Re: [asterisk-users] How determine extension of who initiated call
${CALLERID(num)} is a channel variable. Read here about it:
http://www.voip-info.org/wiki/index.php?page=Asterisk+variables
What kind of script is it? AGI? Language?
Regards,
Prince Singh
http://www.drishti-soft.com
On Sat, Jul 25, 2009 at 3:29 AM, Philipp Kempgen
philipp.kemp...@amooma.dewrote:
Michelle Dupuis schrieb:
I'm working on a script that needs to determine the extension (eg: 123)
of
the phone that initiated the call, or CALLERID number if an externall
caller.
Is there a simple way to do this?
${CALLERID(num)} ?
Philipp Kempgen
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Re: [asterisk-users] how to match no callerid in 1.6 ?
On Fri, Jul 24, 2009 at 11:14:47AM +0200, Philipp Kempgen wrote:
Louis-David Mitterrand schrieb:
On Fri, Jul 24, 2009 at 10:37:38AM +0200, Michiel van Baak wrote:
On 10:17, Fri 24 Jul 09, Louis-David Mitterrand wrote:
This used to work fine in 1.4:
exten = 2131/,1,NoOp(reject3: ${CALLERID(num)})
exten = 2131/,n,Playback(no_unknow_callerid_here)
exten = 2131/,n,Hangup
And now, after upgrading to 1.6.1.x it matches every callerid.
Why remove the elegant and minimal exten/emtpy
notation
Not that need the exten/callerid syntax for anything but I'd say
this is a bug and a regression.
The syntax is exten[/callerid] so the / clearly says that there
is a second argument even if that happens to be an empty string.
Dear asterisk devs: should I file a bug report? (exten/,prio
matching all callerid's)
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[asterisk-users] how to remove MWI from a Polycom phone
Hi, I'd like to disable MWI on certain lines of my IP650 Polycom phone. So I removed the mailbox= parameter from that line's peer section in sip.conf. Yet the envelope still appears in front of that line and the phone MWI keeps blinking. Where should I look to completely disable MWI on a certain line? Thanks, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TLS Manager
On 25/07/09 00:08, John A. Sullivan III wrote: Hello, all. After many pages of googling and testing in the lab, I'm still a bit perplexed about how to implement tls protection for the asterisk manager. manager.conf allows one to specify the cert file but one normally must also specify the private key file. If I simply enter the cert file: sslenable=yes sslbindport=5038 sslbindaddr=172.x.x.8 sslcert=/etc/pki/tls/certs/pbxc.pem ; path to the certificate. ; sslcipher=cipher string It errors as I expect it would: pbx*CLI manager reload == Parsing '/etc/asterisk/manager.conf': == Found SSL cert error/etc/pki/tls/certs/pbxc.pem How does one specify the private key for the manager.conf file? Thanks - John Not quite the same thing I know, but it might help. I use stunnel for the AMI so the connection is transported in a SHH tunnel. It's quite easy to setup. Alan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Reasons to use AEL (was: Re: Goto from a feature macro is not working?)
Miguel Molina schrieb:
Philipp Kempgen escribió:
Use macros in AEL so you don't have to care about the underlying
implementation. :-) scnr
Right now for every implementation I made, I didn't have the need to
program in AEL, only plain extensions, some AMI and AGI. But well, it
seems to have a lot of advantages. Please tell me some, I may take a
look to it too see if it's worth spending the time to learn and get the
best out of it.
I'd say control structures (and proper indentation) are one of the
most important reasons to use AEL (conditionals: if .. else, switch
.. case, ..., loops: for, while) because they look so familiar.
Imagine nested control structures in extensions.conf with Goto(),
GotoIf(), While(), EndWhile(), ExitWhile(), ContinueWhile() and
priorities - such code is not what I call maintainable.
