How can I find out, which queue member did answer a given call?
I wish, from within my dialplan (extensions.conf) to write a record
tying a given incoming call to the (possible) answering queue member.
However, I can't find any easy way to get that info, if not
using/intercepting AMI (not that
Hello All,
After a brief summer vacation, the Asterisk Jobs staff have returned and
are gearing up for an eventful fall season here in North America.
Asterisk Jobs (www.asterisk-jobs.com) has removed the free access for
new employers after a successful 4 month promotion. Asterisk Jobs will
contin
Yes, that's according to our telco's specs. 'D' is the
start character, followed by up to 12 digits
representing the number, then finally a 'C' which is
the stop signal. I'm almost sure that a modification
needs to be done in the callerid handler but have no
idea where and how to do it. yes, maybe
> > I am curious as to what hardware folks are using successfully from HP
> > or DELL. I will likely be running just a quad span T1 card with the
> > system.
>
> HP DL380 G4, 4GB mem, 2x 146GB U320 in RAID1, dual hotswap PS
> HP DL360 G4, 2GB mem, 2x 146GB U320 in RAID1, dual hotswap PS
>
> Some
On 8/20/06, Mario <[EMAIL PROTECTED]> wrote:
How can I find out, which queue member did answer a given call?
I wish, from within my dialplan (extensions.conf) to write a record
tying a given incoming call to the (possible) answering queue member.
However, I can't find any easy way to get that in
On Sun, Aug 20, 2006 at 01:17:07PM +0200, Woodoo People .pGa! wrote:
> btw: i prefer HP servers (above 3xx) because you can do health monitoring
> really nice (fans, temp, ps status, etc)
configure lm_sensors on just about any system built in the recent years
and you'll get those.
--
Tzafrir Co
BJ Weschke wrote:
On 8/20/06, Mario <[EMAIL PROTECTED]> wrote:
How can I find out, which queue member did answer a given call?
I wish, from within my dialplan (extensions.conf) to write a record
tying a given incoming call to the (possible) answering queue member.
However, I can't find any easy
On 8/20/06, Mario <[EMAIL PROTECTED]> wrote:
BJ Weschke wrote:
> On 8/20/06, Mario <[EMAIL PROTECTED]> wrote:
>> How can I find out, which queue member did answer a given call?
>>
>> I wish, from within my dialplan (extensions.conf) to write a record
>> tying a given incoming call to the (possibl
Does anything pop up on the Asterisk screen?
Does music on hold work fine?
PaulH
On Fri, 2006-08-18 at 13:13 +0800, Nathan Alberti wrote:
>
>
> Hi all,
>
> I am currently having issues with a Polycom 601 on a customers site
> with Asterisk 1.2.6 and sip firmware 1.6.7 (upgraded from
BJ Weschke wrote:
On 8/20/06, Mario <[EMAIL PROTECTED]> wrote:
BJ Weschke wrote:
> On 8/20/06, Mario <[EMAIL PROTECTED]> wrote:
>> How can I find out, which queue member did answer a given call?
>>
>> I wish, from within my dialplan (extensions.conf) to write a record
>> tying a given incoming c
For the ATA, I would
recommend Mediatrix. They have some enterprise grade FXS external gateway with
echo can.
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Lito Lampitoc
Envoyé : 20 août 2006 00:11
À : Asterisk
Users Mailing List - Non-Commercial Discussion
Obj
hi all
I have a few boxes in my setup, most connected to one or more PRIs.
Today, we're making the telco do the number filtering in the switch,
this number range to this PRI etc. This is all fine, but not too
flexible, as we're depended upon some switch monkey being available
to do the jo
Hi,
I have problem connecting Sangoma A101 with asterisk-1.2.10 supporting
MFC/R2 signalling. The incoming calls is received but this messages
written in /var/log/asterisk/mesages.localhost.localdomain. (My
asterisk-1.2.10 came from latest version of ScopServ
http://www.scopserv.com)
Aug 1
Olle,
I think he means to include other contexts into a specfic context. Though I
was thinking if asterisk would ever support just being able to use a context
in real time without having to have the context in extenssions.conf with a
switch statement. I created a system that requires new contex
Hi,
There are problems with Indonesian MFC/R2. My software implements one
version of Indonesian R2 signaling, but there are others. Some are very
different from normal MFC/R2 signaling, and would require a considerable
rewrite of he existing software. You need to find out a little more
about
On 8/20/06, Roy Sigurd Karlsbakk <[EMAIL PROTECTED]> wrote:
hi all
I have a few boxes in my setup, most connected to one or more PRIs.
