Google multitech CallFinder 100, has both FXO and FXS interface you can
connect.
Problem is you call out and don't get simulated ring back while the GSM
call is being setup.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Totaro
Sent: Saturday, June
There is a commercial product that does just that. I cannot reveal
the company name since they are clients of mine but they have a BTS, a
retractable 15 foot tower, a laptop or small PC running Asterisk and
either do VoIP over VSAT or connect via T1/E1.
Mostly government work but they are busy an
I am thinking about using an existing asterisk box and turning it into a gsm
gateway. Has anyone tried this before, adding sonme gsm cards and an
antenna. Any ideas.
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Gentlemen,
I'm using 1.4.21 SVN Tag, and have the queues set up to use Realtime.
This system works fine with 1.2.28, and everything loads fine with no
errors, but when I log an agent in I see the extra message "(not in
use)" by their listing and they are not rang by asterisk when their
queue is
Can Realtime be used with OOH323 ala sip_buddies?
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OK, tried changing DTMF tone as described on URL and no difference
Bart
Steve, I fooled with dtmf mode and it was 2833 - However, got stranger
results with inband, seems it would take digits, but audio goes to 1 way
afterwards first push.
As far as changing the code per the URL, I think I get w
Jay R. Ashworth wrote:
> On Fri, Jun 20, 2008 at 03:42:28PM -0600, Arturo Ochoa wrote:
>
>>I have a customer who owns a little Motel, and he wants to upgrade to a
>>Asterisk PBX. There is one analog phone per room (aprox 80), and the cable
>>is CAT 3.
>>
>
> You might want to co
Steve, I fooled with dtmf mode and it was 2833 - However, got stranger
results with inband, seems it would take digits, but audio goes to 1 way
afterwards first push.
As far as changing the code per the URL, I think I get what's it doing, but
wonder what else would be effected afterwards - I gues
The scenario:
This is all done SIP with a VOIP provider (have to register to single IP)
We have two inbound DID numbers / Accounts.
We have to register each individually with the VOIP provider.
I'd like inbound from each registered account (DID)
to be able to come into a unique PEER or dialplan
On Fri, Jun 20, 2008 at 03:42:28PM -0600, Arturo Ochoa wrote:
>I have a customer who owns a little Motel, and he wants to upgrade to a
>Asterisk PBX. There is one analog phone per room (aprox 80), and the cable
>is CAT 3.
You might want to consider snagging an FXS channelbank off of eB
On Sat, Jun 21, 2008 at 12:11 PM, Barton Fisher <[EMAIL PROTECTED]> wrote:
> I place SIP DID call towards ZAP (TE410P). ZAP uses e&m signaling to an
> external IVR system. I can hear the asterisk sending the DTMFs properly
> toward ZAP at call setup. After the call connects, any further DTMF digits
I place SIP DID call towards ZAP (TE410P). ZAP uses e&m signaling to an
external IVR system. I can hear the asterisk sending the DTMFs properly
toward ZAP at call setup. After the call connects, any further DTMF digits
from SIP is very short sounding or distorted (barely audible) on the ZAP
and ig
> Yes, both Asterisk and Cisco are behind Nat.
My asterisk box is behind a dsl modem and router. All traffic is bridged
from the modem to the router. Here are the settings on the router;
http://dwabbott.com/pictures/port_forward.png
http://dwabbott.com/pictures/range_forward.png
The asterisk box i
Hi,
I'm having trouble connecting two Asterisk boxes via a IAX2 friend trunk.
"iax2 show peers" on both boxes seem to show that all's fine (Status OK on
qualify=yes peer).
voip1 is an Asterisk 1.2.27 production server.
voip2 is an Asterisk 1.4.21 experimental server in the same gigabit LAN.
If I
Have you tried keeping asterisk in on the call with a /n connection in
the dial-plan?
Is there any firewall that is blocking udp ports to any of your clients?
Fred Posner
[EMAIL PROTECTED]
On Jun 21, 2008, at 12:36 AM, Sam Tam wrote:
Well to be honest, our experience with asterisk never w
-- Forwarded message --
From: Arun Kumar Chaudhary <[EMAIL PROTECTED]>
Date: Sat, Jun 21, 2008 at 4:51 PM
Subject: Detection of Answer, hangup,busy etc while using Dial command
To: [EMAIL PROTECTED]
Hi Guys,
I am in kanpur, India.
I am using Dial() command in my phpagi script. I a
Hi All,
I am still working on an TAPI solution for my customer.
They are trying to connect Asterisk to Navision.
I am using the Activa TSP and an TAPI connector for Navision.
When a customer calls I use the following rule:
exten => s,n,Dial(LOCAL/11&LOCAL/991013,25,tTr)
991013 is an extension
Hi all,
can anybody tell me how I get asterisk calling an executable after a queue
call?
only setting MONITOR_EXEC and MONITOR_FILENAME does not work anymore!
We use normal monitor to record _in and _out files. does work perfectly.
But calling an exec does not do anything.
We also tried se
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