Thread Scott Brown


Thanks Brian:

Sorry for the late reply.  That was helpful.  Problem solved.

Scott

At 10:43 PM 7/10/2005, you wrote:

You might want to recompile the res_config_mysql or configure
res_config_odbc which works via myodbc and is just as good!

/b
---
Anakin: "You're either with me, or you're my enemy."
Obi-Wan: "Only a Sith could be an absolutist."

On Jul 7, 2005, at 2:46 PM, [EMAIL PROTECTED] wrote:


After doing an update from SUSE 9.2 to 9.3 and Checking out the
latest from CVS, Asterisk crashes on startup with an apparent MySQL
(res_config_register) error:
# asterisk -vvvgc > asterisk_startup_error1.log
asterisk: symbol lookup error:
/usr/lib/asterisk/modules/res_config_mysql.so: un
   defined symbol: ast_cust_config_register
The log is shown below.  I've seen the posts from
1/25/05 and several more recent ones regarding this
same issue or a similar one with the
ast_cust_config_register being undefined, however
reverting to that build of 1/24/05 does not solve the
problem in my case.
Is there another issue with mySQL that may cause this
problem?  I'm using SUSE 9.3 on an Athlon 64 with 64
bit release 2.6 of Linux.  I've made sure that all the
ODBC and MySQL modules for SUSE 9.3 are installed.
I'm a rank noob with * and would appreciate any help.
Thanks!!!
Log Pasted below for more info:
[0;37;40m
[1;30;40m  == [0;37;40mParsing
'/etc/asterisk/asterisk.conf': Found
[1;30;40m  == [0;37;40mParsing
'/etc/asterisk/extconfig.conf': Found
[1;30;40m  == [0;37;40mParsing
'/etc/asterisk/asterisk.conf': Found
Asterisk CVS-HEAD, Copyright (C) 1999 - 2005 Digium.
Written by Mark Spencer <[EMAIL PROTECTED]>
== ===
[1;30;40m  == [0;37;40mParsing
'/etc/asterisk/logger.conf': Found
Asterisk Event Logger Started
/var/log/asterisk/event_log
Asterisk Dynamic Loader loading preload modules:
[1;30;40m  == [0;37;40mParsing
'/etc/asterisk/modules.conf': Found
[1;30;40m  == [0;37;40mManager registered action
Ping
[1;30;40m  == [0;37;40mManager registered action
Events
[1;30;40m  == [0;37;40mManager registered action
Logoff
[1;30;40m  == [0;37;40mManager registered action
Hangup
[1;30;40m  == [0;37;40mManager registered action
Status
[1;30;40m  == [0;37;40mManager registered action
Setvar
[1;30;40m  == [0;37;40mManager registered action
Getvar
[1;30;40m  == [0;37;40mManager registered action
Redirect
[1;30;40m  == [0;37;40mManager registered action
Originate
[1;30;40m  == [0;37;40mManager registered action
Command
[1;30;40m  == [0;37;40mManager registered action
ExtensionState
[1;30;40m  == [0;37;40mManager registered action
AbsoluteTimeout
[1;30;40m  == [0;37;40mManager registered action
MailboxStatus
[1;30;40m  == [0;37;40mManager registered action
MailboxCount
[1;30;40m  == [0;37;40mManager registered action
ListCommands
[1;30;40m  == [0;37;40mParsing
'/etc/asterisk/manager.conf': Found
[1;30;40m  == [0;37;40mParsing
'/etc/asterisk/cdr.conf': Not found (No such file or
directory)
Jul  6 21:32:24 [1;33;40mNOTICE[0;37;40m[8492]:
[1;37;40mcdr.c[0;37;40m:[1;37;40m1162[0;37;40m
[1;37;40mdo_reload[0;37;40m: CDR simple logging
enabled.
[1;30;40m  == [0;37;40mParsing
'/etc/asterisk/rtp.conf': Found
[1;30;40m  == [0;37;40mRTP Allocating from port
range 1 -> 2
Asterisk PBX Core Initializing
Registering builtin applications:
[1;30;40m [0;37;40m[AbsoluteTimeout]
[1;30;40m  == [0;37;40mRegistered application
'[1;36;40mAbsoluteTimeout[0;37;40m'
[1;30;40m [0;37;40m[Answer]
[1;30;40m  == [0;37;40mRegistered application
'[1;36;40mAnswer[0;37;40m'
[1;30;40m [0;37;40m[BackGround]
[1;30;40m  == [0;37;40mRegistered application
'[1;36;40mBackGround[0;37;40m'
[1;30;40m [0;37;40m[Busy]
[1;30;40m  == [0;37;40mRegistered application
'[1;36;40mBusy[0;37;40m'
[1;30;40m [0;37;40m[Congestion]
[1;30;40m  == [0;37;40mRegistered application
'[1;36;40mCongestion[0;37;40m'
[1;30;40m [0;37;40m[DigitTimeout]
[1;30;40m  == [0;37;40mRegistered application
'[1;36;40mDigitTimeout[0;37;40m'
[1;30;40m [0;37;40m[Goto]
[1;30;40m  == [0;37;40mRegistered application
'[1;36;40mGoto[0;37;40m'
[1;30;40m [0;37;40m[GotoIf]
[1;30;40m  == [0;37;40mRegistered application
'[1;36;40mGotoIf[0;37;40m'
[1;30;40m [0;37;40m[GotoIfTime]
[1;30;40m  == [0;37;40mRegistered application
'[1;36;40mGotoIfTime[0;37;40m'
[1;30;40m [0;37;40m[ExecIfTime]
[1;30;40m  == [0;37;40mRegistered application
'[1;36;40mExecIfTime[0;37;40m'
[1;30;40m [0;37;40m[Hangup]
[1;30;40m  == [0;37;40mRegistered application
'[1;36;40mHangup[0;37;40m'
[1;30;40m [0;37;40m[NoOp]
[1;30;40m  == [0;37;40mRegistered application
'[1;36;40mNoOp[0;37;40m'
[1;30;40m [0;37;40m[Prefix]
[1;30;40m  == [0;37;40mRegistered application
'[1;36;40mPrefix[0;37;40m'
[1;30;40m [0;37;40m[Progress]
[1;30;40m  == [0;37;40mRegistered application
'[1;36;40mProgress[0;37;40m'
[1;30;40m [0;37;40m[ResetCDR]
[1;30;40m  == [0;37;40mRegistered application
'[1;36;40mResetCDR[0;37;40m'
[1;30;40m [0;37;40m[ResponseTimeout]
[1;30;40m  == [0;37;40mRegistered appl

RE: [Asterisk-Users] Help with [EMAIL PROTECTED]


 OK, I added in the rule you gave me below, and yep it said 201, so your right
on with that one.

So with the following extension rules:

;setup SIP extension for BroadVoice
[globals]
BVNUMBER=240524 ; your calling number 
BVRINGS=201 ; the phone to ring
BVVMBOX=201 ; the VM box for this user


[outrt-003-BroadVoice]
include => outrt-003-BroadVoice-custom
exten => _8.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30)
exten => _8.,2,Congestion()
exten => _8.,102,Busy()

[frombroadvoice]
exten => _X.,1,Noop(Incoming call for extension ${EXTEN}in context
frombroadvoice)
exten => 201,1,Macro(exten-vm,[EMAIL PROTECTED],${BVRINGS})
exten => ${VM_PREFIX}${BVVMBOX},1,Macro(vm,${BVVMBOX})



I now see:


Jul 24 18:25:37 DEBUG[1078]: Setting NAT on RTP to 0
Jul 24 18:25:37 DEBUG[1078]: Check for res for 240524
Jul 24 18:25:37 DEBUG[1078]: 240524 is not a local user
Jul 24 18:25:37 DEBUG[1078]: build_route: Contact hop:

Jul 24 18:25:37 VERBOSE[1078]: -- Executing NoOp("SIP/240524-13be",
"Incoming call for extension 701in context frombroadvoice") in new stack
Jul 24 18:25:47 VERBOSE[1078]:   == CDR updated on SIP/240524-13be
Jul 24 18:25:47 VERBOSE[1078]: -- Executing Macro("SIP/240524-13be",
"vm|") in new stack
Jul 24 18:25:47 VERBOSE[1078]: -- Executing Goto("SIP/240524-13be",
"s-|1") in new stack
Jul 24 18:25:47 VERBOSE[1078]: -- Goto (macro-vm,s-,1)
Jul 24 18:25:47 VERBOSE[1078]: -- Executing
VoiceMail("SIP/240524-13be", "u") in new stack
Jul 24 18:25:47 WARNING[1078]: No entry in voicemail config file for ''
Jul 24 18:25:47 VERBOSE[1078]: -- Executing Hangup("SIP/240524-13be",
"") in new stack
Jul 24 18:25:47 VERBOSE[1078]:   == Spawn extension (macro-vm, s-, 2) exited
non-zero on 'SIP/240524-13be' in macro 'vm'
Jul 24 18:25:47 VERBOSE[1078]:   == Spawn extension (frombroadvoice, , 1)
exited non-zero on 'SIP/240524-13be'
Jul 24 18:25:47 DEBUG[1078]: cdr_mysql: inserting a CDR record.
Jul 24 18:25:47 DEBUG[1078]: cdr_mysql: SQL command as follows:  INSERT INTO
cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,b
illsec,disposition,amaflags,accountcode) VALUES ('2005-07-24
18:25:37','\"Leadmon H \" <410515>','410515','','frombroadvoice',
'SIP/240524-13be','','Hangup','',10,0,'ANSWERED',3,'')
Jul 24 18:25:47 DEBUG[1078]: update_user_counter(240524) - decrement inUse
counter
Jul 24 18:25:47 DEBUG[1078]: 240524 is not a local user
Jul 24 18:25:47 DEBUG[1078]: Stopping retransmission on
'SD2o9vc01-cd0c7ebdf9d4cc726c74f1e511173834-js11002' of Response 628736564:
Found
Jul 24 18:25:47 DEBUG[1078]: Stopping retransmission on
'SD2o9vc01-cd0c7ebdf9d4cc726c74f1e511173834-js11002' of Request 102: Found



On a side note, and I want to mention it as maybe relevant, something is now
screwy with my vMail, which is strange as I haven't done anything but the
tests here for you.   I can access the vmail, and if an extension is busy it
goes to the vmail.   What is messed up is that if I dial an extension and just
let it ring, it just keeps on ringing and doesn't transfer out.  For the life
of me I can't figure that one out right now, and don't see how just changing
this broadvoice context now has it so extension to extension (say 200 calling
202) is no longer getting a mailbox.   Never easy..  :(


Still when I call the broadvoice number, something changed, as now I get a
moment of silence, and then it hangs up.  No ringing, no vmail, actually I
think I got a fraction of a ring one time, then it hung up.  

I am for sure still confused on this one, so any help much appreciated...


---
Howard Leadmon - howard at leadmon.net
http://www.leadmon.net 


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of dbruce
> Sent: Sunday, July 24, 2005 5:04 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Help with
> [EMAIL PROTECTED]
> 
> Ok.. I screwed up...
> 
> You have a register statement:
> [EMAIL PROTECTED]:123abc:[EMAIL PROTECTED]/
> 201
> 
> so, the incoming call from broadvoice will be sent to extension 201 in the
> frombroadvoice context.
> 
> To ensure what is going on, use this as your context.
> exten => _X.,1,Noop(Incoming call for extension ${EXTEN} in context
> frombroadvoice)
> 
> That will tell you exactly what is being sent into the context.
> 
> You are using the latest AAH, so the variable substitutions will work.
> 
> I expect that you will end up using the following in frombroadvoice:
> 
> exten => ${BVRINGS},1,Macro(exten-vm,[EMAIL PROTECTED],${BVRINGS})
> 
> Sorry for the confusion
> 
> Regards,
> Derek

(Old parts removed)



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[Asterisk-Users] sip messaging (tested on eyeBeam) support

2005-07-24 Thread Juraj Bednar

Hello,

 I added queuing support (based on SQLite database to store the queue)
for my SIP Messaging patch. Works with eyeBeam, probably lots of bugs,
but it's at least something.

 I created page about installation on the tips and tricks of voip-info.org:

 http://www.voip-info.org/tiki-index.php?page=Asterisk+SIP+Messaging

 Any bugfixes are welcome.

 Yes, it's a huge hack and supports only sip to sip messaging based on
presence hints.