== extensions.conf:
exten = 30,1,Set(x=5)
exten = 30,n,While($[${x} = 9])
exten = 30,n,NoOp(x ist ${x})
exten = 30,n,ExecIf($[${x} 5],ExitWhile)
exten = 30,n,Playback(beep)
exten = 30,n,Set(x=$[${x} + 1])
exten = 30,n,EndWhile()
exten = 30,n,NoOp(done)
== extensions.ael:
30 = {
x=0;
while (${x} = 9) {
NoOp(x ist ${x});
if (${x} 5) {
break;
}
Playback(beep);
y=${x} + 1;
}
NoOp(done);
}
In this example, we needed more lines in AEL; if we had added another
command to the if condition in our while loop, ExecIf() would not be
enough anymore and we would be forced to use a more complex construc-
tion with GotoIf(). Our extensions.conf would be a lot longer.
Philipp Kempgen
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Re: [asterisk-users] Asterisk CSTA
Thanks Steve. But I couldn't find anything about a CSTA to AMI gateway for asterisk at the quintum site. All I be able to find are TDM and FXO/FXS to SIP gateways among others. I'm talking about (and I think gergis.rasmy too) 3rd party call control gateways, not interoperable gateways. By the way, what about an open source csta gateway project? Jose 2009/7/24 Steve Totaro stot...@totarotechnologies.com Without any research, I would check out the Quintum lineup. They have feature sets which are amazing (and confusing as all heck) and work great once configured. Thanks, Steve T On Fri, Jul 24, 2009 at 1:03 PM, Jose Arias cyr2...@gmail.com wrote: Do you know what names those gateways have? Jose 2009/7/24 Olivier oza-4...@myamail.com 2009/7/22 gergis.rasmy gergis.ra...@gmail.com does Asterisk suppoet CSTA protocol for CTI applications? No it doesn't but I've heard some gateways exist (software translating CST to AMI). Regards -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to match no callerid in 1.6 ?
Philipp Kempgen wrote:
Louis-David Mitterrand schrieb:
On Fri, Jul 24, 2009 at 10:37:38AM +0200, Michiel van Baak wrote:
On 10:17, Fri 24 Jul 09, Louis-David Mitterrand wrote:
This used to work fine in 1.4:
exten = 2131/,1,NoOp(reject3: ${CALLERID(num)})
exten = 2131/,n,Playback(no_unknow_callerid_here)
exten = 2131/,n,Hangup
And now, after upgrading to 1.6.1.x it matches every callerid.
Why remove the elegant and minimal exten/emtpy
notation
Not that need the exten/callerid syntax for anything but I'd say
this is a bug and a regression.
The syntax is exten[/callerid] so the / clearly says that there
is a second argument even if that happens to be an empty string.
While this could be a bug and a regression, I don't see how using the
exten[/callerid] notation is really better than the GotoIf()
Personally, the GotoIf() makes much more sense to me, because you're placing
the
matching logic in a single place, as opposed to an error prone method of adding
an ending / at the end of every line of that extension.
Typically I try to get away from a pattern match as soon as I can, by doing
something like:
exten = _[A-Za-z0-9].,1,Set(EXTENSION=${EXTEN})
exten = _[A-Za-z0-9].,n,Goto(start,1)
exten = start,1,Verbose(2,Incoming call from ${CALLERID(num)} to extension
${EXTENSION})
exten = start,n,GotoIf($[${CALLERID(num)} = 5551212]?bad_callerid,1)
exten = start,n,...
exten = bad_callerid,1,Verbose(2,A very bad man!)
exten = bad_callerid,n,Hangup()
I think that is a better method than constantly typing a complex pattern match,
or adding additional extra characters that could potentially be missed, and
leading to additional debugging, or errors in dialplan.
Leif Madsen.
http://www.leifmadsen.com
http://www.oreilly.com/catalog/asterisk
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Re: [asterisk-users] how to match no callerid in 1.6 ?
