I keep the AGI in Git as a version control system. But, you can view the AGI
source here:
http://messinet.com/trac/browser/gv/gv.agi
And at the very bottom of that page is a link to download it as an individual
file here:
On Thu, Oct 28, 2010 at 10:11 AM, Stephen Reese rsre...@gmail.com wrote:
Is there a way to prevent Google Chat from staying logged in but still
be able to dial outbound? People think I'm logged in persistently and
send me messages that I miss. Even if I set a status message in
asterisk most
On Tuesday, October 26, 2010 01:16:29 pm Stephen Reese wrote:
http://messinet.com/trac/wiki/AsteriskGVGateway (AGI script)
Is your .agi and .git the same script? I do not have a git client on
this host to see for myself.
I keep the AGI in Git as a version control system. But, you can view
Since Google Voice (GV) doesn't let us connect diretly via SIP, IAX2, etc.,
for outbound calls, it acts basically like a fancy click-to-call application.
So...
You need Asterisk to login into GV, and initiate the call. GV will dial
the number you tell it to, then connect it to one of your
On Mon, Oct 25, 2010 at 12:50 AM, Anthony Messina amess...@messinet.com wrote:
On Sunday, October 24, 2010 05:23:13 pm Stephen Reese wrote:
Evening,
Has anyone seen a how-to on getting Asterisk to work with Google Talk
and Google Voice?
Thanks
For Google Voice, I use an ipKall number for
On Monday, October 25, 2010 07:30:22 am Stephen Reese wrote:
Does the AGI have to be used? In this example
http://www.davidvossel.com/?p=28 I see mention of a script, but not in
this one:
http://blog.polybeacon.com/2010/10/17/asterisk-1-8-and-google-voice/
I believe I missing the connection
Evening,
Has anyone seen a how-to on getting Asterisk to work with Google Talk
and Google Voice?
Thanks
--
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
On Sun, Oct 24, 2010 at 6:23 PM, Stephen Reese rsre...@gmail.com wrote:
Has anyone seen a how-to on getting Asterisk to work with Google Talk
and Google Voice?
I wrote one last week:
http://blog.polybeacon.com/2010/10/17/asterisk-1-8-and-google-voice/
Also: http://www.davidvossel.com/?p=28
--
On Sun, Oct 24, 2010 at 7:06 PM, Paul Belanger
paul.belan...@polybeacon.com wrote:
On Sun, Oct 24, 2010 at 6:23 PM, Stephen Reese rsre...@gmail.com wrote:
Has anyone seen a how-to on getting Asterisk to work with Google Talk
and Google Voice?
I wrote one last week:
On Sun, Oct 24, 2010 at 9:24 PM, Stephen Reese rsre...@gmail.com wrote:
On Sun, Oct 24, 2010 at 7:06 PM, Paul Belanger
paul.belan...@polybeacon.com wrote:
On Sun, Oct 24, 2010 at 6:23 PM, Stephen Reese rsre...@gmail.com wrote:
Has anyone seen a how-to on getting Asterisk to work with Google
On Sunday, October 24, 2010 05:23:13 pm Stephen Reese wrote:
Evening,
Has anyone seen a how-to on getting Asterisk to work with Google Talk
and Google Voice?
Thanks
For Google Voice, I use an ipKall number for the inbound trunk. Here are the
relevant sections of my extensions.conf:
;
Deat all,
I am in middle of integrate Asterisk with Toshiba astrata legacy pbx.
Following is my setup
*Asterisk - Digium TE110P - E1 card in toshiba pbx - Toshiba PBX*
A = B
C D
Asterisk PBX and strata
: [asterisk-users] Integrating asterisk with Toshiba Astrata DK380
Deat all,
I am in middle of integrate Asterisk with Toshiba astrata legacy pbx.
