Hi,
I made good experienes with Siemens Gigaset C610 IP. This model is about
90 Euro. Configuration via web interface. But encryption (SIPS/SRTP) is
*not* possible with this phones.
-Thorsten-
Am 11.12.2013 11:30, schrieb Mario Giammarco:
Hello,
I need to setup this configuration:
-
Hello,
I need to setup this configuration:
- asterisk as IVR;
- dect phones.
So basically I need a standard set of features:
- each dect phone has its extension so I can call it directly;
- handover of a call with R key;
- if a call is not replied by someone ring all phones.
I have little
Hello Mario,
nice to meet you on this mailing list!
Gigaset phones are a very high quality/price ratio, so I'll suggest you to
go with the dect ip models. Then you'll need to configure asterisk to act
as IVR, configure a queue and a failover to ring all hunt list.
Drop me a phone call and I'll be
...@lists.digium.com] On Behalf Of Dean Collins
Sent: Thursday, February 17, 2011 7:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]Newbie´s question about Asterisk...
If you already have experience with linux asterisk will be easy for you.
Other people
List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users]Newbie´s question about Asterisk...
If you already have experience with linux asterisk will be easy for you.
Other people will reply with official links but here is how I use Asterisk
in my small home office www.cognation.net
Discussion
Subject: Re: [asterisk-users]Newbie´s question about Asterisk...
i prefer to go with Elastix very easy to setup and maintain and reach UI rather
than freePBX
cheers
Dhaval
On Fri, Feb 18, 2011 at 4:14 PM, Terry Brummell te...@brummell.net wrote:
Dean's link has references to Trixbox. TB
-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *DHAVAL INDRODIYA
*Sent:* Friday, February 18, 2011 6:11 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users]Newbie´s question about Asterisk...
i prefer
-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *DHAVAL INDRODIYA
*Sent:* Friday, February 18, 2011 6:11 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users]Newbie´s question about Asterisk...
i prefer
-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *DHAVAL INDRODIYA
*Sent:* Friday, February 18, 2011 6:11 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users]Newbie´s question about Asterisk...
i prefer
...@lists.digium.com on behalf of Francisco Javier
Cintrón Olguín
Sent: Fri 2/18/2011 12:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]Newbie´s question about Asterisk...
Is there another way to interface to 3 external and 6 internal lines??
Thank you
(Please don't top-post and please trim posts that are no longer relevant.)
On Fri, 18 Feb 2011, Francisco Javier Cintrón Olguín wrote:
I think I have 3 PSTN lines because I can connect a normal telephone to
them all and make calls between each of them. We have 5 normal
telephones and 1
First of all, thank you for your help.
I was seing Cisco and Linsys web sites and I just came across this 2
devices:
Linksys SPA8000 8 phone ports, 1 port ethernet.
Cisco SPA8800 4 phone ports, 4 lines, 1 port ethernet.
I think they could work for us, because I need maximum 10 normal phones and
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Francisco Javier
Cintrón Olguín
Sent: Thursday, February 17, 2011 7:26 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Newbie´s question about Asterisk...
Hi, My name is Francisco from México.
Here
On Fri, 4 Feb 2011 10:54:56 +0330, Pezhman Lali l...@lopl.net wrote:
Meetme is a default conference application, but you can try conference or
konference
http://www.voip-info.org/wiki/view/Asterisk+cmd+Conference
http://www.voip-info.org/wiki/view/Asterisk+cmd+Conference
Hello
I've never used Asterisk for a three-person call, and would like to
check that MeetMe is the way to do this.
The ADSL modem provided by my ISP offers free calls to
landlines/cellphones when using a handset connected to an RJ11 port on
the modem.
A three-person call can be set up
Dear,
Meetme is a default conference application, but you can try conference or
konference
http://www.voip-info.org/wiki/view/Asterisk+cmd+Conference
http://www.voip-info.org/wiki/view/Asterisk+cmd+Conference
http://www.voip-info.org/wiki/view/Asterisk+cmd+Konference
Hello!
Im new to Asterisk configuration and I have few questions regarding its
configuration.