Juraj.
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[Asterisk-Users] Incoming call prob

2005-07-24 Thread Michael Beale

I am having a problem with your my nufone service. 
I'm trying to setup incoming calls and I'm having no
success. Outgoing works fine though.  The message I'm
getting is "the person you are call is not currently
reachable".  I'm going to give you as much info as I
can.  I'm also an asterisk newb! Anyways, I installed
[EMAIL PROTECTED]  Set up extensions which communicate to
each other fine.  Everything is default except here is
my iax.conf:

[general]
bindport = 4569   ; Port to bind to (IAX is
4569)
bindaddr = 0.0.0.0; Address to bind to (all
addresses on machine)
disallow=all
allow=ulaw
allow=alaw
allow=gsm
mailboxdetail=yes
register => meadmaker:@switch-2.nufone.net
#include iax_additional.conf
#include iax_custom.conf

[NuFone]
type=peer
host=switch-1.nufone.net
secret=

and here is what I added to extension.conf:

;added for nufone
exten => _1NXXNXX,1,SetCallerID(8662521540)
exten =>
_1NXXNXX,2,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}


[inbound]
exten => 8662521540,1,Answer()
exten => 8662521540,2,Wait(2)
exten => 8662521540,3,Dial(200)

I would appreciate some help.  Trying to get this
working today.  Have wiped out my machine because of
experimentation 3 times, all I changed was to add 2
extension and the conf files above.  Please let me
know, probably a dumb newb error!  

Thanks

Michael Beale




Start your day with Yahoo! - make it your home page 
http://www.yahoo.com/r/hs 
 
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Re: [Asterisk-Users] Why can't sip/200 call sip/202


Sorry to be a pain... but

I restarted my * and now when I launch * get this:

 == Parsing '/etc/asterisk/zapata.conf': Found
Jul 24 18:52:45 WARNING[6817]: chan_zap.c:932 zt_open: Unable to specify 
channel 1: No such device or address
Jul 24 18:52:45 ERROR[6817]: chan_zap.c:6473 mkintf: Unable to open channel 
1: No such device or address

here = 0, tmp->channel = 1, channel = 1
Jul 24 18:52:45 ERROR[6817]: chan_zap.c:10303 setup_zap: Unable to register 
channel '1-2'
Jul 24 18:52:45 WARNING[6817]: loader.c:345 ast_load_resource: chan_zap.so: 
load_module failed, returning -1

 == Unregistered channel type 'Tor'
 == Unregistered channel type 'Zap'
Jul 24 18:52:45 WARNING[6817]: loader.c:440 load_modules: Loading module 
chan_zap.so failed!

linux:~ # Ouch ... error while writing audio data: : Broken pipe

I have a Junghanns quadBRI card installed.  I have modprobe qozap - so it is 
loaded and seems to be working OK.  I assume there was something in 
extensions.conf which was somehow required.  something to do with [channels] 
?


The error in chan_zap.c seems to be saying that channel 1 cannot be opened.

Angus




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


Sent: Sunday, July 24, 2005 10:13 PM
Subject: Re: [Asterisk-Users] Why can't sip/200 call sip/202


OK Angus

just start here

mv extensions.conf extensions.conf.old

and create a new extensions.conf

[default]
exten => _2XX,1,Dial(SIP/${EXTEN},20,Ttm)
exten => _2XX,2,Hangup



just those 3 lines

do an 'extensions reload' in the CLI or just restart Asterisk

and see if it works

regards,

Mark.
On 7/25/05, Angus Comber <[EMAIL PROTECTED]> wrote:

I think the 777 may be a bit of a Red Herring.  I dialed 777 as a test.  I
can't dial 202 from 200 if I actually dial 202!

My extensions.conf file:


;
; Static extension configuration file, used by
; the pbx_config module. This is where you configure all your
; inbound and outbound calls in Asterisk.
;
; This configuration file is reloaded
; - With the "extensions reload" command in the CLI
; - With the "reload" command (that reloads everything) in the CLI

;
; The "General" category is for certain variables.
;
[general]
;
; If static is set to no, or omitted, then the pbx_config will rewrite
; this file when extensions are modified.  Remember that all comments
; made in the file will be lost when that happens.
;
; XXX Not yet implemented XXX
;
static=yes
;
; if static=yes and writeprotect=no, you can save dialplan by
; CLI command 'save dialplan' too
;
writeprotect=no

; You can include other config files, use the #include command (without 
the

';')
; Note that this is different from the "include" command that includes
contexts within
; other contexts. The #include command works in all asterisk configuration
files.
;#include "filename.conf"

; The "Globals" category contains global variables that can be referenced
; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental
variable
; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid
;
[globals]
CONSOLE=Console/dsp; Console interface for demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
IAXINFO=guest ; IAXtel username/password
;IAXINFO=myuser:mypass
TRUNK=Zap/g2 ; Trunk interface
;
; Note the 'g2' in the TRUNK variable above. It specifies which group
(defined
; in zapata.conf) to dial, i.e. group 2, and how to choose a channel to 
use

in
; the specified group. The four possible options are:
;
; g: select the lowest-numbered non-busy Zap channel (aka. ascending
sequential hunt group).
; G: select the highest-numbered non-busy Zap channel (aka. descending
sequential hunt group).
; r: use a round-robin search, starting at the next highest channel than
last time (aka. ascending rotary hunt group).
; R: use a round-robin search, starting at the next lowest channel than 
last

time (aka. descending rotary hunt group).
;
TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
;TRUNK=IAX2/user:[EMAIL PROTECTED]

;
; Any category other than "General" and "Globals" represent
; extension contexts, which are collections of extensions.
;
; Extension names may be numbers, letters, or combinations
; thereof. If an extension name is prefixed by a '_'
; character, it is interpreted as a pattern rather than a
; literal.  In patterns, some characters have special meanings:
;
;   X - any digit from 0-9
;   Z - any digit from 1-9
;   N - any digit from 2-9
;   [1235-9] - any digit in the brackets (in this example, 
1,2,3,5,6,7,8,9)

;   . - wildcard, matches anything remaining (e.g. _9011. matches
; anything starting with 9011 excluding 9011 itself)
;
; For example the extension _NXX would match normal 7 digit dialings,
; while _1NXXNXX would represent an area code plus phone number
; preceeded by a one.
;
; Each step of an extension is ordered by priority, which must
; always start with 1 to be considered a valid extension.
;
; Contexts contain several l

Re: [Asterisk-Users] why this macro works Only with a * Stable Version? ; -(

2005-07-24 Thread Kevin P. Fleming


xAD wrote:

Hi all, i have a weird problem, i need to know why this 'simple' macro works 
Only with a * STABLE version (1.0.0.x to 1.0.0.9) , if i try it with a CVS HEAD 
version.. don't works ;-(

Thanks in advance.

[local]

include => default

[macro-tono_simulato]
exten => s,1,Set(DIALED=${ARG1})
exten => s,2,Set(TOCONTEXT=${ARG2})
exten => s,3,Set(STRIP=${ARG3})
exten => s,4,Set(TONES=dial)  
exten => s,5,Goto(tono_simulato_pausa|s|1)


This macro _cannot_ work in version 1.0.x, because Set() is not 
supported there (only SetVar()).

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Re: [Asterisk-Users] web managment

2005-07-24 Thread C F

Try VIM and Apache. If you know what you doing you will have one that
at least you like.

On 7/22/05, Dante Renda <[EMAIL PROTECTED]> wrote:
> what is the best web based managment aplication for asterisk ???
> 
> Dante
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Re: [Asterisk-Users] Did anyone else get spammed by GIZMO?

2005-07-24 Thread C F

Interesting, I've been on this list for almost a year now, and I
didn't recieve this spam. Are you sure you didn't download any sip
softphones and gave your email address?
In which case it is NOT spam, you gave your email address to them.


On 7/22/05, Jay Milk <[EMAIL PROTECTED]> wrote:
> > -Original Message-
> > From: Jerry Glomph Black [mailto:[EMAIL PROTECTED]
> > Sent: Friday, July 22, 2005 2:18 AM
> > Subject: Re: [Asterisk-Users] Did anyone else get spammed by GIZMO?
> >
> > It's not a spam.  They are not yokels.  Don't know about you
> >
> > Gizmo is basically a different front end offered by the
> > Sipphone.com people, to offer an alternative to Skype which
> > is not a closed jail (interoperates with all
> > SIP devices, asterisk, etc.).
> >
> > I think they sent the mail to all registered sipphone.com users.
> 
> Dear Glomph,
> 
> Thanks, I know what the Gizmo project is.  I'm just not interested.
> 
> And yes, it's spam:
> 1. It's unsolicited -- I'm not registered on Sipphone.com.  It came in
> through an email address I use exclusively for the asterisk-users list
> and a small volume of personal mail.
> 2. It's commercial -- #5 of "10 more things to do" reads "Add $10 in
> CallOut credit to your account and talk for hours".  So, they're trying
> to sell me something.
> 
> Yokel: "A clumsy, unsophisticated person".  I think that would be their
> preferred label... Why?  Because if you send unsolicited commercial
> email in the US without offering a way to opt out of future mailings,
> you're either clumsy and unsophisticated, or you're a criminal.  I gave
> them the benefit of the doubt and called them "yokels".  If that term
> offends you in any way, then let's use the more factual "criminals."
> 
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Re: [Asterisk-Users] Why can't sip/200 call sip/202


No please use ${EXTEN}, ${ARG1} is for macros.

And of course you will use the protocol in front of ${EXTEN}

So for SIP use:

exten =>  _2XX,1,Dial(SIP/${EXTEN},30)

and for IAX2 use:

exten =>  _2XX,1,Dial(IAX2/${EXTEN},30)

Regards,

Marc

Angus Comber wrote:

Would this do it:

exten =>  _2XX,1,Dial(${ARG1},30)

Then I would fallback to voicemail (or something else) after the 30 
seconds?


Angus



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


Sent: Sunday, July 24, 2005 10:06 PM
Subject: Re: [Asterisk-Users] Why can't sip/200 call sip/202


Ok your extensions.conf doesn't mention anything about an 
extension/number equal to 202 or 200. You must know that the name of a 
SIP and IAX2 peer is only an "address", you will have to assign a 
number via extensions.conf to this address.


Have a look at www.voip-info.org and of course google.com to get to 
know extensions.conf.


Regards,

Marc

Angus Comber wrote:

I think the 777 may be a bit of a Red Herring.  I dialed 777 as a 
test. I can't dial 202 from 200 if I actually dial 202!


My extensions.conf file:


;
; Static extension configuration file, used by
; the pbx_config module. This is where you configure all your
; inbound and outbound calls in Asterisk.
;
; This configuration file is reloaded
; - With the "extensions reload" command in the CLI
; - With the "reload" command (that reloads everything) in the CLI

;
; The "General" category is for certain variables.
;
[general]
;
; If static is set to no, or omitted, then the pbx_config will rewrite
; this file when extensions are modified.  Remember that all comments
; made in the file will be lost when that happens.
;
; XXX Not yet implemented XXX
;
static=yes
;
; if static=yes and writeprotect=no, you can save dialplan by
; CLI command 'save dialplan' too
;
writeprotect=no

; You can include other config files, use the #include command 
(without the ';')
; Note that this is different from the "include" command that 
includes contexts within
; other contexts. The #include command works in all asterisk 
configuration files.

;#include "filename.conf"

; The "Globals" category contains global variables that can be 
referenced
; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for 
Environmental variable

; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid
;
[globals]
CONSOLE=Console/dsp; Console interface for demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
IAXINFO=guest ; IAXtel username/password
;IAXINFO=myuser:mypass
TRUNK=Zap/g2 ; Trunk interface
;
; Note the 'g2' in the TRUNK variable above. It specifies which group 
(defined
; in zapata.conf) to dial, i.e. group 2, and how to choose a channel 
to use in

; the specified group. The four possible options are:
;
; g: select the lowest-numbered non-busy Zap channel (aka. ascending 
sequential hunt group).
; G: select the highest-numbered non-busy Zap channel (aka. 
descending sequential hunt group).
; r: use a round-robin search, starting at the next highest channel 
than last time (aka. ascending rotary hunt group).
; R: use a round-robin search, starting at the next lowest channel 
than last time (aka. descending rotary hunt group).

;
TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
;TRUNK=IAX2/user:[EMAIL PROTECTED]

;
; Any category other than "General" and "Globals" represent
; extension contexts, which are collections of extensions.
;
; Extension names may be numbers, letters, or combinations
; thereof. If an extension name is prefixed by a '_'
; character, it is interpreted as a pattern rather than a
; literal.  In patterns, some characters have special meanings:
;
;   X - any digit from 0-9
;   Z - any digit from 1-9
;   N - any digit from 2-9
;   [1235-9] - any digit in the brackets (in this example, 
1,2,3,5,6,7,8,9)

;   . - wildcard, matches anything remaining (e.g. _9011. matches
; anything starting with 9011 excluding 9011 itself)
;
; For example the extension _NXX would match normal 7 digit 
dialings,

; while _1NXXNXX would represent an area code plus phone number
; preceeded by a one.
;
; Each step of an extension is ordered by priority, which must
; always start with 1 to be considered a valid extension.
;
; Contexts contain several lines, one for each step of each
; extension, which can take one of two forms as listed below,
; with the first form being preferred.  One may include another
; context in the current one as well, optionally with a
; date and time.  Included contexts are included in the order
; they are listed.
;
;[context]
;exten => someexten,priority,application(arg1,arg2,...)
;exten => someexten,priority,application,arg1|arg2...
;
; Timing list for includes is
;
;   |||
;
;include => daytime|9:00-17:00|mon-fri|*|*
;
; ignorepat can be used to instruct drivers to not cancel dialtone upon
; receipt of a particular pattern.  The most commonly used example is
; of course '9' like this:
;
;ignorepat => 9
;
; s

Re: [Asterisk-Users] Stupid hold music

2005-07-24 Thread Steve Gladden

OK was actually able to pull it out of the archives!

