Re: [Asterisk-Users] How to check Digium TE410P firmware version?

2006-01-07 Thread Dinesh Nair



On 12/27/05 22:39 Kevin P. Fleming said the following:

Steve 4 wrote:


Field-upgradeable?  Does that mean that I can do it myself?  That would
be great since some systems are in production and sending the board to
Digium takes time.



The 2nd gen firmware has field-upgradeability. The 1st gen firmware does 
not, unfortunately. There is not currently any 3rd gen firmware, but 
when there is, you'll be able to do it yourself :-)


we've got a number of TE410P 1st gen firmware cards. could we send them in 
to digium for a firmware upgrade ? the cards were purchased in october 2004 
from atp in melbourne.


--
Regards,   /\_/\   "All dogs go to heaven."
[EMAIL PROTECTED](0 0)http://www.alphaque.com/
+==oOO--(_)--OOo==+
| for a in past present future; do|
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[Asterisk-Users] Processor Update?

2006-01-07 Thread Mike Hammett



I've been Googling around for some time now (a few 
hours on dial-up).  I find all kinds of questions similar to mine, but 
either there is no answer or the answer has nothing to do with the 
question.  Hopefully this post isn't another one of those.
 
Does Asterisk favor FPU performance or clock 
speed?  (Meaning AMD or Intel)
 
I see Asterisk can be compiled for x64 
systems.  Does it run any better\worse on x64 versus i386?
 
Dual-core CPU performance isn't as good as dual 
CPU's, but is more often than not a better deal as it's close, but a lot 
cheaper.  Dual-core vs. Dual CPU performance depends on the 
application.  How does Asterisk respond?
 
Looking to build out a system capable of 100 
concurrent calls (IP pass-through, so in effect 200).  Looking at the 
dimensioning, I can get some ideas, but few of them state the call 
quality.  I'm hoping to be able to use a single CPU that is dual core as 
the price to go to a dual CPU that is single core puts me at least 60% to a 
redundant server.
 
More information about the setup
--
No cards of any kind (Tyan board with integrated 
video, NIC, SATA nothing more needed)
some IAX2 trunks (ztdummy)
mainly SIP clients, but an increasing number of 
IAX2
mainly ulaw\alaw, if not solely (the transcoding 
chews CPU cycles needlessly)
 
 
Mike HammettIntelligent Computing 
Solutionshttp://www.ics-il.com
 
 
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Re: [Asterisk-Users] Asterisk Jobs

2006-01-07 Thread Rusty Dekema
On 1/8/06, Douglas Garstang <[EMAIL PROTECTED]> wrote:
I'm not sure if the fact that my boss is an open source advocate is a good thing or not... ie yes it's great to work with Asterisk and see all the features coming together (especially with Polycom phones). On the other hand I wonder how useful this experience will really be. I see a lot of VOIP jobs requiring Cisco experience.
The answer to your question probably depends on what type of "IT Guy" you are (nope, we're not all the same), and the nature of your career goals. If you are the kind of guy who will learn, from working with Asterisk, a good overall view of the technologies behind and issues with semi-traditional and IP telephony, in addition to developing a sense of how things work and how to get things done with telephony systems in general, then I think it will be valuable experience. 
If you either aren't particularly interested in gaining a high-level overview of telephony systems, and/or if you are just not the kind of guy who will get that kind of a "big picture" by working with only one specific platform or system, then it may not be so valuable to you. 
Even if the latter is true, I don't think that would make you any less-qualified or less-skilled as an IT worker; I have simply noticed that people in the field see things and learn about systems in totally different ways. Some learn more by focusing on the specifics of various systems or platforms one by one, and some learn by constructing and updating a conceptual framework that contains within it the specifics of whatever system (Asterisk, in this case, but may include Cisco, etc in your future) is being worked on. 
If you learn things the former way, by focusing on a specific system at a time, your Asterisk experience will probably not be as valuable to you in terms of future jobs as would equivalent Cisco experience. But, if you tend to learn things in the latter way, your Asterisk experience is probably pretty much just as useful as if it were Cisco experience, or experience with any other vendor. 
Also, if you want to work for a large, formal company that places a lot of importance on titles and buzzwords, your lack of specific, major-vendor experience may present a problem. On the other hand, if you want to work for an outfit that is perhaps less formal and more unconventional, your Asterisk experience could stand you in very good stead, since Asterisk is such a flexible system, and since experience with it may indicate that you are a flexible programmer/net/sysadmin :). 
Anyway, those are my two cents on it. -Rusty
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[Asterisk-Users] Line Sharing or Better Call Pickup

2006-01-07 Thread Michael J. Liberatore








I have been trying to figure out for quite a while now how
to better setup asterisk in a small office environment..  For example,
small offices usually want to be able to have shared lines, so one can put a
line on hold and another person can pick that call up if it’s on
hold.  The astra and snom phones have the ability to use parking spots,
but asterisk doesn’t seem to support them, as in you cant put a call in a
slot and have a button and light for that slot light up on all the phones in a
group…  I found a patch for snom’s so that a call could be
picked up but its extreme beta and doesn’t seem stable enough for
production use.  People who are use to working in a small 4 line 5 –
10 phone office don’t want to go through trouble of parking calls and
having to tell someone else the parking spot number, and transfer is no good
cause sometimes they have to put the call to a person but that person is on the
phone or busy so they have to wait for a while which makes transfer not the
best option either….  I am hoping there is a solution for this I am
not seeing, maybe someone has some experience with small office setups because
this would seem to be major for any small setup.  Any suggestions or
experience would be greatly appreciated…  Thanks…

 

 






 
 
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RE: [Asterisk-Users] Asterisk Jobs

2006-01-07 Thread Douglas Garstang
Thanks all for the replies. I started working for a CLEC a few months ago and 
we've chosen to implement Asterisk. I'm not sure if the fact that my boss is an 
open source advocate is a good thing or not... ie yes it's great to work with 
Asterisk and see all the features coming together (especially with Polycom 
phones). On the other hand I wonder how useful this experience will really be. 
I see a lot of VOIP jobs requiring Cisco experience. I worked with VOIP back in 
1998, for a global VOIP wholesaler called OzEmail Interline in Australia before 
there where any standards... before SIP even. Until a few months ago I was 
working with SAN's and storage. Anywho...
 
Doug.

-Original Message- 
From: Robert La Ferla [mailto:[EMAIL PROTECTED] 
Sent: Sat 1/7/2006 8:56 PM 
To: Asterisk Users Mailing List - Non-Commercial Discussion 
Cc: 
Subject: Re: [Asterisk-Users] Asterisk Jobs



Douglas Garstang wrote:
> I'm curious why the number of jobs out there requiring Asterisk seems 
to be pretty low. After looking around dice, monster, careerbuilder etc, I was 
surprised to find no more than 3-4 employment opportunities with Asterisk 
throughout the US.
> 
> Is it really that low? There seems to be a job of opportunities for 
Cisco and other vendors solutions (duh... GUI's are good... duh). I wonder if 
demand will increase, or am I just looking in the wrong places?
>  
I think that the Asterisk customer profile can shed some light on this. 
If you are a big company, you'll buy into an expensive system because
you can afford it and rely on it.  If you are a small company, you will
look to Asterisk as an inexpensive way to set up your telephone system. 
You will also likely have staff that is willing to work with it and not
enough money or need to hire external consultants exclusively for
Asterisk.  You may have a telecom or networking consultant that will put
together the network and set up the system but Asterisk is a small piece
of it.  I'd say Asterisk is more of a "plus" in a job description but
not a "requirement".


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[Asterisk-Users] Agi Perl Talk Time

2006-01-07 Thread Code Lover
Hi all,

Can anyone tell me from where i can call my update query when the call
is pickuped using AGI Perl.

I writen the code to store active calls in MySQL DB. and display the
real time call counter. i mean to insert record when the call is
pickuped
--
Thank You,
Code Lover
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Re: [Asterisk-Users] Some advice on routing DID's

2006-01-07 Thread Darren Wiebe
Just grab the script.  I can help you with it off the mailing list if 
you like


Darren

Tom Vile wrote:


Do I need to install the complete ASTPP package or just utilize your
AGI script with the context for AMP?

Thanks
On 1/7/06, Tom Vile <[EMAIL PROTECTED]> wrote:
 


It's funny you mentioned that Darren, I was looking at your scripts
today.  I will evaluate it some more.
On 1/7/06, Darren Wiebe <[EMAIL PROTECTED]> wrote:
   


I have written an agi script that I use for that.  Then I can just have
a list of dids and extensions in a db.


Tom Vile wrote:

 


Would like some advice on the best way to route DID's to remote
asterisk servers.  Currently I have multiple DID's on my main Asterisk
server in a datacenter and have remote servers that connect via an IAX
trunk and when a call comes into my server I pass it to the iax peer.

Just wondering what the best way it is to do this without having to
have multiple line contexts for each remote server.

Thanks,
--
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Fax: 518-631-2856
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--
Darren Wiebe
[EMAIL PROTECTED]
Aleph Communications
ASTPP - Open Source Voip Billing & Calling Cards
www.aleph-com.net/astpp

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--
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Fax: 518-631-2856

   




--
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Fax: 518-631-2856
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--
Darren Wiebe
[EMAIL PROTECTED]
Aleph Communications
ASTPP - Open Source Voip Billing & Calling Cards
www.aleph-com.net/astpp

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Re: [Asterisk-Users] Immediate routing on "0" (DNIS)?

2006-01-07 Thread Ken D'Ambrosio

> I am stumped as well, you don't have any extension defined for either
> 0, _0X, or _0X.
> So I got no clue why *you* are stumped, in fact 1625 is treated
> special, because it got an extension.

Okay; thanks!  I mis-understood the mechanism.  I didn't think
extensions.conf actually came into play until it had all the DID stuff.  I
guess DTMF is DTMF, regardless of the source, huh?  In which case, I'm all
set -- again, thanks for setting me on the right road!

-Ken


>
>>
>> Thanks for any suggestions/ideas/etc...
>>
>> -Ken
>>
>> > Ken D'Ambrosio wrote:
>> >> I've got a T1 (E&M wink).  Our four-digit inbound DNIS numbers are in
>> the
>> >> range of 0600 - 1699.  However, the second that the "0" is seen on an
>> in-bound 06xx call, it stops listening for any more digits, and
>> immediately tries to route the call.  My 16xx numbers wait for all four
>> digits before trying to route.  Is there something, somewhere, that
>> tells
>> >> it to do an immediate route on seeing "0"?  I don't have much of
>> anything
>> >> in my extensions.conf file.  I'm seeing what's going on via
>> >> "tail -f /var/log/asterisk/full"
>> >>
>> >> Any suggestions?
>> >>
>> >> Thanks!
>> >>
>> >> -Ken
>> >> ___
>> >> --Bandwidth and Colocation provided by Easynews.com --
>> >>
>> >> Asterisk-Users mailing list
>> >> To UNSUBSCRIBE or update options visit:
>> >>http://lists.digium.com/mailman/listinfo/asterisk-users
>> >>
>> > Post your extensions.conf and what's on the CLI (asterisk -r)
>> >
>> > --
>> > Roman Volf
>> > Keystreams Internet Solutions
>> > [EMAIL PROTECTED]
>> >
>> > ___
>> > --Bandwidth and Colocation provided by Easynews.com --
>> >
>> > Asterisk-Users mailing list
>> > To UNSUBSCRIBE or update options visit:
>> >http://lists.digium.com/mailman/listinfo/asterisk-users
>> >
>>
>>
>>
>> ___
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>>
>> Asterisk-Users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
> ___
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> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
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>

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Re: [Asterisk-Users] Some advice on routing DID's

2006-01-07 Thread Tom Vile
Do I need to install the complete ASTPP package or just utilize your
AGI script with the context for AMP?