Leif Madsen schrieb:
I don't see how using the
exten[/callerid] notation is really better than the GotoIf()
Personally, the GotoIf() makes much more sense to me, because you're placing
the
matching logic in a single place,
True.
as opposed to an error prone method of adding
an ending / at the end of every line of that extension.
exten = _[A-Za-z0-9].,1,Set(EXTENSION=${EXTEN})
exten = _[A-Za-z0-9].,n,Goto(start,1)
exten = start,1,Verbose(2,Incoming call from ${CALLERID(num)} to extension
${EXTENSION})
exten = start,n,GotoIf($[${CALLERID(num)} = 5551212]?bad_callerid,1)
exten = start,n,...
exten = bad_callerid,1,Verbose(2,A very bad man!)
exten = bad_callerid,n,Hangup()
I think that is a better method than constantly typing a complex pattern
match,
BTW: If you hate having to type the same exten = stuff over
and over for each priority: use AEL and let the AEL compiler do the
work. :-)
_[A-Za-z0-9]. = {
Verbose(2,Incoming call from ${CALLERID(num)} to extension ${EXTEN});
if (${CALLERID(num)} = 5551212) {
Verbose(2,A very bad man!);
Hangup();
}
...
}
And while we're at it: In many cases database lookups by means of
DB() or AGI() or a custom ODBC_*() function make even more sense
than a hard-coded list of GotoIf()s resp. if clauses.
Philipp Kempgen
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[asterisk-users] DeadAgi application issue
Hi All, I have working asterisk 1.4.24.1, but I have issues with DeadAgi application. I am using hylafax and iaxmodem with asterisk, mail 2 fax and fax 2 mail feature. My system details are below: OS: Centos 5.3 Asterisk Version: 1.4.24.1 Dahdi version: dahdi-linux-2.1.0.4, dahdi-tools-2.1.0.2 Zap device: Network controller: Sangoma Technologies Corp. A104d QUAD T1/E1 AFT card Kernel: 2.6.18-128.1.10 We have used HylaFAX+ for the mail 2 fax and fax2mail feature. We have setup incoming and outgoing phpagi scripts for the calculation for fax and billing too. But when we are sending and receiving the faxes it is working fine, But some times the deadagi application got stuck the channels. And becuase of that the phpagi scripts are not completed. In that case we need to restart asterisk complusary. We are using zap lines for the outbound and inbound faxes. Can any one suggest solution for this? Thanks in advance!!! Thanks, Max Alex Voip Developer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on OpenWRT
On Fri, 24 Jul 2009 21:27:12 -0400, David Cook wrote: I have installed them on a Linksys WRT54GL or WRT54GS v4/v3/v2/v1.1 devices. My mother-in-law's runs fine and she doesn't notice the difference. I know that is very subjective but to be honest I never looked at it for more than home-use/1 line applications. Can't say I've had a problem that caused me to look at its load level transcoding. I can tell you she has been on the phone and received VM at the same time so there are two concurrent sessions. It means she keeps her number even though she moved to a retirement home that is out-of-area-code so she's more than happy. Plus calling between us is traditional 10-digit dialing although it is a SIP trunk - not that she (or my family) notice any difference. Yeah, I really like the embedded systems approach to Asterisk, but there's a limit. Some hardware is just too constrained IMHO. OTOH, have you been following what David Rowe and Village Telco has been doing with the Mesh Potato? That's extremely cool stuff. Michael -- Michael Graves mgravesatmstvp.com http://www.mgraves.org o713-861-4005 c713-201-1262 sip:mgra...@mstvp.onsip.com skype mjgraves Twitter mjgraves ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to match no callerid in 1.6 ?
2009/7/24 Louis-David Mitterrand vindex+lists-asterisk-us...@apartia.org:
This used to work fine in 1.4:
exten = 2131/,1,NoOp(reject3: ${CALLERID(num)})
exten = 2131/,n,Playback(no_unknow_callerid_here)
exten = 2131/,n,Hangup
And now, after upgrading to 1.6.1.x it matches every callerid.