Following is my setup
Asterisk - Digium TE110P - E1 card in toshiba pbx - Toshiba PBX
A = B
C
I'll give strace a try on Monday and see if I can figure that out. In any case, that's not a huge deal right now - I can bind anonymously for now and get the information out, and I'm not terribly concerned about support LDAP writes from Asterisk - I'd just like to get the configuration read out
(Got a quarantine notice on the first one, so I'm resending - sorry if this ends up a duplicate...)
I'll give strace a try on Monday and see if I can figure that out. In any case, that's not a huge deal right now - I can bind anonymously for now and get the information out, and I'm not terribly
On Thu, 21 Sep 2006, Nick Couchman wrote:
When I try to set the port to 636 in the res_ldap.conf file, I get bind
errors (Can't contact server...). I imagine this is an issue with
certificates and trust, but I'm not exactly sure where I need to put my
CA certificate in order to make the ldap
Hi All:
Im starting to jump into the Asterisk world and try to figure out a VoIP solution for my company. I stumbled across the VoiceRD company/project which is supposed to integrate Asterisk into Novell eDirectory via LDAP. Unfortunately the project is in its very
Hi Steve,
I notice that everytime I command a reload on the
asterisk my unicall channels gets also reset. So all
going calls got cut-off. Any suggestion that I should
do avoid this problem?
Regards,
Leonimar
--- Steve Underwood [EMAIL PROTECTED] wrote:
Thierry Querette wrote:
Hi
Carlos Chavez wrote:
I am trying to get my Asterisk server to talk to a Panasonic D500 PBX
using an E1 connection. The card for the Panasonic uses MFC/R2 and I
have installed Unicall. Calls from the Asterisk server to the Panasonic
go through without a hitch and I can call any
Thierry Querette wrote:
Hi Carlos,
I had the same problem and spent a lot of time studying the MFC/R2
protocol but the problem is in the libmfcr2 package version!!
Try using the packages in:
http://www.soft-switch.org/downloads/unicall/unicall-0.0.3pre7
And not in pre9.
Both pre7
About the ANI problem, in Brazil I use the following parameter for protocolvariant.protocolvariant=br,20,16,16I have the following configuration:Astetrisk E1 - PBXE1Telco
But Steve, after changing versions it really started to work without any modification in the .conf files. It
I am trying to get my Asterisk server to talk to a Panasonic D500 PBX
using an E1 connection. The card for the Panasonic uses MFC/R2 and I
have installed Unicall. Calls from the Asterisk server to the Panasonic
go through without a hitch and I can call any extension I want. The
problem
I had a similar problem with a Siemens, most probably you are
specifying the wrong number of expected ANI digits. Try with mx,0,4
as protocolvariant, that will tell Unicall to expect 0 ANI digits, but
of course, in Asterisk environment you wont be able to get callerid.
Play around incrementing
Hi Carlos,I had the same problem and spent a lot of time studying the MFC/R2 protocol but the problem is in the libmfcr2 package version!!Try using the packages in:
http://www.soft-switch.org/downloads/unicall/unicall-0.0.3pre7And not in pre9.Both pre7 and pre9 have
libmfcr2-0.0.3.tar.gz
Have either of you any experience integrating Asterisk-related devices into existing phone equipment (trunks/pots lines, etc. (I'm somewhat new to legacy voip based telephony)) to ensure
specific or random outbound calls route through Asterisk vs bell company (ATT)? Thanks in advance,Dakota
On Jun 3, 2006, at 12:53 PM, Dakota Burns wrote:
Have either of you any experience integrating Asterisk-related devices
into existing phone equipment (trunks/pots lines, etc. (I'm somewhat
new to legacy voip based telephony)) to ensure specific or random
outbound calls route through
What I was attempting to visualize is the following case: 10 people in an organization pick-up their phones to make an outbound call. Before integrating Asterisk, all calls route through their current non-VoIP based phone provider. After integrating 1 trunk from a VoIP service provider into their
Dakota Burns wrote:
What I was attempting to visualize is the following case:
10 people in an organization pick-up their phones to make an outbound
call. Before integrating Asterisk, all calls route through their
current non-VoIP based phone provider. After integrating 1 trunk
from a VoIP
On Jun 3, 2006, at 2:04 PM, Dakota Burns wrote:
What I was attempting to visualize is the following case:
10 people in an organization pick-up their phones to make an outbound
call. Before integrating Asterisk, all calls route through their
current non-VoIP based phone provider. After
Just be sure that if you ditch your POTS line that you have a proper
way to terminate 911 calls!