Thanks
On 1/7/06, Tom Vile <[EMAIL PROTECTED]> wrote:
> It's funny you mentioned that Darren, I was looking at your scripts
> today.  I will evaluate it some more.
> On 1/7/06, Darren Wiebe <[EMAIL PROTECTED]> wrote:
> > I have written an agi script that I use for that.  Then I can just have
> > a list of dids and extensions in a db.
> >
> >
> > Tom Vile wrote:
> >
> > >Would like some advice on the best way to route DID's to remote
> > >asterisk servers.  Currently I have multiple DID's on my main Asterisk
> > >server in a datacenter and have remote servers that connect via an IAX
> > >trunk and when a call comes into my server I pass it to the iax peer.
> > >
> > >Just wondering what the best way it is to do this without having to
> > >have multiple line contexts for each remote server.
> > >
> > >Thanks,
> > >--
> > >Tom Vile
> > >Baldwin Technology Solutions, Inc
> > >Consulting - Web Design - VoIP Telephony
> > >www.baldwintechsolutions.com
> > >Phone: 518-631-2855 x205
> > >Fax: 518-631-2856
> > >___
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> > >
> > >Asterisk-Users mailing list
> > >To UNSUBSCRIBE or update options visit:
> > >   http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> > >
> >
> >
> > --
> > Darren Wiebe
> > [EMAIL PROTECTED]
> > Aleph Communications
> > ASTPP - Open Source Voip Billing & Calling Cards
> > www.aleph-com.net/astpp
> >
> > ___
> > --Bandwidth and Colocation provided by Easynews.com --
> >
> > Asterisk-Users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
> --
> Tom Vile
> Baldwin Technology Solutions, Inc
> Consulting - Web Design - VoIP Telephony
> www.baldwintechsolutions.com
> Phone: 518-631-2855 x205
> Fax: 518-631-2856
>


--
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Fax: 518-631-2856
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Re: [Asterisk-Users] Immediate routing on "0" (DNIS)?

2006-01-07 Thread C F
On 1/7/06, Ken D'Ambrosio <[EMAIL PROTECTED]> wrote:
> > Post your extensions.conf and what's on the CLI (asterisk -r)
>
> As requested:
>
> # cat /etc/asterisk/extensions.conf
> [incoming]
> exten => s,1,Answer()
> exten => s,n,NoOp(CallerID is ${CALLERID})
> exten => s,n,NoOp(DID is ${DNID})
> exten => s,n,Background(enter-ext-of-person)
>
> exten => 1625,1,Playback(digits/1)
> exten => 1625,n,Goto(digits/1)
>
> exten => i,1,NoOp(CallerID is ${CALLERID})
> exten => i,n,NoOp(DID is ${DNID})
>
> And the console stuff for a 06xx DID:
>
>-- Starting simple switch on 'Zap/1-1'
>   == Unknown extension '0' in context 'incoming' requested
> -- Playing 'ss-noservice' (language 'en')
> -- Hungup 'Zap/1-1'
>
> For completeness' sake, here's the stuff from /var/log/asterisk/full; note
> that my 16xx extensions get all four DTMF digits before continuing:
>
> Jan  7 22:45:30 VERBOSE[5819] logger.c: -- Starting simple switch on
> 'Zap/1-1'
> Jan  7 22:45:30 DEBUG[5819] chan_zap.c: DTMF digit: 0 on Zap/1-1
> Jan  7 22:45:30 DEBUG[5819] chan_zap.c: No echo cancellation requested Jan
>  7 22:45:30 VERBOSE[5819] logger.c:   == Unknown extension '0' in context
> 'incoming' requested
> Jan  7 22:45:33 DEBUG[5819] channel.c: Scheduling timer at 160 sample
> intervals
> Jan  7 22:45:33 VERBOSE[5819] logger.c: -- Playing 'ss-noservice'
> (language 'en')
> [...]
>
> So, I'm kind of stumped.  As far as I can see, "0" is being treated
> special, and is trying to route "immediate" to the operator or something.
> [I've deleted the extensions.ael sample file, so that's not playing a
> role, either.]

I am stumped as well, you don't have any extension defined for either
0, _0X, or _0X.
So I got no clue why *you* are stumped, in fact 1625 is treated
special, because it got an extension.

>
> Thanks for any suggestions/ideas/etc...
>
> -Ken
>
> > Ken D'Ambrosio wrote:
> >> I've got a T1 (E&M wink).  Our four-digit inbound DNIS numbers are in the
> >> range of 0600 - 1699.  However, the second that the "0" is seen on an
> in-bound 06xx call, it stops listening for any more digits, and
> immediately tries to route the call.  My 16xx numbers wait for all four
> digits before trying to route.  Is there something, somewhere, that
> tells
> >> it to do an immediate route on seeing "0"?  I don't have much of anything
> >> in my extensions.conf file.  I'm seeing what's going on via
> >> "tail -f /var/log/asterisk/full"
> >>
> >> Any suggestions?
> >>
> >> Thanks!
> >>
> >> -Ken
> >> ___
> >> --Bandwidth and Colocation provided by Easynews.com --
> >>
> >> Asterisk-Users mailing list
> >> To UNSUBSCRIBE or update options visit:
> >>http://lists.digium.com/mailman/listinfo/asterisk-users
> >>
> > Post your extensions.conf and what's on the CLI (asterisk -r)
> >
> > --
> > Roman Volf
> > Keystreams Internet Solutions
> > [EMAIL PROTECTED]
> >
> > ___
> > --Bandwidth and Colocation provided by Easynews.com --
> >
> > Asterisk-Users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
>
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> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
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Re: [Asterisk-Users] Immediate routing on "0" (DNIS)?

2006-01-07 Thread Ken D'Ambrosio
> Post your extensions.conf and what's on the CLI (asterisk -r)

As requested:

# cat /etc/asterisk/extensions.conf
[incoming]
exten => s,1,Answer()
exten => s,n,NoOp(CallerID is ${CALLERID})
exten => s,n,NoOp(DID is ${DNID})
exten => s,n,Background(enter-ext-of-person)

exten => 1625,1,Playback(digits/1)
exten => 1625,n,Goto(digits/1)

exten => i,1,NoOp(CallerID is ${CALLERID})
exten => i,n,NoOp(DID is ${DNID})

And the console stuff for a 06xx DID:

   -- Starting simple switch on 'Zap/1-1'
  == Unknown extension '0' in context 'incoming' requested
-- Playing 'ss-noservice' (language 'en')
-- Hungup 'Zap/1-1'

For completeness' sake, here's the stuff from /var/log/asterisk/full; note
that my 16xx extensions get all four DTMF digits before continuing:

Jan  7 22:45:30 VERBOSE[5819] logger.c: -- Starting simple switch on
'Zap/1-1'
Jan  7 22:45:30 DEBUG[5819] chan_zap.c: DTMF digit: 0 on Zap/1-1
Jan  7 22:45:30 DEBUG[5819] chan_zap.c: No echo cancellation requested Jan
 7 22:45:30 VERBOSE[5819] logger.c:   == Unknown extension '0' in context
'incoming' requested
Jan  7 22:45:33 DEBUG[5819] channel.c: Scheduling timer at 160 sample
intervals
Jan  7 22:45:33 VERBOSE[5819] logger.c: -- Playing 'ss-noservice'
(language 'en')
[...]

So, I'm kind of stumped.  As far as I can see, "0" is being treated
special, and is trying to route "immediate" to the operator or something. 
[I've deleted the extensions.ael sample file, so that's not playing a
role, either.]

Thanks for any suggestions/ideas/etc...

-Ken

> Ken D'Ambrosio wrote:
>> I've got a T1 (E&M wink).  Our four-digit inbound DNIS numbers are in the
>> range of 0600 - 1699.  However, the second that the "0" is seen on an
in-bound 06xx call, it stops listening for any more digits, and
immediately tries to route the call.  My 16xx numbers wait for all four
digits before trying to route.  Is there something, somewhere, that
tells
>> it to do an immediate route on seeing "0"?  I don't have much of anything
>> in my extensions.conf file.  I'm seeing what's going on via
>> "tail -f /var/log/asterisk/full"
>>
>> Any suggestions?
>>
>> Thanks!
>>
>> -Ken
>> ___
>> --Bandwidth and Colocation provided by Easynews.com --
>>
>> Asterisk-Users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
> Post your extensions.conf and what's on the CLI (asterisk -r)
>
> --
> Roman Volf
> Keystreams Internet Solutions
> [EMAIL PROTECTED]
>
> ___
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>



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Re: [Asterisk-Users] Some advice on routing DID's

2006-01-07 Thread Tom Vile
It's funny you mentioned that Darren, I was looking at your scripts
today.  I will evaluate it some more.
On 1/7/06, Darren Wiebe <[EMAIL PROTECTED]> wrote:
> I have written an agi script that I use for that.  Then I can just have
> a list of dids and extensions in a db.
>
>
> Tom Vile wrote:
>
> >Would like some advice on the best way to route DID's to remote
> >asterisk servers.  Currently I have multiple DID's on my main Asterisk
> >server in a datacenter and have remote servers that connect via an IAX
> >trunk and when a call comes into my server I pass it to the iax peer.
> >
> >Just wondering what the best way it is to do this without having to
> >have multiple line contexts for each remote server.
> >
> >Thanks,
> >--
> >Tom Vile
> >Baldwin Technology Solutions, Inc
> >Consulting - Web Design - VoIP Telephony
> >www.baldwintechsolutions.com
> >Phone: 518-631-2855 x205
> >Fax: 518-631-2856
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> >
> >
>
>
> --
> Darren Wiebe
> [EMAIL PROTECTED]
> Aleph Communications
> ASTPP - Open Source Voip Billing & Calling Cards
> www.aleph-com.net/astpp
>
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--
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Fax: 518-631-2856
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Re: [Asterisk-Users] Some advice on routing DID's

2006-01-07 Thread Darren Wiebe
I have written an agi script that I use for that.  Then I can just have 
a list of dids and extensions in a db.



Tom Vile wrote:


Would like some advice on the best way to route DID's to remote
asterisk servers.  Currently I have multiple DID's on my main Asterisk
server in a datacenter and have remote servers that connect via an IAX
trunk and when a call comes into my server I pass it to the iax peer.

Just wondering what the best way it is to do this without having to
have multiple line contexts for each remote server.

Thanks,
--
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Fax: 518-631-2856
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--
Darren Wiebe
[EMAIL PROTECTED]
Aleph Communications
ASTPP - Open Source Voip Billing & Calling Cards
www.aleph-com.net/astpp

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[Asterisk-Users] Some advice on routing DID's

2006-01-07 Thread Tom Vile
Would like some advice on the best way to route DID's to remote
asterisk servers.  Currently I have multiple DID's on my main Asterisk
server in a datacenter and have remote servers that connect via an IAX
trunk and when a call comes into my server I pass it to the iax peer.

Just wondering what the best way it is to do this without having to
have multiple line contexts for each remote server.

Thanks,
--
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Fax: 518-631-2856
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Re: [Asterisk-Users] Asterisk Jobs

2006-01-07 Thread Robert La Ferla

Douglas Garstang wrote:

I'm curious why the number of jobs out there requiring Asterisk seems to be 
pretty low. After looking around dice, monster, careerbuilder etc, I was 
surprised to find no more than 3-4 employment opportunities with Asterisk 
throughout the US.
 
Is it really that low? There seems to be a job of opportunities for Cisco and other vendors solutions (duh... GUI's are good... duh). I wonder if demand will increase, or am I just looking in the wrong places?
  
I think that the Asterisk customer profile can shed some light on this.  
If you are a big company, you'll buy into an expensive system because 
you can afford it and rely on it.  If you are a small company, you will 
look to Asterisk as an inexpensive way to set up your telephone system.  
You will also likely have staff that is willing to work with it and not 
enough money or need to hire external consultants exclusively for 
Asterisk.  You may have a telecom or networking consultant that will put 
together the network and set up the system but Asterisk is a small piece 
of it.  I'd say Asterisk is more of a "plus" in a job description but 
not a "requirement".



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Re: [Asterisk-Users] Screening incoming calls.

2006-01-07 Thread C F
Thanks
Sorry, I missed that local/8600 channel.

On 1/6/06, Philipp von Klitzing <[EMAIL PROTECTED]> wrote:
> Hi!
>
> > Thanks for that post thats a good one
>
> :-)
>
> > just one thing, what happens if the user doesn't want to connect to the
> > caller? does it get saved as VM? Looking thru the code I couldn't see
> > where that happens.
>
> The 1st MeetMe has three participants:
>
>caller
>AGI & .call file --> VoiceMail()
>AGI & .call file --> VM box owner (listen-only mode)
>
> VoiceMail() simply records the MeetMe conference.
>
>
> Cheers, Philipp
>
>
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RE: [Asterisk-Users] Asterisk Jobs

2006-01-07 Thread Steve Totaro
Asterisk is still virtually unknown to endusers. The only reason why
it's even a blip on the radar of PBX manufacturers is because how
quickly the community is growing, and how feature rich the system is
already.  The biggest threat is that it is free and not proprietary
which totally flies in the face of the whole tradition of the greedy
industry.  They see what happened to the music industry when it moved
too slowly and did not anticipate the market paradigm shift.