I'm not sure if it's the same reason, but have a look at this bug
(exists in 1.6.1.1):
https://issues.asterisk.org/view.php?id=15476
Whether you want to use the function, or a pattern match in version
1.6, you might want to upgrade past revision 206705. Especially if
you're trying to detect no callerid. Otherwise you'll get a wrong
result. (assuming you're using SIP)
HTH
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Re: [asterisk-users] Reasons to use AEL
Philipp Kempgen philipp.kemp...@amooma.de writes: Miguel Molina schrieb: Philipp Kempgen escribió: Use macros in AEL so you don't have to care about the underlying implementation. :-) scnr Right now for every implementation I made, I didn't have the need to program in AEL, only plain extensions, some AMI and AGI. But well, it seems to have a lot of advantages. Please tell me some, I may take a look to it too see if it's worth spending the time to learn and get the best out of it. I'd say control structures (and proper indentation) are one of the most important reasons to use AEL (conditionals: if .. else, switch .. case, ..., loops: for, while) because they look so familiar. Imagine nested control structures in extensions.conf with Goto(), GotoIf(), While(), EndWhile(), ExitWhile(), ContinueWhile() and priorities - such code is not what I call maintainable. Can you recommend a good tutorial or book that covers AEL? I tend to use extensions.conf because most of the examples I come across use it, and it's covered by my first edition of the O'Reilly Asterisk book. Do later editions of the O'Reilly book cover AEL thoroughly? Thanks, Scott. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reasons to use AEL
On Saturday 25 July 2009 10:53:48 Scott Gifford wrote: Philipp Kempgen philipp.kemp...@amooma.de writes: Miguel Molina schrieb: Philipp Kempgen escribió: Use macros in AEL so you don't have to care about the underlying implementation. :-) scnr Right now for every implementation I made, I didn't have the need to program in AEL, only plain extensions, some AMI and AGI. But well, it seems to have a lot of advantages. Please tell me some, I may take a look to it too see if it's worth spending the time to learn and get the best out of it. I'd say control structures (and proper indentation) are one of the most important reasons to use AEL (conditionals: if .. else, switch .. case, ..., loops: for, while) because they look so familiar. Imagine nested control structures in extensions.conf with Goto(), GotoIf(), While(), EndWhile(), ExitWhile(), ContinueWhile() and priorities - such code is not what I call maintainable. Can you recommend a good tutorial or book that covers AEL? I tend to use extensions.conf because most of the examples I come across use it, and it's covered by my first edition of the O'Reilly Asterisk book. Do later editions of the O'Reilly book cover AEL thoroughly? Not as of this date. We're looking at covering AEL in the 3rd edition, though. -- Tilghman Teryl with Peter, Cottontail, Midnight, Thumper, Johnny (bunnies) and Harry, BB, George (dogs) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reasons to use AEL
Scott Gifford schrieb: Can you recommend a good tutorial or book that covers AEL? http://www.das-asterisk-buch.de/2.1/extensions.ael.html has some examples but unfortunately the explanations are in German. :-) voip-info has some examples as well: http://www.voip-info.org/wiki/view/Asterisk+AEL http://www.voip-info.org/wiki/view/Asterisk+AEL2 Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Test Function if SIP Device is Still Alive
Hello! Thank you for the information. Regarding using the sip show peers command, I remember somewhere seeing that it only works for static sip accounts and does not list accounts that are dynamically stored in a database. Most of my accounts are database entries, so would the sip show peers command work? Thanks, Elliot On Thu, Jul 23, 2009 at 5:08 PM, Ishfaq Maliki...@pack-net.co.uk wrote: Hi You can retrieve it in real time using the AMI from a script http://www.voip-info.org/wiki/view/Asterisk+manager+API Ish Elliot Murdock wrote: Hello Philipp, Thank you. I could set that up, but is that status (of qualifying) stored anywhere (besides the log files) that a script could use? Regards, Elliot On Thu, Jul 23, 2009 at 12:47 PM, Philipp Kempgenphilipp.kemp...@amooma.de wrote: Elliot Murdock schrieb: I am looking for a way to test if a SIP device is still alive or not. What about qualify=yes in sip.conf? I want to add this functionality in an AGI or independent script in order ensure all the SIP phones are properly connected to the system. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Test Function if SIP Device is Still Alive
Elliot Murdock schrieb: Regarding using the sip show peers command, I remember somewhere seeing that it only works for static sip accounts and does not list accounts that are dynamically stored in a database. Most of my accounts are database entries, so would the sip show peers command work? Yes it does work, at least with rtcachefriends=yes in sip.conf. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Audiocodes MP114, 2xFXS, @xFXO - does any one have configuration files they can share for trixbox?