Today, we're making the telco do the number filtering in the switch,
this number range to this PRI etc. This is all fine, but not too
flexible, as we're depended
Hi
Is there a way to connect an Cellphone to asterisk in order to route calls
though it?.
This is what I want to do:
Here is much cheaper to call from cell to cell than from fixed line to cell.
So I want to connect a cell to the asterisk box and create a rule to route
calls to a cell through the
Hi there,
I was using the old
metermaid patch with Asterisk 1.2.9.1. I just updated to the current trunk and
the hint for my parking slot doesn’t work anymore. I’m always
getting State:Unavailable. Is there anything I need to change of this features
is broken ?
[EMAIL PR
So, a few questions:
- If the call received by asterisk from the PRI is sent to a number
not in the dialplan, what will asterisk do? Will the call be
cancelled, or will asterisk signal something back to the switch to
indicate "dunno about this, try another"?
Asterisk will do whatever you tell
junghanns.net has some neat gsm boards that can do this
On Aug 20, 2006, at 6:38 PM, Alvaro Cornejo wrote:
Hi
Is there a way to connect an Cellphone to asterisk in order to
route calls
though it?.
This is what I want to do:
Here is much cheaper to call from cell to cell than from fixed li
On 8/20/06, Roy Sigurd Karlsbakk <[EMAIL PROTECTED]> wrote:
>> So, a few questions:
>>
>> - If the call received by asterisk from the PRI is sent to a number
>> not in the dialplan, what will asterisk do? Will the call be
>> cancelled, or will asterisk signal something back to the switch to
>>
I have succeeded in outbound dialling through my cell (AT&T Sony Ericsson T616 and an old bluetooth adapter with CSR chipset connected to *) using instructions given here
http://www.thetechguide.com/howto/asterisk/chanbluetooth.html
I am still trying to get inbound route (outside calls reaching
Finally, in the trunk all
the states of my device are broken. If I downgrade to 1.2.10, everything is
fine. The device get busy and ringing. But in the current trunk Asterisk
SVN-trunk-r40632M none of my hints works.
Anyone could confim this
bugs ?
David
De :
[EMAIL PROT
On 8/17/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
Hi folks,Just a small question, i always use a X101 Card where in my office withasterisk, works great etc.
Now I receive a T1/E1 30 Lines and buy a Digium Card for it.Very well, how my extensions change ??Did i still use ZAP/01, ZAP/02 etc f
Hi,
Im experiencing no audio problem on my SIP user. The callee's
reception is alright but there is no audio on my SIP user. codec I use
is ulaw.
Note that its a realtime sip users and incoming (extension) is also realtime.
John
___
--Bandwi
Let me start by saying when I first plugged it in, I didn't have the
files set up on my ftp server yet, and the phone used it's default
settings and it completed bootup. Now...
I started with sip v1.6.6b and bootrom 3.1.3 on the ftp server. Phone
boots, d/l's files, reaches "Welcome" screen and
Keyboardot ragadtam, hogy va'laszoljak Tzafrir Cohen osszedobalt bytejaira:"
>
> > btw: i prefer HP servers (above 3xx) because you can do health monitoring
> > really nice (fans, temp, ps status, etc)
>
> configure lm_sensors on just about any system built in the recent years
> and you'll get th
You can use FXO card to connect gsm adapter with analogue line,
(170Eur for used one, and about 350 for new)
also you can use bri card, with isdn gsm adapters
(about 800Eur for a 2channel)
and you can go for junghanns and voismart for a pci card
with asterisk support (with sms)
> junghanns.net ha
Alvaro,
you probably need a GSM Gateway. We've been using the one from Topex
(http://www.topex.ro) and it works quite well (at least here in Italy).
They have models with a single GSM card on board and with two cards.
The gateway gets tied either to a Zap (single SIM) or ISDN (double SIM)
channe
I´m using chan_bluetooth to use a bluetooth enabled cell phone to make call
to another cells. The thing is that it is a cheap solution but not
professional. I mean, it can fail.
Good luck,
Danko
- Original Message -
From: "Alvaro Cornejo" <[EMAIL
How about a Cell Socket? Just plug it into your FXO card and you're
set.
http://www.ctdi.com/cellsockets.htm
- Original Message -
From: "Alvaro Cornejo" <[EMAIL PROTECTED]>
To:
Sent: Sunday, August 20, 2006 1:38 PM
Subject: [asterisk-users] Connecting an cellphone to asterisk
Hi
Is
Be aware that Cellsocket is a dead end.
WHP wireless is gone
When these are gone, they are gone
They do work, however.