3com seems to have knowledge and insight into the future, at least that
is my take on the V3000.  A 1U IP PBX with one FXS port and four FXOs at
a slightly higher price to a decent Asterisk box.  I could easily sell
one of these boxes to an end user for slightly more than a comparable
Asterisk rig based on name recognition, the fact that they can call any
number of dealers in the area for support, the GUI, and the nice glossy
brochures!  If I demo the system, it is a no-brainer for the bean
counters, the suits, and even the overburdened techs.  Of course I would
mention that the system is very expandable and all you have to do is
plug in a phone and it will be ready to go automagically.  What I would
leave out is the fact that if, lets say you wanted to upgrade to a T1/E1
you would have to buy a different several U sized chassis and a card
that will probably set them back about $4k or so.  Also, I would not
mention that they were locked into 3com phones and that besides the high
price of the phone, they will need to also buy a license for it to work.


I think I read that Digium did something like $20 or $80 million in
business last year (obviously I am not sure of the figure but it was
impressive and more than I would have guessed for them).  That is a nice
chunk of change, but it is chump change to the overall PBX industry.  It
is like the penny that someone drops and doesn't even bother to pick up.

Looking quickly, I found this reference, "The PBX market at $13.2
billion in 2002 is forecast to reach $17.9 billion by 2008 during the
forecast period."
http://www.researchandmarkets.com/reportinfo.asp?report_id=34161 .

It is no wonder why you see very few jobs listed for Asterisk skilled
workers compared to the real market share holders.  The numbers just
aren't there.  There are plenty of consultants around the globe that can
SSH into a box on the other side of the world and configure it.  If I
was in the market to hire an Asterisk consultant, I would watch the list
and see who knows what the heck they are talking about and has a good
attitude or I would look at the wiki.  So far I have a few guys I would
call on if I needed some work done that I know would be high quality.  I
would call on Nicolas Gudino (the creator of AMP), Darren Wiebe (who
knows all about pre-paid, post paid, and what I am especially looking
forward to, the integration of OSCommerce to his platform, or the
Coalescent Systems guys.  I have some others but I keep them secret so
they are available when I need them.

I also have a feeling that most Asterisk jobs are self created.  

Will Asterisk get a chunk of the market share?  Probably some years down
the road but I will not hold my breath.  Not until there is something
similar to the marketing, documentation, sales channels, and name
recognition.  None of this will be easy since anyone can learn Asterisk
and start a company.  It is not like NEC for instance, where there can
only be so many distributors in an area and they must be certified.  If
they screw up, they lose their distributorship.  The Open Source
business model makes it almost impossible to emulate the same type of
marketing, sales, and business model.

Thanks,
Steve

I am going to cross post this to the biz list, since that is really
where it belongs.




> 
> Most of the Asterisk work I have found out and about is either done by
> internal staff or by companies wanting work done by external
contractors.
> 
> Like you, I have found very little in the way of full time jobs for
> 'asterisk people'
> 
> PaulH
> 
> - Original Message -
> From: "Douglas Garstang" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> ; "Asterisk Users Mailing List -
> Non-Commercial Discussion" 
> Sent: Sunday, January 08, 2006 10:47 AM
> Subject: [Asterisk-Users] Asterisk Jobs
> 
> 
> > I'm curious why the number of jobs out there requiring Asterisk
seems to
> be pretty low. After looking around dice, monster, careerbuilder etc,
I
> was
> surprised to find no more than 3-4 employment opportunities with
Asterisk
> throughout the US.
> >
> > Is it really that low? There seems to be a job of opportunities for
> Cisco
> and other vendors solutions (duh... GUI's are good... duh). I wonder
if
> demand will increase, or am I just looking in the wrong places?
> >
> > - Doug.
> >
> >
> 
> 
>

--
> --
> 
> 
> 
> > ___
> > --Ba

[Asterisk-Users] Up to 4 seconds delay to play prompt?

2006-01-07 Thread Andre Courchesne - Consultant

Hi,

 Some background... I have the following directories:
   /var/lib/asterisk/sounds/custom/  -> Here are french prompts
   /var/lib/asterisk/sounds/custom/en   -> Here are the english prompts

 If I do:
   SetLanguage(en)
   Playback(custom/mypromp)

 The prompt file is played immediatly

 If I do:
   SetLanguage(fr)
   Playback(custom/myprompt)

 There is a good 4 seconds delay before tha prompt is played...

 Any hints?

Andre Courchesne
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[Asterisk-Users] Kudzu and Zaptel Cards

2006-01-07 Thread Bart Fisher



Redhat has a 'Hardware Discovery Utility' called 
Kudzu.
 
When I change cards, kudzu pops up and ask to 
remove/config the card.
Most of the time kudzu has trouble recognizing the 
Digium Zaptel cards and calls them something wrong, like calling the TDM card a 
network card. 
 
I'm having a devil of a time getting 3 
TE410P cards to come up with all green lights. For example one or two cards 
full green, and the other has one red and yellow.  Swap cards give me some 
other form of workingness.
 
My question are: 
 
1) How necessary is Kudzu?
2) Should it ran at all?
3) If I choose ignore or disable kudzu, will it 
stop the zaptel cards from being detected or working?
 
Any hints will be welcome
 
TIA
 
Bart
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RE: [Asterisk-Users] Asterisk Market Share

2006-01-07 Thread Dean Collins
You could probably pay $15-20 for a paul budde report with relatively
accurate figures. www.budde.com.au (even if he does believe asterisk is
a passing fad - hi Paul :) he's still one of the best telco resources in
Australia.

Telsyste might be another option.


Cheers,
Dean


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dakota
Sent: Saturday, 7 January 2006 8:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Asterisk Market Share

Does anyone know where I can find some stats on the following

The % market share for the following types of Ip PBX

Mitel
Nortel
Asterisk
etc

Thanks!

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Re: [Asterisk-Users] Asterisk Jobs

2006-01-07 Thread pdhales
Most of the Asterisk work I have found out and about is either done by
internal staff or by companies wanting work done by external contractors.

Like you, I have found very little in the way of full time jobs for
'asterisk people'

PaulH

- Original Message - 
From: "Douglas Garstang" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
; "Asterisk Users Mailing List -
Non-Commercial Discussion" 
Sent: Sunday, January 08, 2006 10:47 AM
Subject: [Asterisk-Users] Asterisk Jobs


> I'm curious why the number of jobs out there requiring Asterisk seems to
be pretty low. After looking around dice, monster, careerbuilder etc, I was
surprised to find no more than 3-4 employment opportunities with Asterisk
throughout the US.
>
> Is it really that low? There seems to be a job of opportunities for Cisco
and other vendors solutions (duh... GUI's are good... duh). I wonder if
demand will increase, or am I just looking in the wrong places?
>
> - Doug.
>
>






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[Asterisk-Users] Asterisk Market Share

2006-01-07 Thread Dakota

Does anyone know where I can find some stats on the following

The % market share for the following types of Ip PBX

Mitel
Nortel
Asterisk
etc

Thanks!

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Re: [Asterisk-Users] how to configure iax account for iaxmodem?

2006-01-07 Thread Lee Howard

Bruno Voigt wrote:


Hi,

I'm running asterisk 1.2.1 and started to play with iaxmodem 0.0.7
running on the same box.

I wonder how to setup the iax account correctly so that I may
initiate outbound calls via iaxmodem?

registration upon iaxmodem startup is okay and I can direct calls to it.

   -- Registered IAX2 'iaxmodem' (AUTHENTICATED) at 127.0.0.1:33874

But upon an outbound call setup request from iaxmodem
(ATD123) I get the following asterisk error:

Jan  8 01:06:29 NOTICE[10273]: chan_iax2.c:6778 socket_read: Rejected
connect attempt from 127.0.0.1, who was trying to reach '123@'

What is missing in the iax.conf to make asterisk accept the outbound calls?

[iaxmodem]
type=peer
callerid="iaxmodem"
username=iaxmodem
secret=password
host=dynamic
disallow=all
allow=alaw
peercontext=from-internal
context=from-internal



Hrmmm... iaxmodem registers properly but gets a rejection when trying to 
place a call.  It's probably a codec problem.  Try adding allow=ulaw and 
allow=slinear to your iaxmodem context and see if that helps.


Lee.
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Re: [Asterisk-Users] wich IAX soft client allow to specify a different server port?

2006-01-07 Thread Antonio Gallo

[[:[EMAIL PROTECTED][:][/[@]]


Well but i don't need to dial out, i need to register to asterisk
using IAX and 8080 port and all the client i've tested will not
allow that into their account config section: they just have the server
name/ip not the port.
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[Asterisk-Users] how to configure iax account for iaxmodem?

2006-01-07 Thread Bruno Voigt
Hi,

I'm running asterisk 1.2.1 and started to play with iaxmodem 0.0.7
running on the same box.

I wonder how to setup the iax account correctly so that I may
initiate outbound calls via iaxmodem?

registration upon iaxmodem startup is okay and I can direct calls to it.

-- Registered IAX2 'iaxmodem' (AUTHENTICATED) at 127.0.0.1:33874

But upon an outbound call setup request from iaxmodem
(ATD123) I get the following asterisk error:

Jan  8 01:06:29 NOTICE[10273]: chan_iax2.c:6778 socket_read: Rejected
connect attempt from 127.0.0.1, who was trying to reach '123@'

What is missing in the iax.conf to make asterisk accept the outbound calls?

[iaxmodem]
type=peer
callerid="iaxmodem"
username=iaxmodem
secret=password
host=dynamic
disallow=all
allow=alaw
peercontext=from-internal
context=from-internal


TIA, Bruno

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Re: [Asterisk-Users] Possible bug with GotoIfTime

2006-01-07 Thread Derek Whitten
Bill Michaelson wrote:
> Running a fairly recent subversion release of Asterisk, I'm running into
> a problem using labels (as opposed to priorities) with this application.
> 
> Here is the dialplan segment:
> 
> ; isolate gotoiftime bug with labels
> ;exten => 806,1,GotoIfTime(8:00-20:00,*,*,*?2:4)
> exten => 806,1,GotoIfTime(8:00-20:00,*,*,*?light:dark)
> exten => 806,n(light),noop(light)
> exten => 806,n,hangup
> exten => 806,n(dark),noop(dark)
> exten => 806,n,hangup
> 
> As coded, this is what happens when it executes:
> 
>-- Executing GotoIfTime("IAX2/hack-2", "8:00-20:00|*|*|*?light:dark")
> in new stack
> Jan  7 18:38:09 NOTICE[28137]: pbx.c:1705 pbx_extension_helper: No such
> label 'light:dark' in extension '806' in context 'default'
> Jan  7 18:38:09 WARNING[28137]: pbx.c:6312 ast_parseable_goto: Priority
> 'light:dark' must be a number > 0, or valid label
>  == Spawn extension (default, 806, 1) exited non-zero on 'IAX2/hack-2'
>-- Hungup 'IAX2/hack-2'
> 
> But if I disable the second exten line instead of the first, it works
> properly.
> 
> Beware.
> 
> 
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GotoIfTime(8:00-20:00|*|*|*?light:dark|s|1)




-- 
.


-BEGIN GEEK CODE BLOCK-
Version: 3.1
GCS/GIT> d-@ s+:+ a? C+++ BLHIS$ U+++ P+> L+++ !E W+++$ N++ o+ K w--
PS+++ PE@ Y+ PGP++ t 5? X !R tv+ b- DI-- D G e+> h r+++ y
 --END GEEK CODE BLOCK--


.


signature.asc
Description: OpenPGP digital signature
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Re: [Asterisk-Users] Draytek Vigor 2900 & Asterisk

2006-01-07 Thread Wayne

Hiya,

I've got a 2900g series, and it works fine (I have the 2200we before I 
upgraded and that was ok too!). I have used its built in wifi to go to 
an ipaq iax extension, and also have asterisk doing sip and iax through 
to fwd and sipgate. There's some port forwarding rules to get the 
protocols through to my asterisk box and that's about it!