I have 4 PSTN lines connected to TDM410. I can make exact 60 minutes of free
calls from each of 4 pstn lines... Can I configure Asterisk to call thru
pstn line that has free minutes? For example
On Mon, 31 Jan 2011, Piotr Górski wrote:
I have 4 PSTN lines connected to TDM410. I can make exact 60 minutes of
free calls from each of 4 pstn lines... Can I configure Asterisk to call
thru pstn line that has free minutes? For example
Outgoing calls are going through PSTN 1 for 60 minutes.
Hi,
I've recently installed Asterisk 1.6.2.13. I'd like to connect GSM Trunk to
it. I purchased a few Mobigater ProOpen gateways. It states that I should
use chan_celliax module to it. On the gsmopen site I see a comment in the
documentation that I can install the module on Asterisk 1.2.x,
Hi,
Is doing this as simple as just creating a queue in queue.conf?
I have the following setup:
1. Single hunting 1-800 number mapped to multiple numbers in a hunt
group by the telco.
2. All calls land up at Asterisk over SIP.
Now I need:
3. A hunt group of operator extensions mapped to the
Hello,
can the Asterisk API be used to automate a MITEL 5330 telephone?
If not, are there any other API which can used to do that?
Many thanks.
phiroc
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing
Sorry for such a silly question but I am VERY new to Linux, Asterisk,
and so forth. I just downloaded and burned the AsteriskNOW ISO to CD
and installed it. Everything went great. I removed the CD and
rebooted and there is a prompt for me to login.
I hate to ask but after searching for a few
UIT DEVELOPMENT wrote:
Sorry for such a silly question but I am VERY new to Linux, Asterisk,
and so forth. I just downloaded and burned the AsteriskNOW ISO to CD
and installed it. Everything went great. I removed the CD and
rebooted and there is a prompt for me to login.
I hate to ask
John, Thank you! That was it. I was trying admin, login...I
should have searched google more. Thank you and thanks for the link -
I've got lots of reading ahead of me this evening!
Mike
On Fri, Dec 25, 2009 at 6:00 PM, John Novack
jnov...@stromberg-carlson.org wrote:
UIT DEVELOPMENT
Hi,
I just started with Asterisk as I am very unhappy with the functionality
of my current PBX at home. I try to understand everything and play
around, but it is not as easy as I thought. So please be patient if this
is a too easy question for You.
I installed Asterisk 1.4.26.3 on a Debian Lenny
Hi,
I'd say Linphone configuration. I suggest to check Linphone
configuration and also asterisk debug - one of those will give you an
answer (most likely asterisk debug as it shows you what it receives ...
if it doesn't receive anything then Linxphone fails)
--
razu
On 11/19/2009 11:48 PM,
Hi All,
When typing 'help' on the command line (* console) is there a way to
keep it from just scrolling most of the information off the top of the
screen? I can't hit ctrl-s fast enough so I miss most of the info. This
makes 'help' be not much help.
Thanks,
Bill
: Tuesday, November 17, 2009 10:34 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] newbie question
Hi All,
When typing 'help' on the command line (* console) is there a way to
keep it from just scrolling most of the information off the top of the
screen? I can't hit ctrl-s fast
On Tue, Nov 17, 2009 at 11:33:45AM -0500, Bill Shaw wrote:
Hi All,
When typing 'help' on the command line (* console) is there a way to
keep it from just scrolling most of the information off the top of the
screen? I can't hit ctrl-s fast enough so I miss most of the info. This
makes
On Tue, Nov 17, 2009 at 11:33:45AM -0500, Bill Shaw wrote:
Hi All,
When typing 'help' on the command line (* console) is there a way to
keep it from just scrolling most of the information off the top of the
screen? I can't hit ctrl-s fast enough so I miss most of the info. This
makes
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] newbie question
On Tue, Nov 17, 2009 at 11:33:45AM -0500, Bill Shaw wrote:
Hi All,
When typing 'help' on the command line (* console) is there a way to
keep it from just scrolling most of the information off the top of the
screen
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Bill Shaw
Sent: Tuesday, November 17, 2009 11:34 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] newbie question
When typing 'help
On Tue, Nov 17, 2009 at 09:09:39AM -0800, Steve Edwards wrote:
On Tue, Nov 17, 2009 at 11:33:45AM -0500, Bill Shaw wrote:
Hi All,
[snip]
2. Run from the external shell prompt:
asterisk -rx 'help whatever' | less
Or, you can use the script command to capture the output to a file
When typing 'help' on the command line (* console) is there a way to
keep it from just scrolling most of the information off the top of the
screen? I can't hit ctrl-s fast enough so I miss most of the info. This
makes 'help' be not much help.