I have an MP114 2fxs,2fxo which I would like to use with Trixbox, does anyone have a setup file they can share to help me work this out. Instructions or a link I can follow - thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry abount Asterisk extensions.conf
Dear John The peer switch is Huawei switch and we need this functionality as we support ISDN PRI but it supports for V5 interface . So another softswitch is functioning between us . The peer side expects to receive the subscriber dialed digits one-by-one as he sees us as an access equipment (not just an PBX) . Regards H.Motamedi On Wed, Jul 22, 2009 at 12:53 PM, John Novack jnov...@stromberg-carlson.org wrote: Curious - Why? What is the peer switch and why does it have this requirement? John Novack hadi motamedi wrote: Dear All Can you please let us know how we can modify our Asterisk extensions.conf file so it interprets the subscriber dialed digits in one-by-one digit manner . At its current configuration , it interprets them in an whole packet . I mean , say the subscriber dials as 665 so we need Asterisk to send it to the peer switch as 6,6,5,0,0,0,0 but not as one 665 packet . Your reply is very welcome Regards H.Motamedi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Dog is my co-pilot ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry abount Asterisk extensions.conf
Dear Leif Can you please provide us with more details on this Overlap Dialing phillosophy ? Regards H.Motamedi On Wed, Jul 22, 2009 at 1:15 PM, Leif Madsen leif.mad...@asteriskdocs.orgwrote: John Novack wrote: Can you please let us know how we can modify our Asterisk extensions.conf file so it interprets the subscriber dialed digits in one-by-one digit manner . At its current configuration , it interprets them in an whole packet . I mean , say the subscriber dials as 665 so we need Asterisk to send it to the peer switch as 6,6,5,0,0,0,0 but not as one 665 packet . Curious - Why? What is the peer switch and why does it have this requirement? That's a funny way of answering the question :) I *think* what he wants is overlap dialing. Leif Madsen. http://www.leifmadsen.com http://www.oreilly.com/catalog/asterisk ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry abount Asterisk extensions.conf
http://www.voip-info.org/wiki/view/Asterisk+Extension+Matching http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns http://wiki.snom.com/Settings/overlap_dialing Hope this helps - John On Sun, 2009-07-26 at 05:07 +0100, hadi motamedi wrote: Dear Leif Can you please provide us with more details on this Overlap Dialing phillosophy ? Regards H.Motamedi On Wed, Jul 22, 2009 at 1:15 PM, Leif Madsen leif.mad...@asteriskdocs.org wrote: John Novack wrote: Can you please let us know how we can modify our Asterisk extensions.conf file so it interprets the subscriber dialed digits in one-by-one digit manner . At its current configuration , it interprets them in an whole packet . I mean , say the subscriber dials as 665 so we need Asterisk to send it to the peer switch as 6,6,5,0,0,0,0 but not as one 665 packet . Curious - Why? What is the peer switch and why does it have this requirement? That's a funny way of answering the question :) I *think* what he wants is overlap dialing. Leif Madsen. http://www.leifmadsen.com http://www.oreilly.com/catalog/asterisk ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