From what I may have heard (although I've never tried this) the 'V' or 
voice version have to terminate SIP calls on itself - i.e. you cant pass 
the ports through to another SIP box - similarly the analogue ports have 
to go to the router. Like I say - I've never tried this - its what I 
found out from the owners forums.


HTH.
Wayne.

Jonathan Attwood wrote:


I'm in conversation with Draytek's pre-sales dept..

Here's the most recent reply:



All I want to know is, if I buy one of these routers, will it break my setup
or not - ie. assuming I set up the relevant port-forwarding, can I
expect any one-way audio issues. Can't get a definitive answer from
suppliers or the manufacturer, so I hope someone here uses this model
with Asterisk.?
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[Asterisk-Users] Asterisk Jobs

2006-01-07 Thread Douglas Garstang
I'm curious why the number of jobs out there requiring Asterisk seems to be 
pretty low. After looking around dice, monster, careerbuilder etc, I was 
surprised to find no more than 3-4 employment opportunities with Asterisk 
throughout the US.
 
Is it really that low? There seems to be a job of opportunities for Cisco and 
other vendors solutions (duh... GUI's are good... duh). I wonder if demand will 
increase, or am I just looking in the wrong places?
 
- Doug.
 
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[Asterisk-Users] Possible bug with GotoIfTime

2006-01-07 Thread Bill Michaelson
Running a fairly recent subversion release of Asterisk, I'm running into 
a problem using labels (as opposed to priorities) with this application.


Here is the dialplan segment:

; isolate gotoiftime bug with labels
;exten => 806,1,GotoIfTime(8:00-20:00,*,*,*?2:4)
exten => 806,1,GotoIfTime(8:00-20:00,*,*,*?light:dark)
exten => 806,n(light),noop(light)
exten => 806,n,hangup
exten => 806,n(dark),noop(dark)
exten => 806,n,hangup

As coded, this is what happens when it executes:

   -- Executing GotoIfTime("IAX2/hack-2", 
"8:00-20:00|*|*|*?light:dark") in new stack
Jan  7 18:38:09 NOTICE[28137]: pbx.c:1705 pbx_extension_helper: No such 
label 'light:dark' in extension '806' in context 'default'
Jan  7 18:38:09 WARNING[28137]: pbx.c:6312 ast_parseable_goto: Priority 
'light:dark' must be a number > 0, or valid label

 == Spawn extension (default, 806, 1) exited non-zero on 'IAX2/hack-2'
   -- Hungup 'IAX2/hack-2'

But if I disable the second exten line instead of the first, it works 
properly.


Beware.


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Re: [Asterisk-Users] Immediate routing on "0" (DNIS)?

2006-01-07 Thread Roman Volf

Ken D'Ambrosio wrote:

I've got a T1 (E&M wink).  Our four-digit inbound DNIS numbers are in the
range of 0600 - 1699.  However, the second that the "0" is seen on an
in-bound 06xx call, it stops listening for any more digits, and
immediately tries to route the call.  My 16xx numbers wait for all four
digits before trying to route.  Is there something, somewhere, that tells
it to do an immediate route on seeing "0"?  I don't have much of anything
in my extensions.conf file.  I'm seeing what's going on via
"tail -f /var/log/asterisk/full"

Any suggestions?

Thanks!

-Ken
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Post your extensions.conf and what's on the CLI (asterisk -r)

--
Roman Volf
Keystreams Internet Solutions
[EMAIL PROTECTED]

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[Asterisk-Users] Immediate routing on "0" (DNIS)?

2006-01-07 Thread Ken D'Ambrosio
I've got a T1 (E&M wink).  Our four-digit inbound DNIS numbers are in the
range of 0600 - 1699.  However, the second that the "0" is seen on an
in-bound 06xx call, it stops listening for any more digits, and
immediately tries to route the call.  My 16xx numbers wait for all four
digits before trying to route.  Is there something, somewhere, that tells
it to do an immediate route on seeing "0"?  I don't have much of anything
in my extensions.conf file.  I'm seeing what's going on via
"tail -f /var/log/asterisk/full"

Any suggestions?

Thanks!

-Ken
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[Asterisk-Users] choppy music on hold - only on PRI PSTN

2006-01-07 Thread Goran Skular








Hello to all…

 

I do not know what is causing choppy music on hold
when call comes in through E1 card (PRI).. but this channel info is somehow
strange.. We use Alaw over PRI (and I think it’s format number 8), 

But why is WriteFormat at 2 ?

 

Thanks!

 

show channel Zap/1-1

 -- General --

   Name: Zap/1-1

   Type: Zap

   UniqueID: 1136667936.0

  Caller ID: 04573573

 Caller ID Name: (N/A)

    DNID Digits: 349

  State: Up (6)

  Rings: 1

   NativeFormat: 72

    WriteFormat: 2

 ReadFormat: 8

1st File Descriptor: 14

  Frames in: 3516

 Frames out: 3352

 Time to Hangup: 0

   Elapsed Time: 0h1m10s

  Direct Bridge: 

Indirect Bridge: 

 --   PBX   --

    Context: OZ0800

  Extension: s

   Priority: 7

 Call Group: 0

   Pickup Group: 0

    Application: Queue

   Data: OZ0800|Tt|||300

    Blocking in: ast_waitfor_nandfds

 






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RE: [Asterisk-Users] Problens to link 2 * servers

2006-01-07 Thread Carlos Alperin
>From this server can you ping 10.0.0.121?

What is your network mask?

10.0.0.121/100 is not a valid address (mask are in the range of /0 to /32)

This is where you should start.

What is your network definition?

Tudo bem?

Carlos Alperin

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Moises Silva
Sent: Saturday, January 07, 2006 10:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Problens to link 2 * servers

network problems. Asterisk wan unable to connect or bind to 10.0.0.121/100

Regards
On 1/7/06, Cleyverson P. Costa <[EMAIL PROTECTED]> wrote:
>
> Hello,
>
> I'm traying to link 2 * servers using SIP and the following errors was
show:
>
> "SIP/AsteriskA:[EMAIL PROTECTED]/100") in new stack
> Dec 13 22:46:57 WARNING[8767]: chan_sip.c:1398 create_addr: No such
> host: 10.0.0.121/100
> Dec 13 22:46:57 NOTICE[8767]: app_dial.c:759 dial_exec: Unable to
> create channel of type 'SIP'
>   == Everyone is busy/congested at this time
> Dec 13 22:47:07 NOTICE[8767]: rtp.c:435 ast_rtp_read: RTP: Received
> packet with bad UDP checksum
> Dec 13 22:47:07 WARNING[8767]: pbx.c:1949 ast_pbx_run: Timeout, but no
> rule 't' in context 'internal'
> linux*CLI>
>
> Someone could help me to fix the following problens??
>
> fraternatly,
>
>
> Cleyverson Pereira Costa
> 
> Phone #: 27+9922-0111
> Skype: cleyverson
> MSN: [EMAIL PROTECTED]
> 
> "Escolhe, pois, a vida, para que vivas, tu e a tua descendência."
> Deuteronômio 30:19
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>


--
"Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org";
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Re: [Asterisk-Users] How to Unload app_rxfax.so

2006-01-07 Thread Alberto Sagredo

Yes, you could do that making some changes on modules.conf

noload => app_rxfax.so

Regards

Alberto

Nitesh Divecha wrote:


Hello All,

Dunno what happen but Asterisk is refusing to start... Went over the  
log and found out that "app_rxfax.so" is failing to load.


Jan  7 11:57:28 VERBOSE[4320] logger.c:  [app_rxfax.so]Jan  7  
11:57:28 WARNING[4320] loader.c: /usr/lib/asterisk/modules/ 
app_rxfax.so: undefined symbol: fax_set_phase_d_handler
Jan  7 11:57:28 WARNING[4320] loader.c: Loading module app_rxfax.so  
failed!


Is there any way to bypass this module and start Asterisk...

I think it was a bad idea to compile Asterisk with fax capability...

Thanks,
Neal


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[Asterisk-Users] How to Unload app_rxfax.so

2006-01-07 Thread Nitesh Divecha

Hello All,

Dunno what happen but Asterisk is refusing to start... Went over the  
log and found out that "app_rxfax.so" is failing to load.


Jan  7 11:57:28 VERBOSE[4320] logger.c:  [app_rxfax.so]Jan  7  
11:57:28 WARNING[4320] loader.c: /usr/lib/asterisk/modules/ 
app_rxfax.so: undefined symbol: fax_set_phase_d_handler
Jan  7 11:57:28 WARNING[4320] loader.c: Loading module app_rxfax.so  
failed!


Is there any way to bypass this module and start Asterisk...

I think it was a bad idea to compile Asterisk with fax capability...

Thanks,
Neal


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RE: [Asterisk-Users] RE: Fax, txfax -bizarre thing

2006-01-07 Thread Colin Anderson



> Of course that's not a problem to 
use hylafax, but I just want to have it on one machine (I'm afraid that Asterisk 
and hylafax > won't run on the same 
machine :( )
[Colin 
Anderson] I am experimenting with IAXmodem to Hylafax running on an 
Asterisk server. It works. Last Thur, I had 98 virtual modems recognized and 
running under Hylafax. So far, I can send and recieve faxes reasonably well but 
there's some configuration issues I have to get out of the way before I would 
beta it on my users. I expect that Hylafax to a couple of plain old USR modems 
running on /ttys0 and /ttys1 would work fine enough even if Asterisk was on the 
box. Even though the postscript conversion is done on the server that can be 
controlled with -nice. After that all that Hylafax has to do is service the 
modems and the clients. This seems to be pretty low overhead - I think the 
client protocol is FTP on a nonstandard port, and servicing a couple of modems 
at 9600 can't be too taxing. Hell, 25 modems shouldn't be taxing, on a modern 
machine.
 
BTW I have used, installed, admin'd etc about a dozen big Windows 
faxing solutions (basically all of the big players) and I've never 
used Hylafax before, and I was really impressed. It's better than 80% of the 
Windows product offerings, IMO. 

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Re: [Asterisk-Users] SPA-3000 is translating vocal sounds into DTMF

2006-01-07 Thread Brian Capouch

Rich Adamson wrote:



Its a fairly common problem with the spa3k, but is somewhat dependent on
how it is configured and the distance to your CO.



Just to add a couple of data points:

I don't know why, but for me the problem has been worse lately than it 
had been during the early time I spent with this unit.  I have been 
using it pretty exclusively for about a month and a half; only in the 
past few days have those tones really become objectionable.


Second, I'm not *any* distance from the CO: my FXO is the port on an 
ATA186 (original issue, 2002 vintage) from Vonage.


Finally, for sure some of the calls on which I'm experiencing the 
problem aren't even going out the FXO port on the thing; they are ITSP 
calls made using the FXS port only.


FWIW.

B.
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Re: [Asterisk-Users] Dialer

2006-01-07 Thread Darren Wiebe
http://www.astpp.org/index.php?n=Misc.AutoDialOut 


I put together what I have on that site.

Darren wiebe
[EMAIL PROTECTED]

Steve Totaro wrote:


Darren,

I am interested in your project.  Let me know if I can help you test.

Thanks,
Steve

 


-Original Message-
From: Wiley Siler [mailto:[EMAIL PROTECTED]
Sent: Friday, January 06, 2006 12:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Dialer

If this or any other example is available, I would be most thankful to
have it.

I got the go ahead on this project to day so now I have to start
   


seeing
 


how to do this.

Thanks,
Wiley


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Darren
Wiebe
Sent: Tuesday, January 03, 2006 5:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Dialer

I'm supposed to have a "mostly" canned script that will do this done
already.  It will pull the list of people to call out of a db and play
them the file specified in the db table.  Contact me offlist if you're
interested.  It will be done real soon but I'm not done testing yet.

Darren Wiebe
[EMAIL PROTECTED]

Kerry Garrison wrote:

   


You actually aren't far from it. If the system only needs to play
 


the
 


same file to each person, a simple script can be used to pull from a
database and create call files. Asterisk will use the call files to
place the calls and play a sound. A few minutes of searching on that
should get you started. I haven't seen anyone else have a canned
script ready to go, but would like to know if anyone does.
-Kerry



 



 


   *From:* [EMAIL PROTECTED]
   [mailto:[EMAIL PROTECTED] *On Behalf Of
   *Wiley Siler
   *Sent:* Tuesday, January 03, 2006 3:32 PM
   *To:* Asterisk Users Mailing List - Non-Commercial Discussion
   *Subject:* [Asterisk-Users] Dialer

   Hello All,

   I am having trouble finding a specific * piece of software so I
   thought I would see If you guys can help me get my terminology
 


clear.
   