It's now at http://stuff.michiganbroadband.com/asterisk

I'll leave it there for about a week or two then remove it.
T othe best of my knowledge it's public domain, if anyone needs more info
please contact me offlist.

This is right up there in 'quality' with "Space People".

Steve










> Oh GOSH
> That was awesome!
>
> I have an even better one in store but gotta capture it from it's
> rather obscure source..
>
> Stay tuned!!!
>
> Steve  (N8LBV)
>
>
>
>
>
>
>
>
>
>
>
>>> Does anyone have a collection of stupid hold music? Y'know, the sort of
>>> thing that would drive a person mad? Silly songs, repetative tunes etc?
>>
>> You should be flogged pubically for bringing up this subject - that
>> last "space people" song almost made me wash out my ears with
>> sulphuric acid!!!
>>
>> 73 de NY5I
>> Hatton
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Re: [Asterisk-Users] Why can't sip/200 call sip/202

You observed correctly.  Yes I just copied the sample file, hoping it would 
work.


I didn't realise I had to do anything special with the dialplan just for 
dialing internal extensions.


Can I use something fairly generic like this (assuming all my extensions are 
three digit starting with 2xx):


exten =>  _2XX,1,Dial(${ARG1})

As a VERY basic first attempt.

By the way can I use (${ARG1}) - is it valid?  Or some other variable name 
for number dialed?



Is there an Asterisk document on the dialplan.  Eg all the variables such as 
Dial, Voicemail, etc?  Or do we need to look in a certain .h file?


Angus




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


Sent: Sunday, July 24, 2005 10:10 PM
Subject: Re: [Asterisk-Users] Why can't sip/200 call sip/202



The extensions.conf file you provided looks suspiciously like the asterisk
configs/extensions.conf.sample file.

Did you create a dialplan for your specific configuration or did you just
copy the sample file?



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

Sent: Sunday, July 24, 2005 2:50 PM
Subject: Re: [Asterisk-Users] Why can't sip/200 call sip/202


I think the 777 may be a bit of a Red Herring.  I dialed 777 as a test. 
I

can't dial 202 from 200 if I actually dial 202!

My extensions.conf file:


;
; Static extension configuration file, used by
; the pbx_config module. This is where you configure all your
; inbound and outbound calls in Asterisk.
;
; This configuration file is reloaded
; - With the "extensions reload" command in the CLI
; - With the "reload" command (that reloads everything) in the CLI

;
; The "General" category is for certain variables.
;
[general]
;
; If static is set to no, or omitted, then the pbx_config will rewrite
; this file when extensions are modified.  Remember that all comments
; made in the file will be lost when that happens.
;
; XXX Not yet implemented XXX
;
static=yes
;
; if static=yes and writeprotect=no, you can save dialplan by
; CLI command 'save dialplan' too
;
writeprotect=no

; You can include other config files, use the #include command (without

the

';')
; Note that this is different from the "include" command that includes
contexts within
; other contexts. The #include command works in all asterisk 
configuration

files.
;#include "filename.conf"

; The "Globals" category contains global variables that can be referenced
; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental
variable
; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid
;
[globals]
CONSOLE=Console/dsp; Console interface for demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
IAXINFO=guest ; IAXtel username/password
;IAXINFO=myuser:mypass
TRUNK=Zap/g2 ; Trunk interface
;
; Note the 'g2' in the TRUNK variable above. It specifies which group
(defined
; in zapata.conf) to dial, i.e. group 2, and how to choose a channel to

use

in
; the specified group. The four possible options are:
;
; g: select the lowest-numbered non-busy Zap channel (aka. ascending
sequential hunt group).
; G: select the highest-numbered non-busy Zap channel (aka. descending
sequential hunt group).
; r: use a round-robin search, starting at the next highest channel than
last time (aka. ascending rotary hunt group).
; R: use a round-robin search, starting at the next lowest channel than

last

time (aka. descending rotary hunt group).
;
TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
;TRUNK=IAX2/user:[EMAIL PROTECTED]

;
; Any category other than "General" and "Globals" represent
; extension contexts, which are collections of extensions.
;
; Extension names may be numbers, letters, or combinations
; thereof. If an extension name is prefixed by a '_'
; character, it is interpreted as a pattern rather than a
; literal.  In patterns, some characters have special meanings:
;
;   X - any digit from 0-9
;   Z - any digit from 1-9
;   N - any digit from 2-9
;   [1235-9] - any digit in the brackets (in this example,

1,2,3,5,6,7,8,9)

;   . - wildcard, matches anything remaining (e.g. _9011. matches
; anything starting with 9011 excluding 9011 itself)
;
; For example the extension _NXX would match normal 7 digit dialings,
; while _1NXXNXX would represent an area code plus phone number
; preceeded by a one.
;
; Each step of an extension is ordered by priority, which must
; always start with 1 to be considered a valid extension.
;
; Contexts contain several lines, one for each step of each
; extension, which can take one of two forms as listed below,
; with the first form being preferred.  One may include another
; context in the current one as well, optionally with a
; date and time.  Included contexts are included in the order
; they are listed.
;
;[context]
;exten => someexten,priority,application(arg1,arg2,...)
;exten => someexten,priority,application,arg1|arg2...
;
; Timing

[Asterisk-Users] why this macro works Only with a * Stable Version? ; -(

2005-07-24 Thread xAD




Hi all, i have a weird problem, i need to know why 
this 'simple' macro works Only with a * STABLE version (1.0.0.x to 1.0.0.9) , if 
i try it with a CVS HEAD version.. don't works ;-(
 
Thanks in advance.
 
[local]
 
include => default
 
[macro-tono_simulato]exten => 
s,1,Set(DIALED=${ARG1})exten => s,2,Set(TOCONTEXT=${ARG2})exten => 
s,3,Set(STRIP=${ARG3})exten => s,4,Set(TONES=dial)  exten => 
s,5,Goto(tono_simulato_pausa|s|1)
 
[tono_simulato_pausa]exten => 
s,1,Set(TIMEOUT(digit)=0)exten => s,2,Wait(0.5)exten => 
s,3,Playtones(${TONES})exten => s,4,WaitExten(10)
 
exten => _X,1,GotoIF(${DIALED}${EXTEN} > 
0?100:2)exten => _X,2,Set(DIALED=${DIALED}${EXTEN})exten => 
_X,3,Set(TIMEOUT(digit)=0)
 
exten => 
_X,100,Set(DIALED=${DIALED}${EXTEN})exten => 
_X,101,StopPlaytonesexten => _X,102,Set(TIMEOUT(digit)=3)
 
exten => 
_X.,1,Goto(${TOCONTEXT}|$[${DIALED:${STRIP}}]${EXTEN}|1)
 
exten => t,1,Hangup
 
 
[default]
 
exten => 
00,1,Macro(tono_simulato,${EXTEN},uscita_bri,1)
 
[uscita_bri]
 
exten => 
_X.,1,Dial(Modem/ttyI1:${EXTEN},30)exten => _X.,2,Hangup
 
 
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Re: [Asterisk-Users] Why can't sip/200 call sip/202


Would this do it:

exten =>  _2XX,1,Dial(${ARG1},30)

Then I would fallback to voicemail (or something else) after the 30 seconds?

Angus



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


Sent: Sunday, July 24, 2005 10:06 PM
Subject: Re: [Asterisk-Users] Why can't sip/200 call sip/202


Ok your extensions.conf doesn't mention anything about an extension/number 
equal to 202 or 200. You must know that the name of a SIP and IAX2 peer is 
only an "address", you will have to assign a number via extensions.conf to 
this address.


Have a look at www.voip-info.org and of course google.com to get to know 
extensions.conf.


Regards,

Marc

Angus Comber wrote:
I think the 777 may be a bit of a Red Herring.  I dialed 777 as a test. 
I can't dial 202 from 200 if I actually dial 202!


My extensions.conf file:


;
; Static extension configuration file, used by
; the pbx_config module. This is where you configure all your
; inbound and outbound calls in Asterisk.
;
; This configuration file is reloaded
; - With the "extensions reload" command in the CLI
; - With the "reload" command (that reloads everything) in the CLI

;
; The "General" category is for certain variables.
;
[general]
;
; If static is set to no, or omitted, then the pbx_config will rewrite
; this file when extensions are modified.  Remember that all comments
; made in the file will be lost when that happens.
;
; XXX Not yet implemented XXX
;
static=yes
;
; if static=yes and writeprotect=no, you can save dialplan by
; CLI command 'save dialplan' too
;
writeprotect=no

; You can include other config files, use the #include command (without 
the ';')
; Note that this is different from the "include" command that includes 
contexts within
; other contexts. The #include command works in all asterisk 
configuration files.

;#include "filename.conf"

; The "Globals" category contains global variables that can be referenced
; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental 
variable

; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid
;
[globals]
CONSOLE=Console/dsp; Console interface for demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
IAXINFO=guest ; IAXtel username/password
;IAXINFO=myuser:mypass
TRUNK=Zap/g2 ; Trunk interface
;
; Note the 'g2' in the TRUNK variable above. It specifies which group 
(defined
; in zapata.conf) to dial, i.e. group 2, and how to choose a channel to 
use in

; the specified group. The four possible options are:
;
; g: select the lowest-numbered non-busy Zap channel (aka. ascending 
sequential hunt group).
; G: select the highest-numbered non-busy Zap channel (aka. descending 
sequential hunt group).
; r: use a round-robin search, starting at the next highest channel than 
last time (aka. ascending rotary hunt group).
; R: use a round-robin search, starting at the next lowest channel than 
last time (aka. descending rotary hunt group).

;
TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
;TRUNK=IAX2/user:[EMAIL PROTECTED]

;
; Any category other than "General" and "Globals" represent
; extension contexts, which are collections of extensions.
;
; Extension names may be numbers, letters, or combinations
; thereof. If an extension name is prefixed by a '_'
; character, it is interpreted as a pattern rather than a
; literal.  In patterns, some characters have special meanings:
;
;   X - any digit from 0-9
;   Z - any digit from 1-9
;   N - any digit from 2-9
;   [1235-9] - any digit in the brackets (in this example, 
1,2,3,5,6,7,8,9)

;   . - wildcard, matches anything remaining (e.g. _9011. matches
; anything starting with 9011 excluding 9011 itself)
;
; For example the extension _NXX would match normal 7 digit dialings,
; while _1NXXNXX would represent an area code plus phone number
; preceeded by a one.
;
; Each step of an extension is ordered by priority, which must
; always start with 1 to be considered a valid extension.
;
; Contexts contain several lines, one for each step of each
; extension, which can take one of two forms as listed below,
; with the first form being preferred.  One may include another
; context in the current one as well, optionally with a
; date and time.  Included contexts are included in the order
; they are listed.
;
;[context]
;exten => someexten,priority,application(arg1,arg2,...)
;exten => someexten,priority,application,arg1|arg2...
;
; Timing list for includes is
;
;   |||
;
;include => daytime|9:00-17:00|mon-fri|*|*
;
; ignorepat can be used to instruct drivers to not cancel dialtone upon
; receipt of a particular pattern.  The most commonly used example is
; of course '9' like this:
;
;ignorepat => 9
;
; so that dialtone remains even after dialing a 9.
;

;
; Here are the entries you need to participate in the IAXTEL
; call routing system.  Most IAXTEL numbers begin with 1-700, but
; there are exceptions.  For more information, and to sign
; up, please go to www.gnophone