my default scroll back buffer is set to around
On Tue, 17 Nov 2009, Noah Miller wrote:
You could also make it much simpler and just set your verbosity very
low or just turn it off, so there are very few messages coming across
your screen. Unless you're on a really busy machine, you should be
able to read most of the help screens.
core
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 was copying tracks from CD into mp3 files so that I could use it in
Asterisk 1.4.21.2 MOH. (BTW, I have already secured proper license to
play MOH to callers.)
I used MS Media Player version 11 and rip it at 128kbps (smallest) but
whenever I listen to MOH, I saw the following message on the
Hi All,
This is my first post. I searched the archives and found something similar
and I tried some of those suggestions. I changed the file permissions on the
scripts directory to 777 (which doesn't seem secure), I also manually ran
the detectdahdi.sh script. The response is None.
I am running
On Wed, 19 Aug 2009, Lee, John (Sydney) wrote:
I was copying tracks from CD into mp3 files so that I could use it in
Asterisk 1.4.21.2 MOH.
Are there any Asterisk+Audio expert that can offer me some advice?
Don't use MP3. Why would you want to burn CPU cycles decompressing the
same stuff
Steve Edwards wrote:
On Wed, 19 Aug 2009, Lee, John (Sydney) wrote:
I was copying tracks from CD into mp3 files so that I could use it in
Asterisk 1.4.21.2 MOH.
Are there any Asterisk+Audio expert that can offer me some advice?
Don't use MP3. Why would you want to burn CPU cycles
Yep, agreed.
Convert the file to the native codec(s) in which it will be played.
Alex, could you please elaborate on this? I am no audio guy.
On Media player, I can rip it into mp3 or wav or windows media audio.
Which one should I use?
___
--
Probably none of the ones you list, though I believe wav files are
uncompressed. Use SOX http://sox.sourceforge.net/ under Linux, Windows or
OSX and RIP/Convert the files to match the codec you are using for calls.
If you are accepting calls that use the GSM codec then have a set of MOH
files
On Thu, 20 Aug 2009, Lee, John (Sydney) wrote:
Convert the file to the native codec(s) in which it will be played.
Alex, could you please elaborate on this? I am no audio guy.
On Media player, I can rip it into mp3 or wav or windows media audio.
Which one should I use?
Neither.
If your
Lee, John (Sydney) wrote:
Thanks Tilghman.
I learnt it the hard way - I never imagined I need to jot down the
serial number of a PCI card :-(
I've had a linecard that's been unregistered now for 4 years or more,
because it's in a production server.
It does of course mean that I didn't get
Sent: Monday, 17 August 2009 1:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Newbie: How to find the serial number
ofDigium card?
On Sunday 16 August 2009 19:59:50 Lee, John (Sydney) wrote:
Does anyone know how to find the serial number of Digium
Does anyone know how to find the serial number of Digium card without
opening the machine?
I was trying to call for support at Digium and they asked me for the
serial number.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
On Sunday 16 August 2009 19:59:50 Lee, John (Sydney) wrote:
Does anyone know how to find the serial number of Digium card without
opening the machine?
I was trying to call for support at Digium and they asked me for the
serial number.
You cannot. The serial number is not anywhere in the
:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Newbie: How to find the serial number
ofDigium card?
On Sunday 16 August 2009 19:59:50 Lee, John (Sydney) wrote:
Does anyone know how to find the serial number of Digium card without
opening
On Sat, 20 Jun 2009, C. Savinovich wrote:
Let me see if I get you: you inserted the installation CD, then you
restarted the computer, and now you want to know what to do next?
How about:
1) Turn off the computer.
2) Read the installation guide for the CD.
3) Install the software.