   First off let me premise this with "no, this is absolutely not
 


for
 


   doing call marketing".
   I need to make my Asterisk box call a group of people and play
   them a message.
   My company deals with education so we need to do follow ups if
   students are not logging on.
   We do this manually now but it would be easier and cheaper to
 


just
 


   play them a message.

   What is the term I should be looking for?  I keep thinking "auto
   dialer" or something like that but I am not quite getting there.

   Any help would be appreciated.  I have been learning a bit of
 


Perl
 


   so I was thinking I could auto generate and AGI file and then
 


just
 


   do a Play() of the mp3 when they pick up at the other end?
 


Seems
 


   a little kludge though.


   Thanks,
   Wiley


 


---
   


-

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--
Darren Wiebe
[EMAIL PROTECTED]
Aleph Communications
ASTPP - Open Source Voip Billing & Calling Cards
   


www.aleph-com.net/astpp
 


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--
Darren Wiebe
[EMAIL PROTECTED]
Aleph Communications
ASTPP - Open Source Voip Billing & Calling Cards
www.aleph-com.net/astpp

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Re: [Asterisk-Users] Recording Calls at the phone

2006-01-07 Thread Ioan Indreias
A (too) simple sollution to your problem is to take the analog audio from
your IP phone using a module atached between the curly handset cord and
the base unit of the IP phone - like
http://www.quasarelectronics.com/tre156.htm

So, basically you need to change the old "RJ11 - 1/8 inch recording -
RJ11" system you have used to a new one with "RJ10 - 1/8 inch recording -
RJ10".
Sure, this solution works only if the handeset it is attached through a
RJ10 port to the handset.

I do not know exactly how your software will deal with this change as
there should be a mechnism to start & stop recording based on the audio
level injected into PC's audio card (mic port).

Hope it helps.

Ioan Indreias
Modulo Consulting - http://www.modulo.ro



> I'm not really trying to monitor anything on the asterisk box at all. I
> guess this is more of an SIP phone question. Really all I need is to get
> the audio from an SIP phone, both the caller and callie, to a 1/8th inch
> stereo jack that I can plug into a mic input.
>
> Michael Sampson
> Information Systems Manager
> Customer Contact Services
> [EMAIL PROTECTED]
> 952-936-4000
>
>
>
> Douglas Garstang wrote:
>
>>On Demand-monitoring? If your referring to monitoring specific agents
>> calls, I'm still trying to work out how to do that. You can either
>> monitor all calls for a queue, or all calls for all agents, but not all
>> calls for a specific agent. I tried to use the Monitor() command on it's
>> own to start recording when an agent receives a call, but that does not
>> appear to work.
>>
>>-Original Message-
>>From: Francesco Peeters (Asterisk) [mailto:[EMAIL PROTECTED]
>>Sent: Friday, January 06, 2006 7:53 AM
>>To: Asterisk Users Mailing List - Non-Commercial Discussion
>>Cc: Asterisk Users Mailing List - Non-Commercial Discussion
>>Subject: Re: [Asterisk-Users] Recording Calls at the phone
>>
>>
>>On Fri, January 6, 2006 15:37, Michael Sampson said:
>>
>>
>>>I work for a call center and we are looking at using asterisk to have
>>>our operators take calls. Our message taking software records all the
>>>calls on the operators computers. Right now we use these recording
>>>controls from radio shack that plug in between the wall jack and the
>>>phone and plug in via a 1/8 inch stereo connector to the mic input on
>>>the computer. If I buy an IP phone I can't do that. I could get an FXO
>>>adapter and regular phones, but I'm looking to get as little equipment
>>>as possible. Radio shack makes a recording control that plugs in to a
>>>2.5 mm headset jack, but it takes batteries so thats not going to work
>>>
>>>Does anyone else do something similar? Does anyone have any ideas about
>>>what producs/setup would work for this.
>>>
>>>
>>>
>>
>>Asterisk has a built in monitoring system. You can chose to do Always,
>>Never or On Demand monitoring, depending on your setup and dialplan
>>
>>
>>
>>Good luck!
>>
>>
>>
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RE: [Asterisk-Users] Latency

2006-01-07 Thread Alexander Lopez
 That would depend heavaly on your netowrk. Would your Swtiches (not
routers as TMDoE is layer 2) I pulled up an old posting from Mark on
TDMoE.

http://www.marko.net/asterisk/archives/0301/0566.html

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Aaron Daniel
> Sent: Friday, January 06, 2006 5:56 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Latency
> 
> Does anyone know if using TDMoE instead of straight SIP 
> between servers would increase or decrease latency?
> 
> Aaron
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RE: [Asterisk-Users] Help Connecting server districts

2006-01-07 Thread Alexander Lopez
I would agree with all but a few issue:

I would incoparate dundi, After using it I have fallen in love with it
for distributed applications such as this. It makes configuration at
first a bit steeper but it picks up monentum as your deploy. With Dundi
you basicaly broadcast what extensions or numbers are served by your
machine and using a set of keys (which negats having to configure a perr
for every machine to create a mesh netowrk)

There is no need to connect to the Public Dundi Peering fabric unless
you want to. You can run your own 'private' Dundi  peers.

Let me know if you need any help..
 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Rich Adamson
> Sent: Saturday, January 07, 2006 8:30 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Help Connecting server districts
> 
> 
> > I am working on a project to unite several local school 
> districts.  We 
> > will have 14 different districts, every district would have 
> their own 
> > asterisk box on location.  We all have fiber lines running to a 
> > central location at our isd. This provides connectivity to 
> all the districts.
> > 1. would it be wiser to install a asterisk box at the 
> central location 
> > of the ISD and have all asterisk boxes register with that?
> 
> One asterisk box at each district/school would be wiser, 
> keeping local sip to sip calls on the LAN, and local 
> community calls on local pstn facilities.
> 
> There is no real need to "register" one asterisk to another. 
> Simply define a single entry in iax.conf using a type=user, 
> and an assoicated entry in the receiving * box with 
> type=peer. That will minimize the register keep- a-live 
> packets, etc. (That assumes your fiber network is very stable 
> in terms of uptime.)
> 
> > or would it be better to create a mesh-network of registrations. 
> 
> Create a mesh network without the registrations (as noted 
> above). The amount of traffic between districts is likely to 
> be rather low given that a large percentage of communications 
> is likely centered are each districts "community of 
> interest". (eg, local calls, room to office calls, etc.)
> 
> > If so can this be acomplished easily using iax or dundi? 
> 
> Yes, with iax very simple. No need for dundi in this case.
> 
> > What would be the best way to link all of these boxes 
> together so that 
> > from any location they can call each other using simple extenion 
> > headers such as _10xxx for district 1, and _11xxx for 
> distrct two and 
> > make that conssistant?
>  
> Personally, I'd look at setting up your dialplan in such a 
> way as to use real 7-digit or 10-digit telephone numbers in 
> all cases. If school district
> #8 has local telephone number like 312-456-1000 (assuming no 
> DID numbers), then use something like that in each remote 
> school district's asterisk dialplan. So, if someone in 
> district #12 dials 312-456-1000, the dialplan determines 
> whether to send that call via iax to district #8 or overflow 
> onto the pstn network. If you already have an extension 
> number scheme in each school, the above can be augmented with 
> something like 312-456-1xxx (even though they aren't DID's right now).
> 
> 
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RE: [Asterisk-Users] Asterisk initialization

2006-01-07 Thread Alexander Lopez



Do not know what version you are 
running,
 
But there are a few ways to do 
this.
 
There is a persistant setting:
 
from agents.conf
;; Define whether callbacklogins should be stored in 
astdb for; persistence. Persistent logins will be reloaded after; 
Asterisk restarts.;persistentagents=yes
If you want to handle it outside of Asterisk via an AGI you can have your 
AGI execute:
 
 AgentCallbackLogin([AgentNo][|[options][|[EMAIL PROTECTED]):
 
this 
is providing that you have the information saved in your DB.
 
 
Personal Opinion:
 
Use 
the builtin features with the persistentagents options and use the php script in 
the contribs directory to see who is on.
 
 
 
 


  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Dov 
  BigioSent: Friday, January 06, 2006 4:24 PMTo: 
  asterisk-users@lists.digium.comSubject: [Asterisk-Users] Asterisk 
  initialization
  
  Hi,
   
  I am doing an AGI that logs to a database every 
  Agent login/logoff.
  My idea is to be able to go to this database and 
  check which agents where logged so that I can force their login in case 
  Asterisk goes down for some reason.
   
  The problem is that I would need to reload their 
  status from this AGI when Asterisk initializes. Is there a way to do 
  this?
   
  One idea I had was to make safe_asterisk to 
  generate a .call file that calls and extension that would call the AGI to log 
  all the agents back on.
   
  Is there another way of running an AGI on 
  initialization?
   
  Thank you
  Dov
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Re: [Asterisk-Users] Non-PRI T1

2006-01-07 Thread Jerry Jones
Any type of circuit available as an analog line is also available  
over a T1. It just minimizes the amount of copper required to deliver  
service.


You must look at you original order from your telephone company to  
determine the type of circuits they are delivering. They may be POTS  
1FB in which case there are no digitis. They may be E&M with none. Or  
they may be DID which would be sending digits. Although only the  
order would identify the exact number of digits. If they are DID then  
you were probably assignen at least one block of 20 numbers to be  
delivered on the circuit.


Of course then you need to know how many trunks/circuits you have  
active on the T1. For outgoing calls you control which ones to use  
for any given call. for incoming they may be divided into multiple  
trunk groups with different numbers coming in on different channels.


If they are non DID circuits then they should hit the s exten in  
whatever context you have defined for them in zapata.conf on an  
incoming call.


Good luck


On Jan 6, 2006, at 6:02 PM, O'Connor, Jonathan wrote:


You can use a normal T1.  I have one between an Asterisk box and a
Vodavi switch.

I use 10 channels between them with E&M signalling.  Zaptel.conf:
loadzone= us
defaultzone = us
span=1,1,0,esf,b8zs
#bchan=1-10 # set this to 1-15,17-31 for E1
#dchan=24 # set this to 16 for E1
e&m=1-10
#fxsgs=1-10

Then Zapata.conf has:
immediate=no
overlapdial=yes
switchtype=national
signalling=em_w
emdigitwait=1000
channel=>1-10


You have to be careful of timing and such, because basically its  
really
no fancier then a set of lines that happen to be on a T1.  It opens  
and
line and literally sends 4 DTMF codes to the server to tell it  
where its
calling.  For all intents and purposes it could be a bundle of 10  
normal

phone lines.


It is our only site using this method because they wanted a fortune  
for

a PRI card.  Our Avaya and Nortel PBXs all talk to their Asterisk PBXs
using PRI, MUCH easier.  The basic T1 signalling and EM Wink is truly
horrid to work with when the PBX is as dumb as a box of rocks like our
Vodavi system.

-Jonathan

Jonathan O'Connor
Senior System Administrator
Inoveris LLC
Direct Line (614) 791-3742
Fax (614) 791-3748
Helpdesk 866-456-1566





-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Rusty Shackleford
Sent: Friday, January 06, 2006 6:54 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Non-PRI T1


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
Phillips
Sent: Friday, January 06, 2006 3:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Non-PRI T1


Are they configured for inbound calls? If so how?

Usually the telco sends the last 4 digits of the called

phone number

down the line.


Uhm, don't you need PRI signalling for that?