Re: [Asterisk-Users] Why can't sip/200 call sip/202

2005-07-24 Thread Mark Edwards

PS you would be better seeing this debugged with "set verbose 5" in the CLI

regards,

Mark

On 7/25/05, Mark Edwards <[EMAIL PROTECTED]> wrote:
> OK Angus
> 
> just start here
> 
> mv extensions.conf extensions.conf.old
> 
> and create a new extensions.conf
> 
> [default]
> exten => _2XX,1,Dial(SIP/${EXTEN},20,Ttm)
> exten => _2XX,2,Hangup
> 
> 
> 
> just those 3 lines
> 
> do an 'extensions reload' in the CLI or just restart Asterisk
> 
> and see if it works
> 
> regards,
> 
> Mark.
> On 7/25/05, Angus Comber <[EMAIL PROTECTED]> wrote:
> > I think the 777 may be a bit of a Red Herring.  I dialed 777 as a test.  I
> > can't dial 202 from 200 if I actually dial 202!
> >
> > My extensions.conf file:
> >
> >
> > ;
> > ; Static extension configuration file, used by
> > ; the pbx_config module. This is where you configure all your
> > ; inbound and outbound calls in Asterisk.
> > ;
> > ; This configuration file is reloaded
> > ; - With the "extensions reload" command in the CLI
> > ; - With the "reload" command (that reloads everything) in the CLI
> >
> > ;
> > ; The "General" category is for certain variables.
> > ;
> > [general]
> > ;
> > ; If static is set to no, or omitted, then the pbx_config will rewrite
> > ; this file when extensions are modified.  Remember that all comments
> > ; made in the file will be lost when that happens.
> > ;
> > ; XXX Not yet implemented XXX
> > ;
> > static=yes
> > ;
> > ; if static=yes and writeprotect=no, you can save dialplan by
> > ; CLI command 'save dialplan' too
> > ;
> > writeprotect=no
> >
> > ; You can include other config files, use the #include command (without the
> > ';')
> > ; Note that this is different from the "include" command that includes
> > contexts within
> > ; other contexts. The #include command works in all asterisk configuration
> > files.
> > ;#include "filename.conf"
> >
> > ; The "Globals" category contains global variables that can be referenced
> > ; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental
> > variable
> > ; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid
> > ;
> > [globals]
> > CONSOLE=Console/dsp; Console interface for demo
> > ;CONSOLE=Zap/1
> > ;CONSOLE=Phone/phone0
> > IAXINFO=guest ; IAXtel username/password
> > ;IAXINFO=myuser:mypass
> > TRUNK=Zap/g2 ; Trunk interface
> > ;
> > ; Note the 'g2' in the TRUNK variable above. It specifies which group
> > (defined
> > ; in zapata.conf) to dial, i.e. group 2, and how to choose a channel to use
> > in
> > ; the specified group. The four possible options are:
> > ;
> > ; g: select the lowest-numbered non-busy Zap channel (aka. ascending
> > sequential hunt group).
> > ; G: select the highest-numbered non-busy Zap channel (aka. descending
> > sequential hunt group).
> > ; r: use a round-robin search, starting at the next highest channel than
> > last time (aka. ascending rotary hunt group).
> > ; R: use a round-robin search, starting at the next lowest channel than last
> > time (aka. descending rotary hunt group).
> > ;
> > TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
> > ;TRUNK=IAX2/user:[EMAIL PROTECTED]
> >
> > ;
> > ; Any category other than "General" and "Globals" represent
> > ; extension contexts, which are collections of extensions.
> > ;
> > ; Extension names may be numbers, letters, or combinations
> > ; thereof. If an extension name is prefixed by a '_'
> > ; character, it is interpreted as a pattern rather than a
> > ; literal.  In patterns, some characters have special meanings:
> > ;
> > ;   X - any digit from 0-9
> > ;   Z - any digit from 1-9
> > ;   N - any digit from 2-9
> > ;   [1235-9] - any digit in the brackets (in this example, 1,2,3,5,6,7,8,9)
> > ;   . - wildcard, matches anything remaining (e.g. _9011. matches
> > ; anything starting with 9011 excluding 9011 itself)
> > ;
> > ; For example the extension _NXX would match normal 7 digit dialings,
> > ; while _1NXXNXX would represent an area code plus phone number
> > ; preceeded by a one.
> > ;
> > ; Each step of an extension is ordered by priority, which must
> > ; always start with 1 to be considered a valid extension.
> > ;
> > ; Contexts contain several lines, one for each step of each
> > ; extension, which can take one of two forms as listed below,
> > ; with the first form being preferred.  One may include another
> > ; context in the current one as well, optionally with a
> > ; date and time.  Included contexts are included in the order
> > ; they are listed.
> > ;
> > ;[context]
> > ;exten => someexten,priority,application(arg1,arg2,...)
> > ;exten => someexten,priority,application,arg1|arg2...
> > ;
> > ; Timing list for includes is
> > ;
> > ;   |||
> > ;
> > ;include => daytime|9:00-17:00|mon-fri|*|*
> > ;
> > ; ignorepat can be used to instruct drivers to not cancel dialtone upon
> > ; receipt of a particular pattern.  The most commonly used example is
> > ; of course '9' like this:
> > ;
> > ;ignorepat => 9
> > ;
> > ; so that dialtone rema

Re: [Asterisk-Users] Why can't sip/200 call sip/202

2005-07-24 Thread Mark Edwards

OK Angus

just start here

mv extensions.conf extensions.conf.old

and create a new extensions.conf

[default]
exten => _2XX,1,Dial(SIP/${EXTEN},20,Ttm)
exten => _2XX,2,Hangup



just those 3 lines

do an 'extensions reload' in the CLI or just restart Asterisk

and see if it works

regards,

Mark.
On 7/25/05, Angus Comber <[EMAIL PROTECTED]> wrote:
> I think the 777 may be a bit of a Red Herring.  I dialed 777 as a test.  I
> can't dial 202 from 200 if I actually dial 202!
> 
> My extensions.conf file:
> 
> 
> ;
> ; Static extension configuration file, used by
> ; the pbx_config module. This is where you configure all your
> ; inbound and outbound calls in Asterisk.
> ;
> ; This configuration file is reloaded
> ; - With the "extensions reload" command in the CLI
> ; - With the "reload" command (that reloads everything) in the CLI
> 
> ;
> ; The "General" category is for certain variables.
> ;
> [general]
> ;
> ; If static is set to no, or omitted, then the pbx_config will rewrite
> ; this file when extensions are modified.  Remember that all comments
> ; made in the file will be lost when that happens.
> ;
> ; XXX Not yet implemented XXX
> ;
> static=yes
> ;
> ; if static=yes and writeprotect=no, you can save dialplan by
> ; CLI command 'save dialplan' too
> ;
> writeprotect=no
> 
> ; You can include other config files, use the #include command (without the
> ';')
> ; Note that this is different from the "include" command that includes
> contexts within
> ; other contexts. The #include command works in all asterisk configuration
> files.
> ;#include "filename.conf"
> 
> ; The "Globals" category contains global variables that can be referenced
> ; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental
> variable
> ; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid
> ;
> [globals]
> CONSOLE=Console/dsp; Console interface for demo
> ;CONSOLE=Zap/1
> ;CONSOLE=Phone/phone0
> IAXINFO=guest ; IAXtel username/password
> ;IAXINFO=myuser:mypass
> TRUNK=Zap/g2 ; Trunk interface
> ;
> ; Note the 'g2' in the TRUNK variable above. It specifies which group
> (defined
> ; in zapata.conf) to dial, i.e. group 2, and how to choose a channel to use
> in
> ; the specified group. The four possible options are:
> ;
> ; g: select the lowest-numbered non-busy Zap channel (aka. ascending
> sequential hunt group).
> ; G: select the highest-numbered non-busy Zap channel (aka. descending
> sequential hunt group).
> ; r: use a round-robin search, starting at the next highest channel than
> last time (aka. ascending rotary hunt group).
> ; R: use a round-robin search, starting at the next lowest channel than last
> time (aka. descending rotary hunt group).
> ;
> TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
> ;TRUNK=IAX2/user:[EMAIL PROTECTED]
> 
> ;
> ; Any category other than "General" and "Globals" represent
> ; extension contexts, which are collections of extensions.
> ;
> ; Extension names may be numbers, letters, or combinations
> ; thereof. If an extension name is prefixed by a '_'
> ; character, it is interpreted as a pattern rather than a
> ; literal.  In patterns, some characters have special meanings:
> ;
> ;   X - any digit from 0-9
> ;   Z - any digit from 1-9
> ;   N - any digit from 2-9
> ;   [1235-9] - any digit in the brackets (in this example, 1,2,3,5,6,7,8,9)
> ;   . - wildcard, matches anything remaining (e.g. _9011. matches
> ; anything starting with 9011 excluding 9011 itself)
> ;
> ; For example the extension _NXX would match normal 7 digit dialings,
> ; while _1NXXNXX would represent an area code plus phone number
> ; preceeded by a one.
> ;
> ; Each step of an extension is ordered by priority, which must
> ; always start with 1 to be considered a valid extension.
> ;
> ; Contexts contain several lines, one for each step of each
> ; extension, which can take one of two forms as listed below,
> ; with the first form being preferred.  One may include another
> ; context in the current one as well, optionally with a
> ; date and time.  Included contexts are included in the order
> ; they are listed.
> ;
> ;[context]
> ;exten => someexten,priority,application(arg1,arg2,...)
> ;exten => someexten,priority,application,arg1|arg2...
> ;
> ; Timing list for includes is
> ;
> ;   |||
> ;
> ;include => daytime|9:00-17:00|mon-fri|*|*
> ;
> ; ignorepat can be used to instruct drivers to not cancel dialtone upon
> ; receipt of a particular pattern.  The most commonly used example is
> ; of course '9' like this:
> ;
> ;ignorepat => 9
> ;
> ; so that dialtone remains even after dialing a 9.
> ;
> 
> ;
> ; Here are the entries you need to participate in the IAXTEL
> ; call routing system.  Most IAXTEL numbers begin with 1-700, but
> ; there are exceptions.  For more information, and to sign
> ; up, please go to www.gnophone.com or www.iaxtel.com
> ;
> [iaxtel700]
> exten => _91700XXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:[EMAIL 
> PROTECTED])
> 
> ;
> ; The SWITCH statement pe

Re: [Asterisk-Users] success story: TE406P (quadspan with hardwareechocan)

2005-07-24 Thread Andrew Kohlsmith

On Sunday 24 July 2005 16:30, Chris Modesitt wrote:
> Man I almost passed from laughing when I read this, that is the best
> description of bad echo I have ever heard:)

:-)  Well the bad bad echo I described I am almost positive occurs because the 
echo canceller either mistakenly turns off (false tone detect?) or settles on 
the wrong set of values and can't be bumped off of them for that call.  

-A.
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Re: [Asterisk-Users] Why can't sip/200 call sip/202

The extensions.conf file you provided looks suspiciously like the asterisk
configs/extensions.conf.sample file.

Did you create a dialplan for your specific configuration or did you just
copy the sample file?



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

Sent: Sunday, July 24, 2005 2:50 PM
Subject: Re: [Asterisk-Users] Why can't sip/200 call sip/202


> I think the 777 may be a bit of a Red Herring.  I dialed 777 as a test.  I
> can't dial 202 from 200 if I actually dial 202!
>
> My extensions.conf file:
>
>
> ;
> ; Static extension configuration file, used by
> ; the pbx_config module. This is where you configure all your
> ; inbound and outbound calls in Asterisk.
> ;
> ; This configuration file is reloaded
> ; - With the "extensions reload" command in the CLI
> ; - With the "reload" command (that reloads everything) in the CLI
>
> ;
> ; The "General" category is for certain variables.
> ;
> [general]
> ;
> ; If static is set to no, or omitted, then the pbx_config will rewrite
> ; this file when extensions are modified.  Remember that all comments
> ; made in the file will be lost when that happens.
> ;
> ; XXX Not yet implemented XXX
> ;
> static=yes
> ;
> ; if static=yes and writeprotect=no, you can save dialplan by
> ; CLI command 'save dialplan' too
> ;
> writeprotect=no
>
> ; You can include other config files, use the #include command (without
the
> ';')
> ; Note that this is different from the "include" command that includes
> contexts within
> ; other contexts. The #include command works in all asterisk configuration
> files.
> ;#include "filename.conf"
>
> ; The "Globals" category contains global variables that can be referenced
> ; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental
> variable
> ; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid
> ;
> [globals]
> CONSOLE=Console/dsp; Console interface for demo
> ;CONSOLE=Zap/1
> ;CONSOLE=Phone/phone0
> IAXINFO=guest ; IAXtel username/password
> ;IAXINFO=myuser:mypass
> TRUNK=Zap/g2 ; Trunk interface
> ;
> ; Note the 'g2' in the TRUNK variable above. It specifies which group
> (defined
> ; in zapata.conf) to dial, i.e. group 2, and how to choose a channel to
use
> in
> ; the specified group. The four possible options are:
> ;
> ; g: select the lowest-numbered non-busy Zap channel (aka. ascending
> sequential hunt group).
> ; G: select the highest-numbered non-busy Zap channel (aka. descending
> sequential hunt group).
> ; r: use a round-robin search, starting at the next highest channel than
> last time (aka. ascending rotary hunt group).
> ; R: use a round-robin search, starting at the next lowest channel than
last
> time (aka. descending rotary hunt group).
> ;
> TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
> ;TRUNK=IAX2/user:[EMAIL PROTECTED]
>
> ;
> ; Any category other than "General" and "Globals" represent
> ; extension contexts, which are collections of extensions.
> ;
> ; Extension names may be numbers, letters, or combinations
> ; thereof. If an extension name is prefixed by a '_'
> ; character, it is interpreted as a pattern rather than a
> ; literal.  In patterns, some characters have special meanings:
> ;
> ;   X - any digit from 0-9
> ;   Z - any digit from 1-9
> ;   N - any digit from 2-9
> ;   [1235-9] - any digit in the brackets (in this example,
1,2,3,5,6,7,8,9)
> ;   . - wildcard, matches anything remaining (e.g. _9011. matches
> ; anything starting with 9011 excluding 9011 itself)
> ;
> ; For example the extension _NXX would match normal 7 digit dialings,
> ; while _1NXXNXX would represent an area code plus phone number
> ; preceeded by a one.
> ;
> ; Each step of an extension is ordered by priority, which must
> ; always start with 1 to be considered a valid extension.
> ;
> ; Contexts contain several lines, one for each step of each
> ; extension, which can take one of two forms as listed below,
> ; with the first form being preferred.  One may include another
> ; context in the current one as well, optionally with a
> ; date and time.  Included contexts are included in the order
> ; they are listed.
> ;
> ;[context]
> ;exten => someexten,priority,application(arg1,arg2,...)
> ;exten => someexten,priority,application,arg1|arg2...
> ;
> ; Timing list for includes is
> ;
> ;   |||
> ;
> ;include => daytime|9:00-17:00|mon-fri|*|*
> ;
> ; ignorepat can be used to instruct drivers to not cancel dialtone upon
> ; receipt of a particular pattern.  The most commonly used example is
> ; of course '9' like this:
> ;
> ;ignorepat => 9
> ;
> ; so that dialtone remains even after dialing a 9.
> ;
>
> ;
> ; Here are the entries you need to participate in the IAXTEL
> ; call routing system.  Most IAXTEL numbers begin with 1-700, but
> ; there are exceptions.  For more information, and to sign
> ; up, please go to www.gnophone.com or www.iaxtel.com
> ;
> [iaxtel700]
> exten =>
_91700XXX,1,Dial(IAX2/[