4) Read
I have an Asterisknow.org CD. When I boot up, it seems ready for me to
choose update, console, etc. I'm assuming I need to do something at the
CLI prompt. Is there a tutorial that would take me from loading CD to
making first test call?
Computer is Dell Optiplex GX260
50GB free disk space
-users] Newbie, Question on making a PSTN call..
Need help pls..Anyone?
On Mon, Jun 15, 2009 at 10:58 PM, Shiva Kumar shi...@gmail.com wrote:
Hello Asterisk-users,
I am new to Asterisk. I got SIP Calls to work between two computers using a
soft phone and asterisk in the middle. Since then, I
On Mon, 15 Jun 2009, Shiva Kumar wrote:
Hello Asterisk-users,
I am new to Asterisk. I got SIP Calls to work between two computers using a
soft phone and asterisk in the middle. Since then, I have been trying to get
my soft phone to make a PSTN call with terrible failure for about two days
now.
Hello Asterisk-users,
I am new to Asterisk. I got SIP Calls to work between two computers using a
soft phone and asterisk in the middle. Since then, I have been trying to get
my soft phone to make a PSTN call with terrible failure for about two days
now.
On Windows using asteriskwin32:
I have a
Need help pls..Anyone?
On Mon, Jun 15, 2009 at 10:58 PM, Shiva Kumar shi...@gmail.com wrote:
Hello Asterisk-users,
I am new to Asterisk. I got SIP Calls to work between two computers using a
soft phone and asterisk in the middle. Since then, I have been trying to get
my soft phone to make a
Hello,
This is my first asterisk installation, and having read up on the
documentation, and trying several tutorials i'm unable to get my
outbound route working. I'm certain it's an issue with my configuration
and my inexperience with asterisk. So i have my POTS phone connected to
my digium
On Mon, Mar 30, 2009 at 5:16 PM, Bruce Thayre br...@mipscomputation.comwrote:
Up to this point, all i have set up are two SIP phones, my POTS phone,
and 1 ring group. My POTS line is connected to channel 1, and my POTS
phone is connected on channel 3. Perhaps my understanding of how the
Show us your dialplan.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce Thayre
Sent: Monday, March 30, 2009 4:16 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Newbie trying to make calls
...@lists.digium.com] On Behalf Of Bruce Thayre
Sent: Monday, March 30, 2009 4:16 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Newbie trying to make calls outside via digiumcard
and POTS line
Hello,
This is my first asterisk installation, and having read up on the
documentation
Geoff Lane wrote:
On Thursday, February 5, 2009, Mark Michelson wrote:
I've tried it and you're correct. So it looks like the docs need a
bug report - any idea how I go about that?
If you're using the 2nd edition of the book, check the preface, page xix for
contact information.
Thanks
On Thu, 2009-02-05 at 22:09 +, Geoff Lane wrote:
Thanks. For info, *TFOT says:
PrivacyManager() sets a channel variable named PRIVACYMGRSTATUS to
either SUCCESS or FAILURE. If Caller ID is received on the channel,
PrivacyManager() does nothing.
I've tried it and you're correct. So it
On Thu, 2009-02-05 at 22:01 +0100, Philipp Kempgen wrote:
Mark Michelson schrieb:
Actually, jumping to priority n + 101 is a thing of the past
And in addition extensions.conf is a thing of the past. ;-)
extensions.ael is cleaner and easier to maintain for most
purposes.
In the same
On Thu, 2009-02-05 at 22:01 +0100, Philipp Kempgen wrote:
Mark Michelson schrieb:
Actually, jumping to priority n + 101 is a thing of the past
And in addition extensions.conf is a thing of the past. ;-)
snip
How about .. dialplan.conf .;-)
Hi All,
Asterisk 1.4.12 on CentOS 5
Sorry for a question that I'm guessing is obvious to most of you.
I'm trying to revamp my dialplan. When I first created it, I had
something like:
exten = s,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
exten = s,2,Dial(${rgMain},${RINGTIME},t)
exten
Geoff Lane wrote:
Hi All,
Asterisk 1.4.12 on CentOS 5
Sorry for a question that I'm guessing is obvious to most of you.