--
No virus found in this outgoing message.
Checked by AVG Free Edition.
Version: 7.1.371 / Virus Database: 267.14.14/222 - Release Date:
01/05/2006


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Re: [Asterisk-Users] RE: Fax, txfax -bizarre thing

2006-01-07 Thread Andrew Nowrot
Hi,>I certainly don't want to integrate fax<->e-mail support into spandsp.I think our problem is not connected with spandsp and fax <-> email integration. All the applications I mean spandsp txfax and rxfax are enough to have emial <-> fax functionality in Asterisk. I wrote a program which allows me to convert emial to tiff, it is also not a problem to turn Asterisk to be a "small" email server (I use qmail). Only problem is with the txfax (or I don't with what, maybe with synchronization to my Telco). When Asterisk launch the txfax application sometimes (quite often) something cause it to hangup the Zap channel and fax transmission goes to  the space!. 
On http://soft-switch.org/spandsp_faq/ar01s08.html   I found that this can be cause by the synchronization problem, correct me if I'm wrong. But there is a little, that I can do about it. What should I change to be synchronized with Telco (linux version, hardware or.)
Of course that's not a problem to use hylafax, but I just want to have it on one machine (I'm afraid that Asterisk and hylafax won't run on the same machine :( )And one more thing. I use fax machine connected 
to TDM400P  I can receive and send faxes without problems. I get crazy things Best wishesAndrew Nowrot



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Re: [Asterisk-Users] transfer application

2006-01-07 Thread Matt Riddell (IT)

Bill Michaelson wrote:
I am having trouble understanding how to use this.  I want to transfer 
certain incoming calls from an IAX ITSP based on caller ID.  From what I 
can make of the docs, I thought I need to do something like this...


exten => _NXXNXX,n(nocid),transfer(1000)
exten => _NXXNXX,n,noop(boo,${TRANSFERSTATUS})
exten => _NXXNXX,n,hangup


exten => 1000,1,Dial(IAX2/jnctn_out/16665551234,45,t)
exten => 1000,n,hangup


Why don't you just do:

exten => _NXXNXX,n(nocid),Dial(IAX2/jnctn_out/16665551234,45,t)

--
Cheers,

Matt Riddell
___

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Re: [Asterisk-Users] RE: Fax, txfax -bizarre thing

2006-01-07 Thread Rich Adamson

> I having similar problem. Unfortunately each thread is archive leads to 
> nowhere. I 
read a post in which similar problem was solved by changing rxgain and txgain 
to 15. 
Maybe this would help.
> Does anyone have common problems?
> 
> I was wondering why asterisk - great telecommunication program - has such a 
> weak fax 
support. I am talking about mail to fax and fax to mail. Or maybe I am the only 
one who 
has the problems with it.
> 
> If someone has some experience please help.

In addition to what others have already mentioned, fax support requires
excellent * timing & bus characteristics, and the digium tdm card does not 
provide
that (nor does the older x100p). Those that have spandsp running are primarily
T1/E1 users.

Running fax over sip channels (and networks) also requires rather tight
requirements in terms of latency, jitter, etc, that has been discussed
many times on this list. 


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Re: [Asterisk-Users] Help Connecting server districts

2006-01-07 Thread Rich Adamson

> I am working on a project to unite several local school districts.  We will 
> have 14 different districts, every district would have their own asterisk box 
> on location.  We all have fiber lines running to a central location at our 
> isd. This provides connectivity to all the districts. 
> 1. would it be wiser to install a asterisk box at the central location 
> of the ISD and have all asterisk boxes register with that? 

One asterisk box at each district/school would be wiser, keeping local
sip to sip calls on the LAN, and local community calls on local pstn
facilities.

There is no real need to "register" one asterisk to another. Simply define
a single entry in iax.conf using a type=user, and an assoicated entry in
the receiving * box with type=peer. That will minimize the register keep-
a-live packets, etc. (That assumes your fiber network is very stable in
terms of uptime.)

> or would it be better to create a mesh-network of registrations. 

Create a mesh network without the registrations (as noted above). The amount
of traffic between districts is likely to be rather low given that a large
percentage of communications is likely centered are each districts 
"community of interest". (eg, local calls, room to office calls, etc.)

> If so can this be acomplished easily using iax or dundi? 

Yes, with iax very simple. No need for dundi in this case.

> What would be the best way to link all of these boxes together so that 
> from any location they can call each other using simple extenion headers 
> such as _10xxx for district 1, and _11xxx for distrct two and make that 
> conssistant?
 
Personally, I'd look at setting up your dialplan in such a way as to use
real 7-digit or 10-digit telephone numbers in all cases. If school district
#8 has local telephone number like 312-456-1000 (assuming no DID numbers),
then use something like that in each remote school district's asterisk
dialplan. So, if someone in district #12 dials 312-456-1000, the dialplan
determines whether to send that call via iax to district #8 or overflow
onto the pstn network. If you already have an extension number scheme in
each school, the above can be augmented with something like 312-456-1xxx
(even though they aren't DID's right now).


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Re: [Asterisk-Users] Draytek Vigor 2900 & Asterisk

2006-01-07 Thread Jonathan Attwood
It certainly does.

How many rules can you create in the port forwarding section of the V2900?

I was told that the V2900 has SIP_ALG. Is this something you've activated?

On 1/7/06, Faris Raouf <[EMAIL PROTECTED]> wrote:
> Jonathan Attwood wrote:
> > I'm in conversation with Draytek's pre-sales dept..
> >
> > Here's the most recent reply:
> >
> >  >
> > We really don't know of anyone who has run an Asterisk server on
> > a Vigor2900. There are doubtless people around, but it's relatively
> > rare. Most people don't run SIP servers.
> >
> > Regards,>
> >
> > All I want to know is, if I buy one of these routers, will it break my setup
> > or not - ie. assuming I set up the relevant port-forwarding, can I
> > expect any one-way audio issues. Can't get a definitive answer from
> > suppliers or the manufacturer, so I hope someone here uses this model
> > with Asterisk.?
> > ___
>
> I have a 2900G behind a Cisco 1720 with dual ADSL WICs in one office,
> and a 2600VGi standalone in another (I don't use the 2600's built-in FXS
> ports -- they aren't very good - seem noisy).
>
> I have Asterisk servers in both offices, linked via IAX. I have incoming
>  voip services going independently to both Asterisk servers.
>
> I've had no problems whatsoever -- everything has worked perfectly. The
> QoS facility in both routers allows you to reserve a certain amount of
> bandwidth (in or out) for IAX and SIP and this seems to work fine though
> it isn't necessary on our networks.
>
> I'm using port forwarding on both routers to route IAX and SIP to the
> private IPs of the Asterisk boxes.
>
> But you will need to open the appropriate ports on the firewall in the
> router, or firewall the Asterisk boxes and DMZ the Asterisk boxes.
>
> However, the new Dreytek 3300 series of routers is even more
> interesting. Multiple WAN ports for backup/load balancing, and optional
> hardware FXO/FXS ports.
>
> I hope this helps.
>
> Faris.
>
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Re: [Asterisk-Users] RE: Fax, txfax -bizarre thing

2006-01-07 Thread Darren Nickerson

"Lee Howard" <[EMAIL PROTECTED]> wrote:


Use IAXmodem and HylaFAX, and then you have a fax server.

http://sourceforge.net/projects/iaxmodem
http://hylafax.sourceforge.net/


If you're looking for more general information on HylaFAX, see 
www.hylafax.org.


-Darren 


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Re: [Asterisk-Users] pattern matching in dialplan problems matching _NNN

2006-01-07 Thread Thomas

thanks... 

 _NXX works for me

best regards

Thomas


On Saturday 07 January 2006 16:37, Peter Bowyer wrote:
> On 07/01/06, Thomas <[EMAIL PROTECTED]> wrote:
> > Hi,
> >
> > I have a problem with pattern matching N what should digit 2 to 9
> > in Asterisk 1.2.1.
> >
> > If I dial 220 I did not get an PlayBack of invalid. Asterisk jumps into
> > the context dialout and find there an matching _2. and is using this.
> >
> > If I change _NNN to _XXX everything works fine. If I dial 220 I hear
> > playtones invalid. It seemed to be that pattern matching with N is not
> > working as designed.
>
> 220 isn't supposed to match _NNN - N is digits 2-9, not 0.
>
> http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Patterns
>
> Peter
>
> --
> Peter Bowyer
> Email: [EMAIL PROTECTED]
> Tel: +44 1296 768003
> VoIP: sip:[EMAIL PROTECTED]
> VoIP: [EMAIL PROTECTED]
> FWD: **275*5048707000
> VoipTalk: **473*5048707000
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Re: [Asterisk-Users] Draytek Vigor 2900 & Asterisk

2006-01-07 Thread Faris Raouf

Jonathan Attwood wrote:

I'm in conversation with Draytek's pre-sales dept..

Here's the most recent reply:



All I want to know is, if I buy one of these routers, will it break my setup
or not - ie. assuming I set up the relevant port-forwarding, can I
expect any one-way audio issues. Can't get a definitive answer from
suppliers or the manufacturer, so I hope someone here uses this model
with Asterisk.?
___


I have a 2900G behind a Cisco 1720 with dual ADSL WICs in one office, 
and a 2600VGi standalone in another (I don't use the 2600's built-in FXS 
ports -- they aren't very good - seem noisy).


I have Asterisk servers in both offices, linked via IAX. I have incoming 
 voip services going independently to both Asterisk servers.


I've had no problems whatsoever -- everything has worked perfectly. The 
QoS facility in both routers allows you to reserve a certain amount of 
bandwidth (in or out) for IAX and SIP and this seems to work fine though 
it isn't necessary on our networks.


I'm using port forwarding on both routers to route IAX and SIP to the 
private IPs of the Asterisk boxes.


But you will need to open the appropriate ports on the firewall in the 
router, or firewall the Asterisk boxes and DMZ the Asterisk boxes.


However, the new Dreytek 3300 series of routers is even more 
interesting. Multiple WAN ports for backup/load balancing, and optional 
hardware FXO/FXS ports.


I hope this helps.

Faris.

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Re: [Asterisk-Users] RE: Fax, txfax -bizarre thing

2006-01-07 Thread Steve Underwood

andrutto wrote:


Hi,

I having similar problem. Unfortunately each thread is archive leads to 
nowhere. I read a post in which similar problem was solved by changing rxgain 
and txgain to 15. Maybe this would help.
Does anyone have common problems?

I was wondering why asterisk - great telecommunication program - has such a 
weak fax support. I am talking about mail to fax and fax to mail. Or maybe I am 
the only one who has the problems with it.

If someone has some experience please help.
 

There are patents related to fax to e-mail and e-mail to fax. These are 
probably US only patents, but the picture isn't very clear. Some large 
users of Asterisk and spandsp have been chased by the patent holders. In 
light of this, I certainly don't want to integrate fax<->e-mail support 
into spandsp.


Regards,
Steve

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Re: [Asterisk-Users] pattern matching in dialplan problems matching _NNN

2006-01-07 Thread Peter Bowyer
On 07/01/06, Thomas <[EMAIL PROTECTED]> wrote:
> Hi,
>
> I have a problem with pattern matching N what should digit 2 to 9
> in Asterisk 1.2.1.
>
> If I dial 220 I did not get an PlayBack of invalid. Asterisk jumps into the
> context dialout and find there an matching _2. and is using this.
>
> If I change _NNN to _XXX everything works fine. If I dial 220 I hear playtones
> invalid. It seemed to be that pattern matching with N is not working as
> designed.

220 isn't supposed to match _NNN - N is digits 2-9, not 0.

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

Peter

--
Peter Bowyer
Email: [EMAIL PROTECTED]
Tel: +44 1296 768003
VoIP: sip:[EMAIL PROTECTED]
VoIP: [EMAIL PROTECTED]
FWD: **275*5048707000
VoipTalk: **473*5048707000
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Re: [Asterisk-Users] RE: Fax, txfax -bizarre thing

2006-01-07 Thread Lee Howard

andrutto wrote:


I was wondering why asterisk - great telecommunication program - has such a 
weak fax support.



Because it's a PBX and not a fax server.