Re: [Asterisk-Users] Why can't sip/200 call sip/202

Ok your extensions.conf doesn't mention anything about an 
extension/number equal to 202 or 200. You must know that the name of a 
SIP and IAX2 peer is only an "address", you will have to assign a number 
via extensions.conf to this address.


Have a look at www.voip-info.org and of course google.com to get to know 
extensions.conf.


Regards,

Marc

Angus Comber wrote:
I think the 777 may be a bit of a Red Herring.  I dialed 777 as a test.  
I can't dial 202 from 200 if I actually dial 202!


My extensions.conf file:


;
; Static extension configuration file, used by
; the pbx_config module. This is where you configure all your
; inbound and outbound calls in Asterisk.
;
; This configuration file is reloaded
; - With the "extensions reload" command in the CLI
; - With the "reload" command (that reloads everything) in the CLI

;
; The "General" category is for certain variables.
;
[general]
;
; If static is set to no, or omitted, then the pbx_config will rewrite
; this file when extensions are modified.  Remember that all comments
; made in the file will be lost when that happens.
;
; XXX Not yet implemented XXX
;
static=yes
;
; if static=yes and writeprotect=no, you can save dialplan by
; CLI command 'save dialplan' too
;
writeprotect=no

; You can include other config files, use the #include command (without 
the ';')
; Note that this is different from the "include" command that includes 
contexts within
; other contexts. The #include command works in all asterisk 
configuration files.

;#include "filename.conf"

; The "Globals" category contains global variables that can be referenced
; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental 
variable

; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid
;
[globals]
CONSOLE=Console/dsp; Console interface for demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
IAXINFO=guest ; IAXtel username/password
;IAXINFO=myuser:mypass
TRUNK=Zap/g2 ; Trunk interface
;
; Note the 'g2' in the TRUNK variable above. It specifies which group 
(defined
; in zapata.conf) to dial, i.e. group 2, and how to choose a channel to 
use in

; the specified group. The four possible options are:
;
; g: select the lowest-numbered non-busy Zap channel (aka. ascending 
sequential hunt group).
; G: select the highest-numbered non-busy Zap channel (aka. descending 
sequential hunt group).
; r: use a round-robin search, starting at the next highest channel than 
last time (aka. ascending rotary hunt group).
; R: use a round-robin search, starting at the next lowest channel than 
last time (aka. descending rotary hunt group).

;
TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
;TRUNK=IAX2/user:[EMAIL PROTECTED]

;
; Any category other than "General" and "Globals" represent
; extension contexts, which are collections of extensions.
;
; Extension names may be numbers, letters, or combinations
; thereof. If an extension name is prefixed by a '_'
; character, it is interpreted as a pattern rather than a
; literal.  In patterns, some characters have special meanings:
;
;   X - any digit from 0-9
;   Z - any digit from 1-9
;   N - any digit from 2-9
;   [1235-9] - any digit in the brackets (in this example, 1,2,3,5,6,7,8,9)
;   . - wildcard, matches anything remaining (e.g. _9011. matches
; anything starting with 9011 excluding 9011 itself)
;
; For example the extension _NXX would match normal 7 digit dialings,
; while _1NXXNXX would represent an area code plus phone number
; preceeded by a one.
;
; Each step of an extension is ordered by priority, which must
; always start with 1 to be considered a valid extension.
;
; Contexts contain several lines, one for each step of each
; extension, which can take one of two forms as listed below,
; with the first form being preferred.  One may include another
; context in the current one as well, optionally with a
; date and time.  Included contexts are included in the order
; they are listed.
;
;[context]
;exten => someexten,priority,application(arg1,arg2,...)
;exten => someexten,priority,application,arg1|arg2...
;
; Timing list for includes is
;
;   |||
;
;include => daytime|9:00-17:00|mon-fri|*|*
;
; ignorepat can be used to instruct drivers to not cancel dialtone upon
; receipt of a particular pattern.  The most commonly used example is
; of course '9' like this:
;
;ignorepat => 9
;
; so that dialtone remains even after dialing a 9.
;

;
; Here are the entries you need to participate in the IAXTEL
; call routing system.  Most IAXTEL numbers begin with 1-700, but
; there are exceptions.  For more information, and to sign
; up, please go to www.gnophone.com or www.iaxtel.com
;
[iaxtel700]
exten => _91700XXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:[EMAIL PROTECTED])

;
; The SWITCH statement permits a server to share the dialplain with
; another server. Use with care: Reciprocal switch statements are not
; allowed (e.g. both A -> B and B -> A), and the switched server needs
; to be on-line or else dialing can be severly de

Re: [Asterisk-Users] Help with [EMAIL PROTECTED] andBroadvoiceincomingcalls..

Ok.. I screwed up...

You have a register statement:
[EMAIL PROTECTED]:123abc:[EMAIL PROTECTED]/
201

so, the incoming call from broadvoice will be sent to extension 201 in the
frombroadvoice context.

To ensure what is going on, use this as your context.
exten => _X.,1,Noop(Incoming call for extension ${EXTEN} in context
frombroadvoice)

That will tell you exactly what is being sent into the context.

You are using the latest AAH, so the variable substitutions will work.

I expect that you will end up using the following in frombroadvoice:

exten => ${BVRINGS},1,Macro(exten-vm,[EMAIL PROTECTED],${BVRINGS})

Sorry for the confusion

Regards,
Derek


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

Sent: Sunday, July 24, 2005 2:27 PM
Subject: RE: [Asterisk-Users] Help with [EMAIL PROTECTED]
andBroadvoiceincomingcalls..


>
> OK, I think I understood what you were saying, but let me type this in
here as
> like I said I am for sure trying to figure this sucker out still..
>
> I just tried the following:
>
> exten => s,1,Macro(exten-vm,[EMAIL PROTECTED],${BVRINGS})
>
> I also tried this:
>
> exten => 240524,1,Macro(exten-vm,[EMAIL PROTECTED],${BVRINGS})
>
>
> Of course I told it to reload the configs, and I get the exact same
reaction
> from it when I call the number.
>
> Is there any other type of debugging or output that would be helpful?
>
> Also funny you mention that the variable wouldn't work on the extension,
as
> maybe it's done a little different, but I have this for my FreeWorld IAX
> connection and incoming calls on it work great.
>
> [fromiaxfwd]
> exten => ${FWDNUMBER},1,Macro(exten-vm,[EMAIL PROTECTED],${FWDRINGS})
> exten => ${VM_PREFIX}${FWDVMBOX},1,Macro(vm,${FWDVMBOX})
>
> With the various variables for FWD set up top.   Not sure about what is in
the
> CVSHEAD, I haven't gotten good enough to try that out yet, but I do have
the
> most current [EMAIL PROTECTED], which is using asterisk 1.0.9 at this time.
>
> Anyway still very confused, and hopefully you will have some ideas.
>
>
> ---
> Howard Leadmon - howard at leadmon.net
> http://www.leadmon.net
>
>
> > -Original Message-
> > From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > [EMAIL PROTECTED] On Behalf Of dbruce
> > Sent: Sunday, July 24, 2005 4:08 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] Help with [EMAIL PROTECTED] and
> > Broadvoiceincomingcalls..
> >
> > Your [frombroadvoice] context is incorrect. You have set a global
variable
> > BVNUMBER and used it as the extension match in the context. The problem
is
> > that the extension match syntax does not support variable substitution
> > unless you are using a relatively current CVS HEAD. As [EMAIL PROTECTED] is
> > based on CVS STABLE, you can't use variable substitution.
> >
> > You will need to replace the ${BVNUMBER} with valid extension match
syntax.
> > You can use the 's' extension or a general match patern '_X." and do the
> > specific matching within the dialplan to determine is you wish to accept
the
> > call ie: gotoif($["${BVNUMBER}" = "${EXTEN}"]?x) (replacing 'x' with a
valid
> > priority).
> >
> > Regards,
> > Derek
> >
> > - Original Message -
> > From: "Howard Leadmon" <[EMAIL PROTECTED]>
> > To: 
> > Sent: Sunday, July 24, 2005 1:34 PM
> > Subject: [Asterisk-Users] Help with [EMAIL PROTECTED] and Broadvoice
> > incomingcalls..
> >
> >
> > >
> > >Hello everyone,
> > >
> > >  Well here is my initial posting to the list, and I will admit
Asterisk is
> > new
> > > to me. I just got everything running here a couple days ago, so still
> > learning
> > > the ropes for sure.
> > >
> > >  OK, here is my problem.   Currently I have it setup talking to a
couple
> > Cisco
> > > IP phones, and some Xten softphones, this works great.   I also got an
> > account
> > > with FreeWorld Dialup using IAX2 and that works super both inbound and
> > > outbound at this time.   I decided to sign up with BroadVoice as they
had
> > good
> > > pricing, seems like well supported in the Asterisk community.
> > >
> > >  So when I setup with BroadVoice I got the outgoing calls to them
working
> > just
> > > fine, I set it up so I can dial 8, and then any number I desire to
reach
> > and
> > > the call goes through.   Now as simple as I thought this would be, if
I
> > try
> > > and get an incoming call, it just doesn't work, I think it rolls right
> > into
> > > the BroadVoice Vmail they provide, as nothing rings here, so figure
> > something
> > > is messed up in the call pathway.
> > >
> > >  I have spend hours looking at the debug output, and though some of it
> > makes
> > > good sense, I am just to green to really dig into the guts of this
sucker
> > yet,
> > > hopefully that will change for me soon.  So I hope someone here on the
> > list
> > > can help me figure out what the heck is wrong with this, and get my
> > incoming

Re: [Asterisk-Users] Stupid hold music

2005-07-24 Thread Steve Gladden

Oh GOSH
That was awesome!

I have an even better one in store but gotta capture it from it's
rather obscure source..

Stay tuned!!!

Steve  (N8LBV)











>> Does anyone have a collection of stupid hold music? Y'know, the sort of
>> thing that would drive a person mad? Silly songs, repetative tunes etc?
>
> You should be flogged pubically for bringing up this subject - that
> last "space people" song almost made me wash out my ears with
> sulphuric acid!!!
>
> 73 de NY5I
> Hatton
> ___
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Re: [Asterisk-Users] Why can't sip/200 call sip/202

I think the 777 may be a bit of a Red Herring.  I dialed 777 as a test.  I 
can't dial 202 from 200 if I actually dial 202!


My extensions.conf file:


;
; Static extension configuration file, used by
; the pbx_config module. This is where you configure all your
; inbound and outbound calls in Asterisk.
;
; This configuration file is reloaded
; - With the "extensions reload" command in the CLI
; - With the "reload" command (that reloads everything) in the CLI

;
; The "General" category is for certain variables.
;
[general]
;
; If static is set to no, or omitted, then the pbx_config will rewrite
; this file when extensions are modified.  Remember that all comments
; made in the file will be lost when that happens.
;
; XXX Not yet implemented XXX
;
static=yes
;
; if static=yes and writeprotect=no, you can save dialplan by
; CLI command 'save dialplan' too
;
writeprotect=no

; You can include other config files, use the #include command (without the 
';')
; Note that this is different from the "include" command that includes 
contexts within
; other contexts. The #include command works in all asterisk configuration 
files.

