Hello
Even the cheapest GSM gateway I found for a simple SOHO server costs
over $300.
If possible, I'd rather have a local solution than use a remote VoIP
provider to have Asterisk make outgoing calls to the GSM network.
So... I was wondering if there are some entry-level cellphones that
can
A user embedded an * in a Read command and it causes my AEL script to
fail.
Does anyone know how I could code to detect it?
Thanks.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15 Phoenix,
I've downloaded and installed Trixbox 2.8 (asterisk 1.6) ..I encounter 2
problems for dynamic agents login and logout -
1. When agent from sip phone dials *11 , he is prmpted to enter extension
number first - but if he feeds the extension number, asterisk doenst allow
him to
2009/9/21 Vincent vincent.delpo...@bigfoot.com
Hello
Even the cheapest GSM gateway I found for a simple SOHO server costs
over $300.
If possible, I'd rather have a local solution than use a remote VoIP
provider to have Asterisk make outgoing calls to the GSM network.
So... I was
Thanks for the answer.
At startup I get the following message :
[r...@asterisk asterisk]# dmesg | grep dahdi
dahdi: Telephony Interface Registered on major 196
dahdi: Version: 2.2.0.2
dahdi_vpmadt032_loader: module license 'Digium Commercial' taints kernel.
dahdi_transcode: Loaded.
There are FTCs available, which you can use to convert GSM to FXs line, and
then hook it to your asterisk PBX using FXo card.
GSM :---à PSTN :--àFXo card (asterisk box)
Regards
Vijay Gandhi
GIPL(An ISO 9001:2000 Company)
+91-9811688460
+44-2080992384
vi...@gandhiinfotech.com
In the 1.6.1.* branch the line type=peer seems to be required on each
user...
d
2009/9/19 Örn Arnarson o...@arnarson.net
Sorry I wasn't more specific.
The error message is just the standard 'Can't find that extension'.
The problem is, however, that asterisk parses users.conf (and doesn't
Check your VICIdial logs and try to debug the VICIdial side of things... It
could be something along the lines of agent hits hangup, web interface goes
to add hangup command into the manager queue, fails due to a lock on the
table so call stays up... This wouldn't be an asterisk issue...
You need
2009/9/21 Vijay Gandhi vi...@gandhiinfotech.com
There are FTC’s available,
What is it (a FTC) ? a cable ?
Any pointer to that (Google is helpless)? ?
which you can use to convert GSM to FXs line, and then hook it to your
asterisk PBX using FXo card.
GSM :---à PSTN :--àFXo card
Thanks Darrick, I will try to upgrade the version. Also I got to know that
if we are going to limit the call length as well as silence detection might
me useful for eradicating the channel lockup.
On Fri, Sep 18, 2009 at 11:52 PM, Darrick Hartman dhart...@djhsolutions.com
wrote:
das sandesh
Olivier schrieb:
2009/9/21 Vijay Gandhi vi...@gandhiinfotech.com
There are FTC’s available,
What is it (a FTC) ? a cable ?
Any pointer to that (Google is helpless)? ?
My guess would be fixed to cell or FX to cell adapter.
Chris
___
--
It is actually FCT, my mistake I wrongly typed in FTC.
FCT is Fixed Cellular Terminal, you can put your GSM card into it and it
gives you an output of a PSTN line (FXs) which can be connected to your FXo
device, normally in india, we get these devices for about $50 (USD Fifty
only).
Regards
There is also chan_sebi and chan_celliax.
I tried chan_mobile without success (too unstable). Those two channels above
are still in my pending list.
On Mon, Sep 21, 2009 at 8:54 AM, Vijay Gandhi vi...@gandhiinfotech.comwrote:
It is actually FCT, my mistake I wrongly typed in FTC.
FCT is
Vijay Gandhi wrote:
It is actually FCT, my mistake I wrongly typed in FTC.
FCT is Fixed Cellular Terminal, you can put your GSM card into it and it
gives you an output of a PSTN line (FXs) which can be connected to your FXo
device, normally in india, we get these devices for about $50 (USD
The FCT's in india, do support 4 bands which are standard for GSM.
Moreover I am not marketing any product, it was just a information I shared
since somebody on the list asked for it.
Regards
Vijay Gandhi
GIPL(An ISO 9001:2000 Company)
+91-9811688460
+44-2080992384
vi...@gandhiinfotech.com
2009/9/21 Vijay Gandhi vi...@gandhiinfotech.com
It is actually FCT, my mistake I wrongly typed in FTC.
FCT is Fixed Cellular Terminal, you can put your GSM card into it and it
gives you an output of a PSTN line (FXs) which can be connected to your FXo
device, normally in india, we get these
That's the only difference.