GSM versions do not forward CLID
GSM requires a "#" to end the dial string
They do work however
John Novack
Don Fanning wrote:
How about a Cell Socket? Just plug it into your
On Sun, Aug 20, 2006 at 04:06:06PM -0400, John Novack wrote:
> Be aware that Cellsocket is a dead end.
> WHP wireless is gone
> When these are gone, they are gone
> They do work, however.
> GSM versions do not forward CLID
> GSM requires a "#" to end the dial string
> They do work however
Anythin
Over the last few months, I've peeked at the web site a few times and there was
_two_ jobs (the same ones every time). Now, there's _zero_. Is that likely to
increase any time soon?
-Original Message-
From: Matt Gibson [mailto:[EMAIL PROTECTED]
Sent: Sun 8/20/
I've seen analog-to-VoIP gateways such as the Audiocodes one -- which,
truthfully, looks very, very nice -- but I've got several hundreds of
analog phones to deal with, and I was wondering if anyone has seen
something with even higher concentrations than the Audiocodes
24-ports-per-rack-unit.
Than
Ken - Quintum's Tenor AX Series analog gateways offer densities up to 48
FXO.
Here's a link http://www.quintum.com/enterprise/en_productdetail.html?id=21
Cory J Andrews
VOIPSupply.com
454 Sonwil Drive
Buffalo, NY 14225
++
voice - 800.398.VoIP X3402
email - [EMAIL PROTEC
Hello, is there any way to send signals to asterisk, for example, I send a
sign to a parallel port and it calls an extension. I can´t modify asterisk
code to make it. Any ideas?
Thanks for your time,
Danko
___
On Wed, Aug 16, 2006 at 08:38:49PM -0400, Robert La Ferla wrote:
> I found an init.d script for asterisk BUT not for asterisk/zaptel
> modules. I'm still looking for a good solution. It seems to me that
> the correct solution would involve /etc/modprobe.d/modpobe.conf.
Have you noticed the s
The best way would be that you have a program read whatever you get
from the parallel port and that program pass it on to asterisk thru
the shell (i.e. asterisk -rx "command") , or you could use the manager
API, or you could use call files.
On 8/20/06, Danko Miocevic <[EMAIL PROTECTED]> wrote:
H
Greetings..
I have a few Linksys SPA-941 IP phones running the latest firmware 4.1.12(a).
I tried turning on the Message Waiting indicator but it doesn't seem
to work correctly for me. This phone is connecting to Asterisk 1.24
running Realtime.
Not sure if it matters but rtcachefriends=yes is s
I'm not sure there's much point in developing it in Erlang anyways. I'll
usually do a quick look and see how popular a language or technology is, in the
job market before I spend time and effort on learning it. A search on dice for
Erlang gets about 3 results.
Doug.
-Original Mess
On Sun, 2006-08-20 at 13:17 +0200, Woodoo People .pGa! wrote:
> > > I am curious as to what hardware folks are using successfully from HP
> > > or DELL. I will likely be running just a quad span T1 card with the
> > > system.
> >
> > HP DL380 G4, 4GB mem, 2x 146GB U320 in RAID1, dual hotswap PS
>
On Thu, Aug 17, 2006 at 08:41:18PM -0500, Eric ManxPower Wieling wrote:
> Untested:
>
> exten => _,1,System(/bin/mail -s \"Happy Message: ${EXTEN}\"
> [EMAIL PROTECTED])
Hmmm... is the standard input on System() set to /dev/null?
--
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755
On Sun, Aug 20, 2006 at 04:18:11PM -0600, Douglas Garstang wrote:
> I'm not sure there's much point in developing it in Erlang anyways.
> I'll usually do a quick look and see how popular a language or
> technology is, in the job market before I spend time and effort on
> learning it. A search on
[Still Off-topic]
On Sun, Aug 20, 2006 at 09:03:42PM +0200, Woodoo People .pGa! wrote:
> Keyboardot ragadtam, hogy va'laszoljak Tzafrir Cohen osszedobalt bytejaira:"
> >
> > > btw: i prefer HP servers (above 3xx) because you can do health monitoring
> > > really nice (fans, temp, ps status, etc)
Hey guys,
Last week I changed my queues from using proper agents and
AgentCallbackLogin() to using the the FreePBX default with fixed agents
(which uses the Local/[EMAIL PROTECTED] style for the member= field).
I've also upgraded to Asterisk 1.2.10 and FreePBX 2.2.0 Beta 1.
Since then, I not
thanks a lot David, its really useful
after googling I found one more link on ramfs http://www.linuxfocus.org/English/July2001/article210.shtml
thought this can be useful for others.