;#include "filename.conf"

; The "Globals" category contains global variables that can be referenced
; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental 
variable

; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid
;
[globals]
CONSOLE=Console/dsp; Console interface for demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
IAXINFO=guest ; IAXtel username/password
;IAXINFO=myuser:mypass
TRUNK=Zap/g2 ; Trunk interface
;
; Note the 'g2' in the TRUNK variable above. It specifies which group 
(defined
; in zapata.conf) to dial, i.e. group 2, and how to choose a channel to use 
in

; the specified group. The four possible options are:
;
; g: select the lowest-numbered non-busy Zap channel (aka. ascending 
sequential hunt group).
; G: select the highest-numbered non-busy Zap channel (aka. descending 
sequential hunt group).
; r: use a round-robin search, starting at the next highest channel than 
last time (aka. ascending rotary hunt group).
; R: use a round-robin search, starting at the next lowest channel than last 
time (aka. descending rotary hunt group).

;
TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
;TRUNK=IAX2/user:[EMAIL PROTECTED]

;
; Any category other than "General" and "Globals" represent
; extension contexts, which are collections of extensions.
;
; Extension names may be numbers, letters, or combinations
; thereof. If an extension name is prefixed by a '_'
; character, it is interpreted as a pattern rather than a
; literal.  In patterns, some characters have special meanings:
;
;   X - any digit from 0-9
;   Z - any digit from 1-9
;   N - any digit from 2-9
;   [1235-9] - any digit in the brackets (in this example, 1,2,3,5,6,7,8,9)
;   . - wildcard, matches anything remaining (e.g. _9011. matches
; anything starting with 9011 excluding 9011 itself)
;
; For example the extension _NXX would match normal 7 digit dialings,
; while _1NXXNXX would represent an area code plus phone number
; preceeded by a one.
;
; Each step of an extension is ordered by priority, which must
; always start with 1 to be considered a valid extension.
;
; Contexts contain several lines, one for each step of each
; extension, which can take one of two forms as listed below,
; with the first form being preferred.  One may include another
; context in the current one as well, optionally with a
; date and time.  Included contexts are included in the order
; they are listed.
;
;[context]
;exten => someexten,priority,application(arg1,arg2,...)
;exten => someexten,priority,application,arg1|arg2...
;
; Timing list for includes is
;
;   |||
;
;include => daytime|9:00-17:00|mon-fri|*|*
;
; ignorepat can be used to instruct drivers to not cancel dialtone upon
; receipt of a particular pattern.  The most commonly used example is
; of course '9' like this:
;
;ignorepat => 9
;
; so that dialtone remains even after dialing a 9.
;

;
; Here are the entries you need to participate in the IAXTEL
; call routing system.  Most IAXTEL numbers begin with 1-700, but
; there are exceptions.  For more information, and to sign
; up, please go to www.gnophone.com or www.iaxtel.com
;
[iaxtel700]
exten => _91700XXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:[EMAIL PROTECTED])

;
; The SWITCH statement permits a server to share the dialplain with
; another server. Use with care: Reciprocal switch statements are not
; allowed (e.g. both A -> B and B -> A), and the switched server needs
; to be on-line or else dialing can be severly delayed.
;
[iaxprovider]
;switch => IAX2/user:[EMAIL PROTECTED]/mycontext

[trunkint]
;
; International long distance through trunk
;
exten => _9011.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _9011.,2,Congestion

[trunkld]
;
; Long distance context accessed through trunk
;
exten => _91NXXNXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91NXXNXX,2,Congestion

Re: [Asterisk-Users] Business Edition

2005-07-24 Thread Brian McManus

Not to continue or feed any flame wars, because flame wars and holy
wars only create hurt feelings.  However, this is my personal
experience:

To the contrary I have found Digiums support to be exceptional. 
They have helped support me with several minor and small problems, and
all I did was buy hardware.  In fact, when my soft switch was
sending calling id and called id on my PRI out of spec, Matt was quick
to write a patch to get me back up and running.  This hardly fell
in their realm of things they are required to support.  If you
receive support less then par the one person you are speaking to may be
having a bad day.  It's human nature and I have to work with my
callcenter manager on this same problem with her employees.  Ask
to speak to someone else!  I always try to find a specific person
I like and I work well with in situations like this, and I never have
another problem.

Once in 2003 when needing support for my new single port t1 span, Mark
answered the support line.  How cool is it that a company is so
dedicated to support that when a CEOs technicians get busy, he helps on
the phones.

I feel it is very important to have a Business Edition, and Digium is
the only company with the ability to provide this within the letter of
the law.  MySQL does this very thing with a supported version of
MySQL, it's called "MySQL Network."  Even though MySQL has wanted
in on the support front by providing a "Business Edition," they have
continued their open source stable releases in a timely manner as well,
I love 4.1.

So as far as digium being less likely to allow for new stable versions,
I highly doubt it.  The great thing is, you know, and everyone
else knows that if Digium
ceased supporting the open source version, the open source release
would fork off in to other versions as long as we followed the existing
open
source license. Besides, for the most part CVS is managed by people
that do not directly work with them, and stable releases are decided by
mostly developers in the community.

Large companies enjoy the reassurance, and $995 is a very small capital
expenditure for a 16 million dollar a year company (the company i work
for)  that wants to enjoy the reassurance of technical
support.  The closed source nature of it is here, nor there. 
If you want the source to CVS HEAD or STABLE, check it out.  For
them to *really* support a product *really* well  they have to
certify a product has X code, Y features, and works on Z hardware and
you haven't added any odd random problems to the equation and nature of
the business edition through patches you brewed.

Mark has always been quick to do the right thing for the community as
well.   In fact, talks of Asterisk 1.2 at Astricon have been
rumoured, and I know we will continue to get a very great and valuable
stable branch..  This is a symbiotic relationship.  Digium
and all of you great developers provide an excellent tree of
Asterisk.  The community enjoys the best most flexible PBX
ever-to-date, and Digium makes money, saves our companies money, and
provide an outlet to make an exponential amount of money in the VoIP
arena...  I have a feeling digium is not interested in breaking
this wonderful symbiotic relationship by burning its customers.

So I prefer to feed my Digium friends who work the oddest long hard
hours bringing us very cool products, and get a little extra added
support.  If you disagree, check out a copy of CVS/HEAD or
CVS/STABLE, and get community support from the mailing lists =)

Your friend, 
Brian McManusOn 7/18/05, Kevin Walsh <[EMAIL PROTECTED]> wrote:
KRTorio [[EMAIL PROTECTED]] wrote:> For $995+ (including support), a technical manual, and scripts, is it> worth switching to the business edition?>Absolutely not.  If you find that you need $995 worth of support, some
time in the future, then I'm sure that you can obtain it from one ofseveral providers.  I don't think it's worth paying up-front forsomething you probably won't need, but that's really for you to decide.
If you really want to pay $995 for a closed source product with "somefeatures removed and license control added," then go for it.  It'syour money.--  
_/  
_/  _/_/_/_/  _/_/  _/_/_/  _/_/  _/_/_/  
_/_/  _/_/_/_/_/  _/  
K e v i n   W a l s h _/
_/_/  _/
_/
_/_/  _/_/[EMAIL PROTECTED]_/  
_/  _/_/_/_/  _/_/_/_/  _/_/___Asterisk-Users mailing listAsterisk-Users@lists.digium.com
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RE: [Asterisk-Users] success story: TE406P (quadspan with hardwareechocan)

2005-07-24 Thread Chris Modesitt

Man I almost passed from laughing when I read this, that is the best
description of bad echo I have ever heard:)

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Sunday, July 24, 2005 12:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] success story: TE406P (quadspan with
hardwareechocan)

I just wanted to post here and let everyone know that the TE406P (quadspan 
T1/E1 with hardware echo can) kicks some serious ass.

We've been running a PRI now for over a year with Asterisk (every single
call 
in and out is through two Asterisk boxes, including faxes) and while the 
software based echo cancellation is more than adequate, we'd get the 
occassional "edgy" echo and very infrequently get full-out "holy shit" echo.

So far the TE406 has eliminated that entirely.

Anyway as I said I just wanted to post here and tell the world that at least

as far as I have been able to determine, the extra cost of the hardware echo

can is *well* worth the money.

-A.
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RE: [Asterisk-Users] Help with [EMAIL PROTECTED] and Broadvoiceincomingcalls..


OK, I think I understood what you were saying, but let me type this in here as
like I said I am for sure trying to figure this sucker out still..

I just tried the following:

exten => s,1,Macro(exten-vm,[EMAIL PROTECTED],${BVRINGS})

I also tried this:

exten => 240524,1,Macro(exten-vm,[EMAIL PROTECTED],${BVRINGS})


Of course I told it to reload the configs, and I get the exact same reaction
from it when I call the number. 

Is there any other type of debugging or output that would be helpful?

Also funny you mention that the variable wouldn't work on the extension, as
maybe it's done a little different, but I have this for my FreeWorld IAX
connection and incoming calls on it work great.

[fromiaxfwd]
exten => ${FWDNUMBER},1,Macro(exten-vm,[EMAIL PROTECTED],${FWDRINGS})
exten => ${VM_PREFIX}${FWDVMBOX},1,Macro(vm,${FWDVMBOX})

With the various variables for FWD set up top.   Not sure about what is in the
CVSHEAD, I haven't gotten good enough to try that out yet, but I do have the
most current [EMAIL PROTECTED], which is using asterisk 1.0.9 at this time.

Anyway still very confused, and hopefully you will have some ideas.


---
Howard Leadmon - howard at leadmon.net
http://www.leadmon.net 


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of dbruce
> Sent: Sunday, July 24, 2005 4:08 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Help with [EMAIL PROTECTED] and
> Broadvoiceincomingcalls..
> 
> Your [frombroadvoice] context is incorrect. You have set a global variable
> BVNUMBER and used it as the extension match in the context. The problem is
> that the extension match syntax does not support variable substitution
> unless you are using a relatively current CVS HEAD. As [EMAIL PROTECTED] is
> based on CVS STABLE, you can't use variable substitution.
> 
> You will need to replace the ${BVNUMBER} with valid extension match syntax.
> You can use the 's' extension or a general match patern '_X." and do the
> specific matching within the dialplan to determine is you wish to accept the
> call ie: gotoif($["${BVNUMBER}" = "${EXTEN}"]?x) (replacing 'x' with a valid
> priority).
> 
> Regards,
> Derek
> 
> - Original Message -
> From: "Howard Leadmon" <[EMAIL PROTECTED]>
> To: 
> Sent: Sunday, July 24, 2005 1:34 PM
> Subject: [Asterisk-Users] Help with [EMAIL PROTECTED] and Broadvoice
> incomingcalls..
> 
> 
> >
> >Hello everyone,
> >
> >  Well here is my initial posting to the list, and I will admit Asterisk is
> new
> > to me. I just got everything running here a couple days ago, so still
> learning
> > the ropes for sure.
> >
> >  OK, here is my problem.   Currently I have it setup talking to a couple
> Cisco
> > IP phones, and some Xten softphones, this works great.   I also got an
> account
> > with FreeWorld Dialup using IAX2 and that works super both inbound and
> > outbound at this time.   I decided to sign up with BroadVoice as they had
> good
> > pricing, seems like well supported in the Asterisk community.
> >
> >  So when I setup with BroadVoice I got the outgoing calls to them working
> just
> > fine, I set it up so I can dial 8, and then any number I desire to reach
> and
> > the call goes through.   Now as simple as I thought this would be, if I
> try
> > and get an incoming call, it just doesn't work, I think it rolls right
> into
> > the BroadVoice Vmail they provide, as nothing rings here, so figure
> something
> > is messed up in the call pathway.
> >
> >  I have spend hours looking at the debug output, and though some of it
> makes
> > good sense, I am just to green to really dig into the guts of this sucker
> yet,
> > hopefully that will change for me soon.  So I hope someone here on the
> list
> > can help me figure out what the heck is wrong with this, and get my
> incoming
> > calls from BroadVoice and get this sucker working.
> >
> >  I am not sure what all information is needed, but I'll post some bits of
> > output below (with numbers changed), so maybe it will give someone a
> chance to
> > help me with this.
> >
> >
> >
> > In my sip.conf I have:
> >
> >
> [EMAIL PROTECTED]:123abc:[EMAIL PROTECTED]/
> 20
> > 1
> >
> > [sip.broadvoice.com]
> > type=peer
> > user=phone
> > host=sip.broadvoice.com
> > fromdomain=sip.broadvoice.com
> > fromuser=240524
> > secret=123abc
> > username=240524
> > insecure=very
> > context=frombroadvoice
> > authname=240524
> > dtmfmode=inband
> > dtmf=inband
> >
> >
> >
> >
> >
> > In my extensions.conf I have:
> >
> > ;setup SIP extension for BroadVoice
> > [globals]
> > BVNUMBER=240524 ; your calling number
> > BVRINGS=201 ; the phone to ring
> > BVVMBOX=201 ; the VM box for this user
> >
> >
> > [outrt-003-BroadVoice]
> > include => outrt-003-BroadVoice-custom
> > exten => _8.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30)
> > ;exten => _8.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30)
> > exten => _8.,2,Congestion()
> > exten

[Asterisk-Users] FS: Zhone Channel Bank

2005-07-24 Thread Daniel

Hi,

I have a mint, like-new (used only once) Zhone Z-Plex channel bank for
sale.  It has 16 FXS and 8 FXO (part number: Z-PLEX-10-24S/O).  It
functions as a channel bank, router, and CSU/DSU, combining T1, analog
voice, and data.  D4, SF, and ESF line formats are supported with AMI
or B8ZS coding.  It has dual T1 jacks, one ethernet jack, one serial
port, one V.35 port, and 24 analog pairs available via 50-pin Amphenol
connector (I'll also include two Leviton 41600-I break-out boxes to
separate the Amphenol 50-pin connector to 25 individual RJ-11
connectors---these are worth around $25 themselves).  Further it comes
with AC power cable and serial cable.