Regards
Vijay Gandhi
GIPL(An ISO 9001:2000 Company)
+91-9811688460
+44-2080992384
vi...@gandhiinfotech.com
www.gandhiinfotech.com
From: olivier.kr...@gmail.com [mailto:olivier.kr...@gmail.com] On Behalf Of
Olivier
Sent: Monday, September 21, 2009 8:31
Hello
According to this article, this nice little unit can only use the PSTN
port for outgoing calls (ie. as a backup in case the connection to the
VoIP provider stops working), but not incoming calls:
http://tinyurl.com/mwjmo8
Can someone confirm that Atcom made this strange decision, and that
On Fri, Sep 18, 2009 at 7:45 PM, sean darcy seandar...@gmail.com wrote:
My set up is 1.6.0.15 with the digium fax modules. I want to capture a
fax from the internal analog fax machine (using an SPA2102), and then
resend it.
Usually this is where somebody asks for fax passthrough support on a
Running 1.4.26.2 on CentOS 5.3
Using ODBC with mysql for voicemail storage. Everytime my dialplan tries to
open a connection to save a message or retrieve a stored message, Asterisk
dumps out and restarts with:
Asterisk ended with exit status 127
Asterisk died with code 127.
Automatically
On Monday 21 September 2009 01:46:55 pm Ekelund, Bryan wrote:
Running 1.4.26.2 on CentOS 5.3
Using ODBC with mysql for voicemail storage. Everytime my dialplan tries to
open a connection to save a message or retrieve a stored message, Asterisk
dumps out and restarts with:
Asterisk ended
I have seen lots of companies offering this as a service and have used
phonetag.com in the past.
They work very nicely, however I have a customer that is not
interested in paying $30-$40 a month but would rather buy the
software. I have googled and googled all I can come up with are
companies that
Upon further review, it is not dumping out, just restarting on its own with
the same error. No .dmp in /tmp
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher
Sent: Monday, September 21, 2009
Hi people, I'm trying to retrieve data from the database (server MySQL).
I have the following dial plan:
exten = s,1,Noop(Start)
exten = s,n,MYSQL(Connect connid localhost user pass asteriskcdrdb)
exten = s,n,Noop(Connid: ${connid})
...
The problem is that the 3º line is not showing the
Ekelund, Bryan escribió:
Upon further review, it is not dumping out, just restarting on its own with
the same error. No .dmp in /tmp
Check that you are running asterisk with the -g option.
Cheers,
--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center
Anahi Ludueña wrote:
...
The problem is that the 3º line is not showing the connid. How can I
know the error?
Why don't you give us some of the basics.
Version of Asterisk
Version of your Asterisk addons
A sanitized bit of the console output..
Doug
--
Ben Franklin quote:
Those who
Running with -f -vvvg -c
After recompiling with DONT_OPTIMIZE I no longer get:
Asterisk ended with exit status 127
Asterisk died with code 127.
Automatically restarting Asterisk.
mpg123: no process killed
but it sends an automatic restart, no core dump.
As soon as it hits
On Monday 21 September 2009 03:37:03 pm Anahi Ludueña wrote:
Hi people, I'm trying to retrieve data from the database (server MySQL).
I have the following dial plan:
exten = s,1,Noop(Start)
exten = s,n,MYSQL(Connect connid localhost user pass asteriskcdrdb)
exten = s,n,Noop(Connid:
Are there any enterprise Asterisk end-users going to VoiceCon?
(sorry, this is a situation where I'm interested in only in end-users
- no consultants or integrators.) Contact me off-list, please, thanks!
JT
---
John Todd email:jt...@digium.com
Digium, Inc. | Asterisk
In my attempts to set up digium fax I get an odd warning:
-- Executing [...@capture-fax:2] ReceiveFAX(SIP/173-b53023e8,
/var/spool/asterisk/fax/20090921_1806.tif) in new stack
-- Channel 'SIP/173-b53023e8' receiving fax
'/var/spool/asterisk/fax/20090921_1806.tif'
[Sep 21 18:06:37]
This needs work, but it's about right for both of the problems -
probably a cut command to filter out the actual extensions being dialled.
PaulH
exten = _x,1,Set(state=${SIPPEER(${EXTEN}:status)})
exten = _x,n,GotoIf($[${state:0:2}=OK]?online:offline)
exten - _x.,n(online),Dial(SIP/${EXTEN})
On Mon, Sep 21, 2009 at 9:26 PM, sean darcy seandar...@gmail.com wrote:
In my attempts to set up digium fax I get an odd warning:
-- Executing [...@capture-fax:2] ReceiveFAX(SIP/173-b53023e8,
/var/spool/asterisk/fax/20090921_1806.tif) in new stack
-- Channel 'SIP/173-b53023e8'
Trying to build oslec. Following dahdi-linux README I copy
drivers/staging/echo to dahdi-linux/drivers/staging. I uncomment the 2
oslec lines in drivers/dahdi/Kbuild.
That doesn't work:
/home/asterisk/build/dahdi/svn/dahdi-linux/drivers/dahdi/dahdi_echocan_oslec.c:33:35:
error:
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