Nitin
On 8/18/06, David Gagnon <[EMAIL PROTECTED]> wrote:
Hi, Take a look at ramfs (http://plume.bxlug.be
On Fri, Aug 18, 2006 at 07:08:13PM -0700, Nitin Gupta wrote:
> Hi,
> Is there any option in asterisk to load all the prompt files into
> memory on startup, so that it doesn;t have to hit the disk to read
> prompts for any call.
> Or any plugin / suggestion to avoid hitting the disk for prompt files
voiplist wrote:
Greetings..
I have a few Linksys SPA-941 IP phones running the latest firmware
4.1.12(a).
I tried turning on the Message Waiting indicator but it doesn't seem
to work correctly for me. This phone is connecting to Asterisk 1.24
running Realtime.
Not sure if it matters but rtca
Hello everyone,
I'm having some problems with Asterisk not parking calls. The server
is setup with [EMAIL PROTECTED] and FreePBX.
If there's an incoming call and we want to park the call (by pressing
#700), sometimes the call is parked, sometimes it's not, sometimes the
notification only pops up
I would like to transfer an incoming call and, when the call is answered,
have the caller id of the call spoken when the call is answered on my cell
phone.
Any tips greatly appreciated
AAH 2.7 using sip trunks exclusively.
Regards
___
--Bandwidth a
I was unclear regarding the CLID
I meant CLID on an INCOMING call.
Cellsocket passes "cellsocket", not what the phone received.
John Novack
Steve Kennedy wrote:
On Sun, Aug 20, 2006 at 04:06:06PM -0400, John Novack wrote:
Be aware that Cellsocket is a dead end.
WHP wireless is gone
When t
On 18/08/2006, at 10:43 PM, Noah Miller wrote:
Hi Nathan -
The problem occurs during transfer and hold retrieval, answering the
call is fine, the call is put on hold then either a transfer is
attempted or the call is retrieved from hold. When this is attempted
the remote party (i.e. the caller
Hello,
I need to send some information to our German HQ regarding my experiences with
VoIP. Asterisk is very prominent in those experiences. I would like to
include information about installations of Asterisk at German
companies/universities. Any info is appreciated.
Vielen dank fuer die
Is it a bug or is it me?
For the longest time I have been using the feature within voicemail to
call back a number by caller ID.
Never had a problem with it at all.
I just updated to the latest (stable) asterisk from asterisk.org
Option 3 (advanced) then 2 then 1
caller number 7347292615
and no
On 8/20/06, Steve Gladden <[EMAIL PROTECTED]> wrote:
Is it a bug or is it me?
The voicemail system in asterisk is very buggy. Can you show us the
text config file thingy that's associated with the voicemail message?
___
--Bandwidth and Colocation pro
Did you put:
include => parkedcalls
in your dialplan?
___
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In this sections there are a context [conferences] Where Can I put this lines? in extension.conf?2006/8/19, RR <[EMAIL PROTECTED]
>:Follow the instructions here:
http://www.voip-info.org/wiki/view/Asterisk+app_conferenceThere's no config file where conferences are stored. You need to addthem to ast
Yes
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Hi Rushowr,Thank you for your response. As you said, I executed these below lines:exten => s,n,Verbose(2|CallerID info received: ${CALLERID(all)}) ; shows CID info exten => s,n,Verbose(2|Presentation Setting: ${CALLINGPRES}) ; shows CID presentationAnd Asterisk is showing this below error on con
sox needs for gsm an optional library.
I was not able to locate this one. Can anybody point me to this place?
bye
Ronald
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In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] says...
> For more information or to start looking for Open source Asterisk VOIP
> employment
> head over to http://www.asterisk-jobs.com
Is it that I don't know how to make search or there is no jobs available in any
country?
--
Tomislav Parči
Dear Leo,As you said, I have tried using dtmf and in different values. But, no reuslt. Finally, I knew that Basic Asterisk setup doesn't recognize callerid in India. To get callerid in India, we have to do some modifications in chan_zap.c source file. But, I dont know what modifications I have to d
On Mon, Aug 21, 2006 at 02:02:31PM +0800, Ronald Wiplinger wrote:
> sox needs for gsm an optional library.
>
> I was not able to locate this one. Can anybody point me to this place?
As there is a Debian package you can grab the orig tarball from:
http://packages.debian.org/unstable/libs/libgsm1
Hi im experiencing no audio problems. ive installed the latest
asterisk 1.2.10 zaptel, libpri & asterisk.
the caller's side reception is fine but i hear nothing on my sip account.
Please help
Regards,
John
___
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