The Zhone Z-Plex is known to work with Asterisk (search mailing lists
for confirmation), though it was a reputation for a more difficult
setup than the much more expensive channel banks such as the CAC, etc.
(but then again those cost $600-$1000).  Supposedly, it runs fine once
configured, though I never used it with Asterisk.

Anyway, this ZPlex has the original box with styrofoam.  I thought I'd
offer it to the asterisk list before putting it on eBay.

I'm pricing it at $130 + $25 s/h (UPS Ground with Insurance).  Payment
via PayPal only (I can provide eBay username to confirm my validity and
eBay 100% reputation).  Email me off-list with questions, please, or if
you need photos.  Fully guaranteed against DOA (Dead-On-Arrival).

Thanks,
Daniel


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Re: [Asterisk-Users] Help with [EMAIL PROTECTED] and Broadvoice incomingcalls..

Your [frombroadvoice] context is incorrect. You have set a global variable
BVNUMBER and used it as the extension match in the context. The problem is
that the extension match syntax does not support variable substitution
unless you are using a relatively current CVS HEAD. As [EMAIL PROTECTED] is
based on CVS STABLE, you can't use variable substitution.

You will need to replace the ${BVNUMBER} with valid extension match syntax.
You can use the 's' extension or a general match patern '_X." and do the
specific matching within the dialplan to determine is you wish to accept the
call ie: gotoif($["${BVNUMBER}" = "${EXTEN}"]?x) (replacing 'x' with a valid
priority).

Regards,
Derek

- Original Message -
From: "Howard Leadmon" <[EMAIL PROTECTED]>
To: 
Sent: Sunday, July 24, 2005 1:34 PM
Subject: [Asterisk-Users] Help with [EMAIL PROTECTED] and Broadvoice
incomingcalls..


>
>Hello everyone,
>
>  Well here is my initial posting to the list, and I will admit Asterisk is
new
> to me. I just got everything running here a couple days ago, so still
learning
> the ropes for sure.
>
>  OK, here is my problem.   Currently I have it setup talking to a couple
Cisco
> IP phones, and some Xten softphones, this works great.   I also got an
account
> with FreeWorld Dialup using IAX2 and that works super both inbound and
> outbound at this time.   I decided to sign up with BroadVoice as they had
good
> pricing, seems like well supported in the Asterisk community.
>
>  So when I setup with BroadVoice I got the outgoing calls to them working
just
> fine, I set it up so I can dial 8, and then any number I desire to reach
and
> the call goes through.   Now as simple as I thought this would be, if I
try
> and get an incoming call, it just doesn't work, I think it rolls right
into
> the BroadVoice Vmail they provide, as nothing rings here, so figure
something
> is messed up in the call pathway.
>
>  I have spend hours looking at the debug output, and though some of it
makes
> good sense, I am just to green to really dig into the guts of this sucker
yet,
> hopefully that will change for me soon.  So I hope someone here on the
list
> can help me figure out what the heck is wrong with this, and get my
incoming
> calls from BroadVoice and get this sucker working.
>
>  I am not sure what all information is needed, but I'll post some bits of
> output below (with numbers changed), so maybe it will give someone a
chance to
> help me with this.
>
>
>
> In my sip.conf I have:
>
>
[EMAIL PROTECTED]:123abc:[EMAIL PROTECTED]/
20
> 1
>
> [sip.broadvoice.com]
> type=peer
> user=phone
> host=sip.broadvoice.com
> fromdomain=sip.broadvoice.com
> fromuser=240524
> secret=123abc
> username=240524
> insecure=very
> context=frombroadvoice
> authname=240524
> dtmfmode=inband
> dtmf=inband
>
>
>
>
>
> In my extensions.conf I have:
>
> ;setup SIP extension for BroadVoice
> [globals]
> BVNUMBER=240524 ; your calling number
> BVRINGS=201 ; the phone to ring
> BVVMBOX=201 ; the VM box for this user
>
>
> [outrt-003-BroadVoice]
> include => outrt-003-BroadVoice-custom
> exten => _8.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30)
> ;exten => _8.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30)
> exten => _8.,2,Congestion()
> exten => _8.,102,Busy()
>
> [frombroadvoice]
> exten => ${BVNUMBER},1,Macro(exten-vm,[EMAIL PROTECTED],${BVRINGS})
> exten => ${VM_PREFIX}${BVVMBOX},1,Macro(vm,${BVVMBOX})
>
>
>
>
> If I look at my normal log output when trying to call in, I see:
>
> Jul 24 15:23:12 DEBUG[1078]: Setting NAT on RTP to 0
> Jul 24 15:23:12 DEBUG[1078]: Check for res for 240524
> Jul 24 15:23:12 DEBUG[1078]: 240524 is not a local user
> Jul 24 15:23:12 DEBUG[1078]: 240524 is not a local user
> Jul 24 15:23:12 DEBUG[1078]: Stopping retransmission on
> 'SD28c9b01-2d5e97b21c9e4e488ce05aeda05558a8-js11002' of Response
623264158:
> Found
>
>
>
>
>
> Now I figured I would turn on 'sip debug' to which I see a lot more, here
is
> some of that output:
>
> Jul 24 15:24:33 VERBOSE[1078]:
>
> Sip read:
> INVITE sip:[EMAIL PROTECTED] SIP/2.0
> Via: SIP/2.0/UDP 147.135.0.128:5060;branch=z9hG4bK2qo3b3307oi0j7onc400.1sr
> From: "Fork
>
MD";tag=SD2o51f01-520831772-1122233
07
> 3802
> To: "Howard Leadmon"
> Call-ID: SD2o51f01-62b2c5360881c20fae16de626946fdf0-js11002
> CSeq: 623304774 INVITE
> Contact:

> Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,UPDATE,NOTIFY
> Supported: 100rel
> Accept: application/sdp,application/dtmf
> Max-Forwards: 69
> Content-Type: application/sdp
> Content-Length: 276
>
> v=0
> o=BroadWorks 24463992 1 IN IP4 147.135.0.128
> s=-
> c=IN IP4 147.135.0.128
> t=0 0
> m=audio 14942 RTP/AVP 0 8 2 18 96 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:2 G726-32/8000
> a=rtpmap:18 G729/8000
> a=rtpmap:96 iLBC/8000
> a=rtpmap:101 telephone-event/8000
>
> Jul 24 15:24:33 VERBOSE[1078]: 13 headers, 12 lines
> Jul 24 15:24:33 VERBOSE[1078]: Using latest request as basis request
> Jul 24 15:24:33 VERBOSE[1078]: Sending t

[Asterisk-Users] TNT and SIP problem

2005-07-24 Thread Dave Weis



I'm trying to get inbound calls from a TNT working but get 407 errors from 
the TNT. This is what I have in sip.conf:


[maxtnt]
type=friend
host=x.x.x.x
dtmfmode=rfc2833
callerid="MaxTNT" 
context=demo
qualify=yes
disallow=all
allow=g729
allow=ulaw
insecure=very

This is what the TNT is spitting out:

Jul 24 14:55:12 tnt1 1/17: Releasing <[EMAIL PROTECTED]>: 
Calling = 201,Called = 2700674, Q850 Cause = 21,Sip Response = 407 (Proxy 
Authentication Required),Progress Cause = NONE
Jul 24 14:55:12 tnt1 1/2: [1/2/11/0] STOP: ''; cause 821.; progress 1407.; 
host 0.0.0.0 [MBID 11; 201->2700674]
Jul 24 14:55:12 tnt1 1/1: [1/1/3/1] Far End Hung Up, External cause code 
021


I just have a T1 port from the asterisk machine cabled to the TNT with a 
T1 crossover trying to send calls out of the asterisk machine via T1 and 
back in via SIP until the PRI's are turned up.


dave

--
Dave Weis "I believe there are more instances of the abridgment
[EMAIL PROTECTED]   of the freedom of the people by gradual and silent
  encroachments of those in power than by violent
  and sudden usurpations."- James Madison
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Re: [Asterisk-Users] Semi-Ot - Cisco IP Phone Password Reset Procedure




The easiest way to reset a Cisco phone is via tftp
hands down
Download the image from Cisco (I can get it for you if you like.
And re provision it via TFTP
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Re: [Asterisk-Users] Why can't sip/200 call sip/202

Marc: My answer is not incorrect... it is incomplete.

The OP stipulated 2 extensions 200 and 202... and provided a sip debug
indicating a call from 200 to 777.

I pointed out the obvious.

If the OP is dialing 202 on the phone, and the phone is dialing 777, then he
needs to look at the dialplan configuration of the phone. If he is dialing
777 on the phone and expecting to reach 202, then he will need to have
translations in the asterisk dialplan. But, the question was "what should I
be looking at?"... Using just the information provided, and the fact that he
is new to asterisk... without any further information... the first thing he
should be looking at is why the phone is trying to reach 777 when he wants
to reach 202... Many new users do not realize the complexity of the SIP
protocol, and only really look at the trace in a general manner...  such as:
INVITE
407 Proxy Authentication Required
ACK
INVITE
404 Not Found
ACK

The idea was to provide a clue... not to provide a complete working dialplan
and phone configuration. Providing new users with "the complete package" is
a dis-service to them. They will only learn from thier mistakes and
experiences.. providing clues allows them to expand their experience and
build their confidence... It requires them to look at the details and learn
to analyse them.

Regards,
Derek


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

Sent: Sunday, July 24, 2005 12:53 PM
Subject: Re: [Asterisk-Users] Why can't sip/200 call sip/202


> Derek: you reply is uncorrect. If Angus has the extension 777 in his
> dialplan/extensions.conf which will dial 202. The name of the peer has
> absolutely nothing to do with which number/name he would have to dial.
> Without dialplan he will be unable to call any extension even 202, as
> 202 is only the name of the peer.
>
> Angus: please paste your extensions.conf to pastebin.ca
>
> Regards,
>
> Marc
>
> dbruce wrote:
> > It appears from the debug that extension 200 is trying to call 777, not
> > 202. Your Asterisk server can't find an extension 777 and returns "404
> > not found". That will explain why you can't call extension 777 from
> > extension 200. If you want to call extension 202, you will need to dial
> > 202 on extension 200, not 777.
> >
> > Regards,
> > Derek
> >
> >
> > - Original Message -
> > *From:* Angus Comber 
> > *To:* asterisk-users@lists.digium.com
> > 
> > *Sent:* Sunday, July 24, 2005 11:51 AM
> > *Subject:* [Asterisk-Users] Why can't sip/200 call sip/202
> >
> > I have 2 sip accounts setup - 200 and 202.  If I do sip show peers I
> > get:
> >
> > sip show peers
> > Name/usernameHostDyn Nat ACL Mask
> > Port Status
> > 202/202  192.168.0.6  D  255.255.255.255
> > 5060 Unmonitored
> > 201/201  (Unspecified)D  255.255.255.255
> > 5060 Unmonitored
> > 200/200  192.168.0.3  D  255.255.255.255
> > 5060 Unmonitored
> >
> > 200 is a Grandstream GXP200 IP Phone and 202 is a Grandstream BT100
> > IP phone.
> >
> > relevant bit of sip.conf:
> >
> > [200]
> > username=200
> > type=friend
> > secret=1234
> > port=5060
> > nat=never
> > dtmfmode=rfc2833
> > context=default
> > callerid="Angus Comber" <200>
> > host=dynamic
> > disallow=all
> > allow=ulaw
> > allow=alaw
> > allow=g723.1
> > allow=g729
> >
> > [202]
> > username=202
> > type=friend
> > secret=1234
> > port=5060
> > nat=never
> > dtmfmode=rfc2833
> > context=default
> > callerid="Sam Comber" <202>
> > host=dynamic
> > disallow=all
> > allow=ulaw
> > allow=alaw
> > allow=g723.1
> > allow=g729
> >
> >
> > But whenever I try to dial between phones I get this:
> >
> >
> > Sip read:
> >
> > 0 headers, 0 lines
> >
> >
> > Sip read:
> > INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0
> > Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKa6cf8b6a7c7198a1
> > From: "Angus Comber"
> > ;tag=a1afaf4fdb0ac845
> > To: 
> > Contact: 
> > Supported: replaces, timer
> > Call-ID: [EMAIL PROTECTED]
> > 
> > CSeq: 45925 INVITE
> > User-Agent: Grandstream GXP2000 1.0.1.9
> > Max-Forwards: 70
> > Allow:
> >
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
> > Content-Type: application/sdp
> > Content-Length: 258
> >
> > v=0
> > o=200 8000 8000 IN IP4 192.168.0.3
> > s=SIP Call
> > c=IN IP4 192.168.0.3
> > t=0 0
> > m=audio 5004 RTP/AVP 18 0 8 101
> > a=sendrecv
> > a=rtpmap:18 G729/8000
> > a=rtpmap:0 PCMU/8000
> > a=rtpmap:8 PCMA/8000
> > a=ptime:20
> > a=rtpmap:101 telephone-event/80

[Asterisk-Users] Help with [EMAIL PROTECTED] and Broadvoice incoming calls..