Use IAXmodem and HylaFAX, and then you have a fax server.

http://sourceforge.net/projects/iaxmodem
http://hylafax.sourceforge.net/

Lee.
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Re: [Asterisk-Users] Problens to link 2 * servers

2006-01-07 Thread Moises Silva
network problems. Asterisk wan unable to connect or bind to 10.0.0.121/100

Regards
On 1/7/06, Cleyverson P. Costa <[EMAIL PROTECTED]> wrote:
>
> Hello,
>
> I'm traying to link 2 * servers using SIP and the following errors was show:
>
> "SIP/AsteriskA:[EMAIL PROTECTED]/100") in new stack
> Dec 13 22:46:57 WARNING[8767]: chan_sip.c:1398 create_addr: No such
> host: 10.0.0.121/100
> Dec 13 22:46:57 NOTICE[8767]: app_dial.c:759 dial_exec: Unable to
> create channel of type 'SIP'
>   == Everyone is busy/congested at this time
> Dec 13 22:47:07 NOTICE[8767]: rtp.c:435 ast_rtp_read: RTP: Received
> packet with bad UDP checksum
> Dec 13 22:47:07 WARNING[8767]: pbx.c:1949 ast_pbx_run: Timeout, but no
> rule 't' in context 'internal'
> linux*CLI>
>
> Someone could help me to fix the following problens??
>
> fraternatly,
>
>
> Cleyverson Pereira Costa
> 
> Phone #: 27+9922-0111
> Skype: cleyverson
> MSN: [EMAIL PROTECTED]
> 
> "Escolhe, pois, a vida, para que vivas, tu e a tua descendência."
> Deuteronômio 30:19
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>
>


--
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Re: [Asterisk-Users] Voice mail messages aren't sent to e-mail

2006-01-07 Thread James Armstrong

I added:

mailcmd=/usr/bin/sendmail -f hostname -t

to the voicemail.conf file under [general]

- James

On Jan 6, 2006, at 10:31 PM, Pisac wrote:


Yes, I found that this is problem with my server. Second server is
connected through second provider, and first server and my domain is
hosted at fist provider. My (first) provider has some stupid logic  
that

reject e-mails from mailservers which don't have public hostname but
private (my second server has "server.local"), but accepting all e- 
mails

from it's IP address space (first server).

So, my temporary solution was that I set up fake (but existing)  
hostname

for second server (gmail.com), and now my (first) provider accepting
e-mails. Very stupid.

How you changed mailcmd to add a -f? Did you used nail/mail instead of
sendmail, in voicemail.conf? Or maybe some .c source changing?

Thanks
Pisac




I had a similar problem, but I was able to see the message getting
rejected to rr.com because they were looking up the hostname pbx
and rejecting it. I changed the mailcmd to add a -f realhostname.com
and it started working.

- James


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[Asterisk-Users] pattern matching in dialplan problems matching _NNN

2006-01-07 Thread Thomas
Hi,

I have a problem with pattern matching N what should digit 2 to 9
in Asterisk 1.2.1.

If I dial 220 I did not get an PlayBack of invalid. Asterisk jumps into the 
context dialout and find there an matching _2. and is using this. 

If I change _NNN to _XXX everything works fine. If I dial 220 I hear playtones 
invalid. It seemed to be that pattern matching with N is not working as 
designed.



Any ideas?

best regards
Thomas



extensions.conf
-
[internal]
#include "/etc/asterisk/x_internal.conf"


x_internal.conf
-

exten => 210,1,Macro(internalsqldial_stand,${EXTEN})

exten => _NNN,1,NoOp(wrong number dialed)
exten => _NNN,n,PlayBack(invalid)
exten => _NNN,n,Hangup()

include => dialout



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[Asterisk-Users] RE: Fax, txfax -bizarre thing

2006-01-07 Thread andrutto

Hi,

I having similar problem. Unfortunately each thread is archive leads to 
nowhere. I read a post in which similar problem was solved by changing rxgain 
and txgain to 15. Maybe this would help.
Does anyone have common problems?

I was wondering why asterisk - great telecommunication program - has such a 
weak fax support. I am talking about mail to fax and fax to mail. Or maybe I am 
the only one who has the problems with it.

If someone has some experience please help.

Best wishes

Andrutto 


--
Kliknij po wiecej! >>> http://link.interia.pl/f18ed

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RE: [Asterisk-Users] 3RD REQUEST - Any Help Is Appreciated

2006-01-07 Thread Steve Totaro
> >-- Executing Goto("Zap/2-1", "ext-local|*101|1") in new stack

I think the problem here is that you have the timeout set to one second
and I am not sure what the * is before 101.

My interpretation is "ext-local" specifies local context. *101 means
dial extension 101 but I am unsure of what the * is for. And the final 1
means a one second timeout.

-or-

I don't know.

Thanks,
Steve
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Re: [Asterisk-Users] wich IAX soft client allow to specify a different server port?

2006-01-07 Thread Mikael Magnusson
On Sat, Jan 07, 2006 at 01:19:34PM +0100, Antonio Gallo wrote:
> I still having problem with remote SIP client,
> trying to use IAX client instead but i've to
> specify TCP port 8080 (because of firewall).

The IAX protocol is based on UDP, not TCP.

> 
> I did this on server in bindport=8080 in iax.conf
> 
> but i cannot find a soft client that allow to set wich
> server port to use.
> 
> Any idea?

Iaxcomm should work. You can use a complete dial string with username,
secret, peer, port number, extension and context if you like, in the 
following format.

[[:[EMAIL PROTECTED][:][/[@]]

/Mikael
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RE: [Asterisk-Users] Dialer

2006-01-07 Thread Steve Totaro
Darren,

I am interested in your project.  Let me know if I can help you test.

Thanks,
Steve

> -Original Message-
> From: Wiley Siler [mailto:[EMAIL PROTECTED]
> Sent: Friday, January 06, 2006 12:53 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Dialer
> 
> If this or any other example is available, I would be most thankful to
> have it.
> 
> I got the go ahead on this project to day so now I have to start
seeing
> how to do this.
> 
> Thanks,
> Wiley
> 
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Darren
> Wiebe
> Sent: Tuesday, January 03, 2006 5:00 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Dialer
> 
> I'm supposed to have a "mostly" canned script that will do this done
> already.  It will pull the list of people to call out of a db and play
> them the file specified in the db table.  Contact me offlist if you're
> interested.  It will be done real soon but I'm not done testing yet.
> 
> Darren Wiebe
> [EMAIL PROTECTED]
> 
> Kerry Garrison wrote:
> 
> > You actually aren't far from it. If the system only needs to play
the
> > same file to each person, a simple script can be used to pull from a
> > database and create call files. Asterisk will use the call files to
> > place the calls and play a sound. A few minutes of searching on that
> > should get you started. I haven't seen anyone else have a canned
> > script ready to go, but would like to know if anyone does.
> > -Kerry
> >
> >
> >
>

> > *From:* [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] *On Behalf Of
> > *Wiley Siler
> > *Sent:* Tuesday, January 03, 2006 3:32 PM
> > *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> > *Subject:* [Asterisk-Users] Dialer
> >
> > Hello All,
> >
> > I am having trouble finding a specific * piece of software so I
> > thought I would see If you guys can help me get my terminology
> clear.
> >
> > First off let me premise this with "no, this is absolutely not
for
> > doing call marketing".
> > I need to make my Asterisk box call a group of people and play
> > them a message.
> > My company deals with education so we need to do follow ups if
> > students are not logging on.
> > We do this manually now but it would be easier and cheaper to
just
> > play them a message.
> >
> > What is the term I should be looking for?  I keep thinking "auto
> > dialer" or something like that but I am not quite getting there.
> >
> > Any help would be appreciated.  I have been learning a bit of
Perl
> > so I was thinking I could auto generate and AGI file and then
just
> > do a Play() of the mp3 when they pick up at the other end?
Seems
> > a little kludge though.
> >
> >
> > Thanks,
> > Wiley
> >
> >
>
>---
> >-
> >
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> >
> >
> 
> 
> --
> Darren Wiebe
> [EMAIL PROTECTED]
> Aleph Communications
> ASTPP - Open Source Voip Billing & Calling Cards
www.aleph-com.net/astpp
> 
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RE: [Asterisk-Users] Dialer

2006-01-07 Thread Steve Totaro
Yes, I would be very interested in this as well.

Thanks,
Steve

> -Original Message-
> From: Wiley Siler [mailto:[EMAIL PROTECTED]
> Sent: Friday, January 06, 2006 4:53 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Dialer
> 
> Very cool!  Is this something you can share the code?
> 
> Thanks,
> Wiley
> 
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of trixter
> aka Bret McDanel
> Sent: Friday, January 06, 2006 2:17 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Dialer
> 
> On Fri, 2006-01-06 at 11:45 -0700, Wiley Siler wrote:
> > Just to make it easy, I will be reading the caller list from a
another
> 
> > server via a web page, parsing it and dialing.
> > After each pass, I just post back to the server web page and it
> > updates the other system.
> > Our tech just needs to review the log once daily.
> 
> That is basically what I did for a customer.  I have a DB that is
> filtered pursuant to 47 CFR 64.1200 and 16 CFR 310 (US federal laws
> concerning these types of systems -- not calling to the US, dont worry
> about it).  I wrote some tools to make that a snap.  I then have 1-N
> clients pull from the DB servier via HTTP to get the next number to
dial
> and context to goto.  The dialplan updates the DB via HTTP so the
status
> of a given number is known and prevents duplicate calls.
> 
> I added answering machine detection to my asterisk server and a few
> other things to make the dialing slightly better.
> 
> The way it works they can have many many calling systems if they need,
> nothing has to be local to each other.  Reports can be generated off
any
> data that is available (timestamps of events, status of calls, etc).
> 
> This is perfect for dr appt reminders, batch calls saying 'your
product
> has been shipped' etc.
> --
> Trixter http://www.0xdecafbad.com Bret McDanel
> UK +44 870 340 4605   Germany +49 801 777 555 3402
> US +1 360 207 0479 or +1 516 687 5200
> FreeWorldDialup: 635378
> http://www.sacaug.org/ Sacramento Asterisk Users Group
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Re: [Asterisk-Users] Re: Cell phone dock/switch as Asterisk FXO source

2006-01-07 Thread Brian McEntire
No Bluetooth in the Samsung T309. I couldn't think of why I'd want
BT... then of course I started looking at cell sockets, etc. after I
got it and found several do not have a cable for the T309 yet. In
hindsight, bluetooth would have made this easier. Live and learn! 
On 1/6/06, Jonathan Attwood <[EMAIL PROTECTED]> wrote:
On 1/5/06, Brian McEntire <[EMAIL PROTECTED]> wrote:> Wow! Thanks for all the responses! Very informative.>> Erik: I'm just looking for simple dial-out and pass-along incoming cell
> calls to *. Looks like the doc-n-talk should do it, except I checked with> them and, silly me, the new Samsung t309 phone I just got is not supported> yet. Hopefully it will be in a few months.>
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RE: [Asterisk-Users] wich IAX soft client allow to specify a differentserver port?

2006-01-07 Thread kevin ling
Try this.
http://www.virbiage.com/firefly/download/firefly-thirdparty.exe 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Antonio Gallo
Sent: Saturday, January 07, 2006 8:20 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] wich IAX soft client allow to specify a
differentserver port?

I still having problem with remote SIP client, trying to use IAX client
instead but i've to specify TCP port 8080 (because of firewall).

I did this on server in bindport=8080 in iax.conf

but i cannot find a soft client that allow to set wich server port to use.

Any idea?

Thanks, Antonio
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RE: [Asterisk-Users] Draytek Vigor 2900 & Asterisk

2006-01-07 Thread kevin ling
Now draytek have some SIP embeded router  (e.g., 2100VG, 2900VG...). Maybe
you can try these new router. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sergio
Chersovani
Sent: Saturday, January 07, 2006 9:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Draytek Vigor 2900 & Asterisk

Jonathan Attwood wrote:

>I'm in conversation with Draytek's pre-sales dept..
>  
>
I bought a 2600 2 years ago and I had alot of NAT problem, because the SPI
was changing the externhost (sip.conf) ip address with the local private
address forwarding the packets, so the audio stream was failing.

I sent all the debug logs to the draytek dev team, but they were slow on
updates to I bought a new and different brand router.
Hope they fixed that issue in the new firmwares

Good luck

Sergio
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RE: [Asterisk-Users] Draytek Vigor 2900 & Asterisk

2006-01-07 Thread kevin ling
Hi,

Draytek 2900 is a great router. Easy to setup & stable. I want known more
detail of your network & configuration. I can setup it and make some test.

Regards,
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jonathan
Attwood
Sent: Saturday, January 07, 2006 9:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Draytek Vigor 2900 & Asterisk

I'm in conversation with Draytek's pre-sales dept..