On 6/3/06, Martin Joseph [EMAIL PROTECTED] wrote:
On Jun 3, 2006, at 2:04 PM, Dakota Burns wrote:
What I was attempting to visualize is the following case:
10 people in an organization pick-up
Hello List!
I would like to integrate a Asterisk box in my current (german) telephone
setup. Right now it looks like this:
Provider -- DSL/ISDN Splitter -- Telephone-System(Box) or TK-Anlage ;)
I have read that you can put Asterisk between my Splitter and the
Telephone-System-Box, so that for
[EMAIL PROTECTED] schrieb:
Hello List!
I would like to integrate a Asterisk box in my current (german) telephone
setup. Right now it looks like this:
Provider -- DSL/ISDN Splitter -- Telephone-System(Box) or TK-Anlage ;)
I have read that you can put Asterisk between my Splitter and the
I have been successful in setting up asterisk and making workstation to
workstation SIP calls. But I am lost when it comes to anything past that.
We are trying to integrate this asterisk server into with our Executone
(432?) PBX to allow us to make outbound SIP calls between our disparate
Correction:
The hardware is a Wildcard T100P (not a TE110P)
Thanks!
-Original Message-
From: Geoff Manning [mailto:[EMAIL PROTECTED]
Sent: Wednesday, May 18, 2005 9:07 AM
To: Asterisk Users (E-mail)
Subject: [Asterisk-Users] Integrating Asterisk into our Legacy PBX
--Newb
I
Hi all.
Does anyone integrated a Siemens Hicom 300E with Asterisk using FXO
interfaces!?
I created an extension group in Hicom and connected my 4FXO(TDM04B) into the
telefony internal network.
What issues I have to care about it!?
Thanks for any help!
Regards,
Denis.
Hello,
i'm looking for informations in integrating Asterisk
to existing TDM-based PBX (particularly Siemens
HiPath4000/Hicom300E) similar to the document you can
find on www.pham.org/asterisk/asterisk-meridian-a1.pdf
for Nortel.
Unfortunately the page
http://www.voip-info.org/wiki-Siemens+Hicom
I am hoping to implement an asterisk system between two offices as below
|---|
| PBX |-(XXX)--[asterisk]---[VPN][asterisk]--(FXO)--phone
|---| |
|-(FXS)--PSTN
I am unsure which interface I
On Sat, 2003-03-29 at 16:43, Richard Scobie wrote:
I am hoping to implement an asterisk system between two offices as below
|---|
| PBX |-(XXX)--[asterisk]---[VPN][asterisk]--(FXO)--phone
|---| |
I have done exactly taht usingdigium x100p cards... :)
Very nice indeed...
On Sun, 30 Mar 2003, Richard Scobie wrote:
I am hoping to implement an asterisk system between two offices as below
|---|
| PBX |-(XXX)--[asterisk]---[VPN][asterisk]--(FXO)--phone
|---|
They are availeable now.
On Sunday 30 Mar 2003 00:57, Richard Scobie shaped the electrons to say:
Steven Critchfield wrote:
You have 2 options here for the (XXX) section. Option 1 is to see if
your panasonic pbx supports a T1 interface, and is afordable for the
number of lines you wish to
Michael Bielicki wrote:
They are availeable now.
That's great news although the website still says ...upcoming release...
Richard
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