   Hello everyone,

 Well here is my initial posting to the list, and I will admit Asterisk is new
to me. I just got everything running here a couple days ago, so still learning
the ropes for sure.

 OK, here is my problem.   Currently I have it setup talking to a couple Cisco
IP phones, and some Xten softphones, this works great.   I also got an account
with FreeWorld Dialup using IAX2 and that works super both inbound and
outbound at this time.   I decided to sign up with BroadVoice as they had good
pricing, seems like well supported in the Asterisk community.  

 So when I setup with BroadVoice I got the outgoing calls to them working just
fine, I set it up so I can dial 8, and then any number I desire to reach and
the call goes through.   Now as simple as I thought this would be, if I try
and get an incoming call, it just doesn't work, I think it rolls right into
the BroadVoice Vmail they provide, as nothing rings here, so figure something
is messed up in the call pathway.

 I have spend hours looking at the debug output, and though some of it makes
good sense, I am just to green to really dig into the guts of this sucker yet,
hopefully that will change for me soon.  So I hope someone here on the list
can help me figure out what the heck is wrong with this, and get my incoming
calls from BroadVoice and get this sucker working.

 I am not sure what all information is needed, but I'll post some bits of
output below (with numbers changed), so maybe it will give someone a chance to
help me with this.



In my sip.conf I have:

[EMAIL PROTECTED]:123abc:[EMAIL PROTECTED]/20
1

[sip.broadvoice.com]
type=peer
user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=240524
secret=123abc
username=240524
insecure=very
context=frombroadvoice
authname=240524
dtmfmode=inband
dtmf=inband 
   




In my extensions.conf I have:

;setup SIP extension for BroadVoice
[globals]
BVNUMBER=240524 ; your calling number 
BVRINGS=201 ; the phone to ring
BVVMBOX=201 ; the VM box for this user


[outrt-003-BroadVoice]
include => outrt-003-BroadVoice-custom
exten => _8.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30)
;exten => _8.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30)
exten => _8.,2,Congestion()
exten => _8.,102,Busy()

[frombroadvoice] 
exten => ${BVNUMBER},1,Macro(exten-vm,[EMAIL PROTECTED],${BVRINGS})
exten => ${VM_PREFIX}${BVVMBOX},1,Macro(vm,${BVVMBOX})




If I look at my normal log output when trying to call in, I see:

Jul 24 15:23:12 DEBUG[1078]: Setting NAT on RTP to 0
Jul 24 15:23:12 DEBUG[1078]: Check for res for 240524
Jul 24 15:23:12 DEBUG[1078]: 240524 is not a local user
Jul 24 15:23:12 DEBUG[1078]: 240524 is not a local user
Jul 24 15:23:12 DEBUG[1078]: Stopping retransmission on
'SD28c9b01-2d5e97b21c9e4e488ce05aeda05558a8-js11002' of Response 623264158:
Found





Now I figured I would turn on 'sip debug' to which I see a lot more, here is
some of that output:

Jul 24 15:24:33 VERBOSE[1078]: 

Sip read: 
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 147.135.0.128:5060;branch=z9hG4bK2qo3b3307oi0j7onc400.1sr
From: "Fork
MD";tag=SD2o51f01-520831772-112223307
3802
To: "Howard Leadmon"
Call-ID: SD2o51f01-62b2c5360881c20fae16de626946fdf0-js11002
CSeq: 623304774 INVITE
Contact: 
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,UPDATE,NOTIFY
Supported: 100rel
Accept: application/sdp,application/dtmf
Max-Forwards: 69
Content-Type: application/sdp
Content-Length: 276

v=0
o=BroadWorks 24463992 1 IN IP4 147.135.0.128
s=-
c=IN IP4 147.135.0.128
t=0 0
m=audio 14942 RTP/AVP 0 8 2 18 96 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000
a=rtpmap:96 iLBC/8000
a=rtpmap:101 telephone-event/8000

Jul 24 15:24:33 VERBOSE[1078]: 13 headers, 12 lines
Jul 24 15:24:33 VERBOSE[1078]: Using latest request as basis request
Jul 24 15:24:33 VERBOSE[1078]: Sending to 147.135.0.128 : 5060 (non-NAT)
Jul 24 15:24:33 VERBOSE[1078]: Found peer 'sip.broadvoice.com'
Jul 24 15:24:33 DEBUG[1078]: Setting NAT on RTP to 0
Jul 24 15:24:33 VERBOSE[1078]: Found RTP audio format 0
Jul 24 15:24:33 VERBOSE[1078]: Found RTP audio format 8
Jul 24 15:24:33 VERBOSE[1078]: Found RTP audio format 2
Jul 24 15:24:33 VERBOSE[1078]: Found RTP audio format 18
Jul 24 15:24:33 VERBOSE[1078]: Found RTP audio format 96
Jul 24 15:24:33 VERBOSE[1078]: Found RTP audio format 101
Jul 24 15:24:33 VERBOSE[1078]: Peer audio RTP is at port 147.135.0.128:14942
Jul 24 15:24:33 DEBUG[1078]: Peer audio RTP is at port 147.135.0.128:14942
Jul 24 15:24:33 VERBOSE[1078]: Found description format PCMU
Jul 24 15:24:33 VERBOSE[1078]: Found description format PCMA
Jul 24 15:24:33 VERBOSE[1078]: Found description format G726-32
Jul 24 15:24:33 VERBOSE[1078]: Found description format G729
Jul 24 15:24:33 VERBOSE[1078]: Found description format iLBC
Jul 24 15:24:33 VERBOSE[1078]: Found description format telephone-event
Jul 24 15:24:33 VERBOSE[1078]: Capabilities: us - 0xc (ulaw|alaw), peer -
audio=0x51

[Asterisk-Users] DID + 800 Providers


Hello,

I'm looking for US DID and US50/CA 800# Providers.

I found voiceconduits.com 8 month ago, there interface looks good, but 
there are still not live, I believe they won't be any time soon.


I found sixtel, but order take eternities, they probably won't get my 
orders right any soon.


So i'm looking for a good provider for DIDs and 800# from the US and CA, 
who offer online signup and ordering. The provisioning should be less 
than 12 hours, preferably instantly.


If anybody knows or even uses such a provider, please leave me a note.

Many thanks,

Marc

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Re: [Asterisk-Users] Re: Re: Business Edition

2005-07-24 Thread Brian West

But I guess I'm wondering ... does the present licensing model discourage other vendors from contributing to *? I'm not sure Sangoma developers could sign the disclaimers even if they wanted to ... but then again I don't know if there's anyone there with anything to offer. I would think that that fact that they're selling hardware that supports * means that there's _some_ sharp cookies there, but perhaps they're just kernel module/driver hackers out to make a quick buck off of Digiums's back without contributing to the core?Sangoma does have a disclaimer on file with Digium as well as a few of their resellers that I know of.  app_dictate was sponsored by Sangoma.. written by anthm.  Competition is great for hardware vendors regardless of who did what... This is the nature of open source.  99% use it... 1% help out!/b___
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Re: [Asterisk-Users] Analog extensions behind E1, how to create them?

2005-07-24 Thread Steve Totaro

Depends how many.  For a few you can go with a TDM40b.  For large numbers
you can get a quad T1/E1 board and hook one port to the PSTN E1 and another
port configure at T1 and attach it to a channel bank such as the Adtran 600
with up to 24 FXS ports.


- Original Message - 
From: "Denis Galvão - iSolve" <[EMAIL PROTECTED]>
To: "Asterisk Users" 
Sent: Saturday, July 23, 2005 11:13 AM
Subject: [Asterisk-Users] Analog extensions behind E1, how to create them?


> I will have some extensions behind an E1. All of them will need the
> features/applications of Asterisk.
>
> Analog Extensions <-> PABX E1 <-> E1 Asterisk IP <-> VoIP trunk
>  ^
>  |
>  |
>  IP Phones
>
> How is the best way to create this users on Asterisk? Some of them
> will have a SIP account to have its extensions with mobile
> functionality when they will be out of office, others will not have
> this feature.
>
> Some examples will be great!
>
> Thanks.
>
> Denis.
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RE: [Asterisk-Users] ASTCC gives me only the time, but no cost

2005-07-24 Thread Rusty Shackleford

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Darren Wiebe
> Sent: Saturday, July 23, 2005 3:09 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] ASTCC gives me only the time, 
> but no cost
> 
> 
> Thank you very, very much Rusty.  I reopened the bug report. 
> http://bugs.digium.com/view.php?id=4479  I made a very slight 
> change to 
> the method it uses to calculate costs but it should implement the 
> connect charge properly.  Initially I rewrote the cost 
> calculation code 
> but that was not necessary, it can be implemented by changing the 
> following lines
> my $adjtime = int(($answeredtime + $increment - 1) / $increment) * 
> $increment
> 
> becomes
> 
> $adjtime = int((($answeredtime - $numdata->{includedseconds}) + 
> $increment - 1) / $increment) * $increment

This can yield a negative number, where $answeredtime <
"includedseconds", can it not?

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Re: [Asterisk-Users] Asterisk 1.2 is getting closer - please help

2005-07-24 Thread Kevin P. Fleming


Nir Simionovich wrote:

 I have a client using Asterisk as a predictive dialer. The dialer 
originates calls to people via a Zap channel,
and awaits input from the users. Problem is: the DTMF's are never picked 
up. Now, when I tried using

the stable branch - it worked.


This problem was fixed in Zaptel CVS HEAD over a month ago. If you are 
still having problems with it, they won't get fixed by keeping it to 
yourself... call Digium support and tell them about the issue, assuming 
you are using Digium hardware to interface to your TDM circuits.

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Re: [Asterisk-Users] Modifying astcc

2005-07-24 Thread Steve Totaro

It is actually found in astcc.agi

Find the line that fits the channel like:
$dialstr = "Zap/$res->{path}/$phone|30|HL(" . ($maxtime * 60 * 1000) .
":6:3)";

Change the |30| to |60| or whatever you want.

Thanks,
Steve

- Original Message - 
From: "Patricio Ku" <[EMAIL PROTECTED]>
To: 
Sent: Sunday, July 24, 2005 4:11 AM
Subject: RE: [Asterisk-Users] Modifying astcc


> in etc/asterisk/extensions.conf
> where you define the desteny you should put the time. example:
> exten=>_009[13456789].,1,Dial(SIP/operador/${EXTEN},60,tr)
> in this example I am giving 60 seconds waiting for someone to pickup the
> phone.
>
> >From: chawki hammoud <[EMAIL PROTECTED]>
> >Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
> >
> >To: Asterisk-Users@lists.digium.com
> >Subject: [Asterisk-Users] Modifying astcc
> >Date: Sat, 9 Jul 2005 12:50:14 -0700 (PDT)
> >
> >Hi:
> >
> >Astcc is working fine, except for one thing. It
> >doesn't give the called party enough time to answer
> >the phone. If nobody picks up in two rings, astcc
> >reports back no answer and hangs-up. The only instant
> >NOANSWER "value" was mentioned in astcc.agi script is:
> >
> >elsif ($res eq "NOANSWER") {
> > $res =
> >&mystreamfile("astcc-noanswer");
> >
> >
> >Please help me find what and where to change to
> >control the time astcc give to the called party to
> >answer.
> >
> >Regards;
> >Chawki Hammoud
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >__
> >Discover Yahoo!
> >Find restaurants, movies, travel and more fun for the weekend. Check it
> >out!
> >http://discover.yahoo.com/weekend.html
> >
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> _
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Amor
> & Amistad. http://match.msn.es/match/mt.cfm?pg=channel&tcid=162349
>
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Re: [Asterisk-Users] Why can't sip/200 call sip/202