Here's the most recent reply:



All I want to know is, if I buy one of these routers, will it break my setup
or not - ie. assuming I set up the relevant port-forwarding, can I expect
any one-way audio issues. Can't get a definitive answer from suppliers or
the manufacturer, so I hope someone here uses this model with
Asterisk.?
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Re: [Asterisk-Users] Draytek Vigor 2900 & Asterisk

2006-01-07 Thread Sergio Chersovani

Jonathan Attwood wrote:


I'm in conversation with Draytek's pre-sales dept..
 

I bought a 2600 2 years ago and I had alot of NAT problem, because the 
SPI was changing the externhost (sip.conf) ip address with the local 
private address forwarding the packets, so the audio stream was failing.


I sent all the debug logs to the draytek dev team, but they were slow on 
updates to I bought a new and different brand router.

Hope they fixed that issue in the new firmwares

Good luck

Sergio
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[Asterisk-Users] Draytek Vigor 2900 & Asterisk

2006-01-07 Thread Jonathan Attwood
I'm in conversation with Draytek's pre-sales dept..

Here's the most recent reply:



All I want to know is, if I buy one of these routers, will it break my setup
or not - ie. assuming I set up the relevant port-forwarding, can I
expect any one-way audio issues. Can't get a definitive answer from
suppliers or the manufacturer, so I hope someone here uses this model
with Asterisk.?
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[Asterisk-Users] Problens to link 2 * servers

2006-01-07 Thread Cleyverson P. Costa



Hello,
 
I'm traying to link 2 * servers using SIP and the 
following errors was show:
 
"SIP/AsteriskA:[EMAIL PROTECTED]/100") 
in new stackDec 13 22:46:57 WARNING[8767]: chan_sip.c:1398 create_addr: No 
such host: 10.0.0.121/100Dec 13 22:46:57 NOTICE[8767]: app_dial.c:759 
dial_exec: Unable to create channel of type 'SIP'  == Everyone is 
busy/congested at this timeDec 13 22:47:07 NOTICE[8767]: rtp.c:435 
ast_rtp_read: RTP: Received packet with bad UDP checksumDec 13 22:47:07 
WARNING[8767]: pbx.c:1949 ast_pbx_run: Timeout, but no rule 't' in context 
'internal'linux*CLI> 
 
Someone could help me to fix the following 
problens??
 
fraternatly,
 
 
Cleyverson Pereira 
CostaPhone #: 
27+9922-0111Skype: cleyversonMSN: [EMAIL PROTECTED]"Escolhe, 
pois, a vida, para que vivas, tu e a tua descendência."Deuteronômio 
30:19
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Re: [Asterisk-Users] SPA-3000 is translating vocal sounds into DTMF

2006-01-07 Thread Rich Adamson

> I am having the same problem with a male voice at the other end.
> 
> It is making the spa3k problem for me.
> 
> Has this been reported to SIPURA ?
> Is this a common problem ?
> has anyone done been able to make this happen less often ?
> 
> 
> I would hope perhaps there's some kind of setting that has to do with
> the way it detects inband DTMF?  I'm pretty sure it's an artifact of
> this particular ATA; my other SIP devices are just fine.
> 
>  
>  
> FWIW, I have the same prob on my spa3k. It's peircing and pisses my 
> wife off greatly.
>  

Its a fairly common problem with the spa3k, but is somewhat dependent on
how it is configured and the distance to your CO.

If the spa3k is configured for ring thru (from the fxo to fxs without passing
through asterisk), the call remains within the spa3k. Its echo cancellation
and tone detection functions are not as good as they could be, and it seems
to have more problems with longer length pstn lines then it does with
shorter lines (eg, distance between the Central Office and the spa3k).

Reducing the audio levels on the pstn/fxo port seems to help somewhat, but
you'll reach a point where the audio is rather low when the issues have
been minimized. (In other words, the spa3k does not seem to have a very
good dynamic range.)

Seems the v3.1.7g firmware is worse then some of their earlier releases.

Linksys/Sipura support typically suggests very elementary things like check
to ensure the impedence is set correctly for your telco line (etc), and
then something like try v3.1.5 if you still have problems. I'd love to
escalate the issue to level 2 support (assuming that even exists).


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[Asterisk-Users] wich IAX soft client allow to specify a different server port?

2006-01-07 Thread Antonio Gallo

I still having problem with remote SIP client,
trying to use IAX client instead but i've to
specify TCP port 8080 (because of firewall).

I did this on server in bindport=8080 in iax.conf

but i cannot find a soft client that allow to set wich
server port to use.

Any idea?

Thanks, Antonio
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Re: [Asterisk-Users] Annoying Notice Message: "Don't know what to do with control frame 15"

2006-01-07 Thread tim panton
On 6 Jan 2006, at 16:28, Joan Bautista wrote:Hi, I haven't found anything about the message below  on the mailing list, Does anyones knows why this notice is being appearing?  -- Executing Dial("Local/[EMAIL PROTECTED],2", "IAX2/CallOut/12365533643|30|otT") in new stack    -- Called CallOut/12365533643     -- Call accepted by 12.11.11.11 (format ulaw)    -- Format for call is ulaw    -- IAX2/10.11.240.110:4569-3 is proceeding passing it to  Local/[EMAIL PROTECTED],2Jan  6 13:20:41 NOTICE[26911]: channel.c:2416 __ast_request_and_dial: Don't know what to do with control frame 15    -- IAX2/10.11.240.110:4569-3 is circuit-busy    -- Hungup 'IAX2/12.11.11.11:4569-3'   == Everyone is busy/congested at this time (1:0/1/0)    -- Executing Goto("Local/[EMAIL PROTECTED],2", "s-CONGESTION|1") in new stack     -- Goto (default,s-CONGESTION,1)    -- Executing NoOp("Local/[EMAIL PROTECTED],2", ""CONG"") in new stack    -- Executing Congestion(" Local/[EMAIL PROTECTED],2", "") in new stack   > Channel Local/[EMAIL PROTECTED],1  was never answered.  == Spawn extension (default, s-CONGESTION, 2) exited non-zero on 'Local/[EMAIL PROTECTED],2'   My calling scenario is like this: server01 server02 pstn server  <--IAX trunking--> agents/sip server server01: Asterisk 1.2.1server02: Asterisk 1.2.1Well, according to the iax RFC, control frame 15 (0xf) means 'call is Proceeding', I guess that  the NOTICE is just telling you that asterisk can't do anything usefulwith that info, it doesn't sound like a problem to me.Tim. http://www.westhawk.co.uk/  ___
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Re: [Asterisk-Users] Asterisk initialization

2006-01-07 Thread pdhales



There is a sample php script in the contribs folder 
that shows who is logged in - one of my clients uses it.
 
PaulH

  - Original Message - 
  From: 
  Dov Bigio 

  To: asterisk-users@lists.digium.com 
  
  Sent: Saturday, January 07, 2006 8:24 
  AM
  Subject: [Asterisk-Users] Asterisk 
  initialization
  
  Hi,
   
  I am doing an AGI that logs to a database every 
  Agent login/logoff.
  My idea is to be able to go to this database and 
  check which agents where logged so that I can force their login in case 
  Asterisk goes down for some reason.
   
  The problem is that I would need to reload their 
  status from this AGI when Asterisk initializes. Is there a way to do 
  this?
   
  One idea I had was to make safe_asterisk to 
  generate a .call file that calls and extension that would call the AGI to log 
  all the agents back on.
   
  Is there another way of running an AGI on 
  initialization?
   
  Thank you
  Dov
  
  

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Re: [Asterisk-Users] re: where can i find all .C files

2006-01-07 Thread trixter aka Bret McDanel
On Sat, 2006-01-07 at 00:26 -0800, Tejas Shah wrote:
> hi all,
> 
>   i m using debian to run my asterisk
> gateway.I want to make some customization in voicemail
> application.For that i need to modify voicmail's 
> .C(source file) file. can any body tell me where
> exactly all .C files resides in the system..
> 
> thanks
> 
> tejas
> 

if you have the deb-src in your sources.lkist then you can install
asterisk-source which will be in /usr/src somewhere..

If you want to build from source you can install subversion and get it
straight from digium, for 1.2 here are some directions..

svn checkout http://svn.digium.com/svn/asterisk/branches/1.2
asterisk-1.2
svn checkout http://svn.digium.com/svn/zaptel/branches/1.2 zaptel-1.2
svn checkout http://svn.digium.com/svn/libpri/branches/1.2 libpri-1.2
svn checkout http://svn.digium.com/svn/asterisk-addons/branches/1.2
asterisk-addons-1.2


cd zaptel-1.2 && make clean install && \
cd ../libpri-1.2 && make clean install && \
cd ../asterisk-1.2 && make clean install


obviously make any changes between the svn checkout and the make :)


-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378
http://www.sacaug.org/ Sacramento Asterisk Users Group


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Re: [Asterisk-Users] GSM Gateway / Terminal for sale

2006-01-07 Thread pdhales
I have used both Telular analog units and Voiceblue SIP units in Australia.

PaulH

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

Sent: Saturday, January 07, 2006 1:40 AM
Subject: Re: [Asterisk-Users] GSM Gateway / Terminal for sale


> Is anyone aware of the details of this in Australia?
>
> I'd love to be able to let tech's have calls route straight to their
> mobiles when 'in-house'
>
> Steve Kennedy wrote:
>
> >On Fri, Jan 06, 2006 at 01:23:26PM -, Chris Bagnall wrote:
> >
> >
> >
> >>>I don't get it. What is the advantage of using a GSM gateway?
> >>>VOIP calls are pretty inexpensive as they are now.
> >>>
> >>>
> >>It largely depends on the country you're calling. Here in the UK, calls
to
> >>mobiles are maintained at an artificially high rate because the
terminating
> >>network (the mobile networks) get a cut of call revenue for calls *to*
your
> >>mobile. By contrast, in the US, the mobile customer often pays a small
> >>charge per minute on incoming calls (as I understand the market over
there).
> >>You'll also find in the UK the mobile phone market is heavily subsidized
by
> >>the networks such that you can get phones for free if you sign up to 12
> >>month contracts. I often find that it's cost-effective to get a new
contract
> >>every 12 months (with a free phone), even if I don't want the phone.
Flog
> >>the phone on ebay and you've got a spare SIM with lots of inclusive
minutes
> >>for almost nothing.
> >>
> >>
> >
> >In the UK the wholesale rates are set by Ofcom (like the FCC), which
> >works out about 7p'ish per minute.
> >
> >However the operators can offer retail bundles (including phones) and
> >for a monthly contract they "throw" in various ammounts of cross network
> >minutes (or free to their own network or whatever). With clever
> >dial-plans and multiple terminals connected to multiple networks you can
> >generally get "free" calls to mobile users (basically clever least cost
> >routing, time of day sometimes needs to be taken into account as well).
> >
> >However there are some disadvantages, the main being you cant set CLI of
> >the outgoing call as it will always be tied to the SIM of the mobile
> >terminal.
> >
> >Another is that you can NOT run a GSM gateway (as they're known) for 3rd
> >parties. So if you want to connect your office PBX to a gateway to make
> >use of cheap mobile termination for your own company that's fine, but as
> >an ITSP (or traditional telco) you can not allow 3rd party traffic to
> >utilise a gateway. If networks find you are using a gateway (as a telco)
> >they can cut it off, no questions asked. Gateways have been determined
> >to be fixed infrastructure, therefore NOT mobile.
> >
> >There is (or maybe was by now) an Ofcom consultation asking whether this
> >should be changed, the mobile operators will fight it, telcos and other
> >users will be asking for it to be changed.
> >
> >Of course this is UK specific, other countries have more lenient
> >policies (I think Belgium allow gateways, France doesn't allow any kind,
> >and some allow them with the co-operation of the operators).
> >
> >
> >Steve
> >
> >
> >
>
> -- 
> Adrian Carter
> Technical Manager
> Leading Edge Internet
>
> Web   http://www.lei.net.au http://support.lei.net.au
> Direct+61 2 6163 6162  Support 1 300 662 415
> E-mail[EMAIL PROTECTED]
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[Asterisk-Users] re: where can i find all .C files

2006-01-07 Thread Tejas Shah

hi all,

  i m using debian to run my asterisk
gateway.I want to make some customization in voicemail
application.For that i need to modify voicmail's 
.C(source file) file. can any body tell me where
exactly all .C files resides in the system..

thanks

tejas



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