I'm trying to revamp my dialplan. When I first created it, I had
something like:
exten = s,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
exten =
On Thursday, February 5, 2009, Mark Michelson wrote:
Actually, jumping to priority n + 101 is a thing of the past, and
this will only occur now if you pass the 'j' option to Dial. Dial
will just go to the next priority on a timeout now, and the
DIALSTATUS channel variable will be set to
Mark Michelson schrieb:
Actually, jumping to priority n + 101 is a thing of the past
And in addition extensions.conf is a thing of the past. ;-)
extensions.ael is cleaner and easier to maintain for most
purposes.
Philipp Kempgen
--
AMOOCON 2009, May 4-5, Rostock / Germany -
On Thursday, February 5, 2009, Philipp Kempgen wrote:
And in addition extensions.conf is a thing of the past. ;-)
extensions.ael is cleaner and easier to maintain for most purposes.
Oh-oh ... I don't think I can keep up with the rate of change ;-)
BTW, on a related note, I'm having some
Philipp Kempgen wrote:
And in addition extensions.conf is a thing of the past. ;-)
extensions.ael is cleaner and easier to maintain for most
purposes.
*gack*
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither
On Thursday 05 February 2009 15:37:19 Geoff Lane wrote:
On Thursday, February 5, 2009, Philipp Kempgen wrote:
And in addition extensions.conf is a thing of the past. ;-)
extensions.ael is cleaner and easier to maintain for most purposes.
Oh-oh ... I don't think I can keep up with the rate
On Thursday, February 5, 2009, Tilghman Lesher wrote:
The correct string is FAILED, not FAILURE.
Thanks. For info, *TFOT says:
PrivacyManager() sets a channel variable named PRIVACYMGRSTATUS to
either SUCCESS or FAILURE. If Caller ID is received on the channel,
PrivacyManager() does nothing.
Tilghman Lesher schrieb:
On Thursday 05 February 2009 15:37:19 Geoff Lane wrote:
BTW, on a related note, I'm having some trouble with Privacy Manager
that I'd appreciate some insight with. In one priority, I'm calling
PrivacyManager(2,8). In the next priority, I've got:
...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Philipp Kempgen
Sent: Thursday, February 05, 2009 16:01
To: Asterisk Users
Subject: Re: [asterisk-users] Newbie query: how to write priority n+101
Mark Michelson schrieb:
Actually, jumping to priority n + 101
Geoff Lane wrote:
On Thursday, February 5, 2009, Tilghman Lesher wrote:
The correct string is FAILED, not FAILURE.
Thanks. For info, *TFOT says:
PrivacyManager() sets a channel variable named PRIVACYMGRSTATUS to
either SUCCESS or FAILURE. If Caller ID is received on the channel,
On Thursday, February 5, 2009, Mark Michelson wrote:
I've tried it and you're correct. So it looks like the docs need a
bug report - any idea how I go about that?
Thanks again,
If you're using the 2nd edition of the book, check the preface, page xix for
contact information.
Thanks -
'
Subject: Re: [asterisk-users] Newbie in Cisco Phone
Hi
I am no expert in the cisco phone
Do you have time to help
Sam
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Federico
Santulli
Sent: Saturday, January 24
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michiel van
Baak
Sent: Friday, January 23, 2009 3:35 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Newbie in Cisco Phone
On 05:39, Fri 23 Jan 09, Sam Tam wrote:
Yes I know too.
Is there anyway
- Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Friday, January 23, 2009 8:56 AM
Subject: Re: [asterisk-users] Newbie in Cisco Phone
Well does it matter if the asterisk server is not located in the same
network?
I am willing to spend a bit of cash to get someone help me
- Non-Commercial Discussion
Cc: tam...@gmail.com
Subject: Re: [asterisk-users] Newbie in Cisco Phone
you can try chan_sccp at www.chan-sccp.org
it supports most of ccm features and all kind of cisco phones with skinny
firmware.
Take a look ;)
If you need support you can write me back.
Federico
Hello all
I have used some low end cisco phones in the past and had no problem setting
up SIP on it.
But today, I have made a big mistake. Buying Cisco Conference phone without
even looking whether it supports SIP on not.
And yes it is the nice 7937G that I am talking about.
Damn this is
-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sam Tam
Sent: Thursday, January 22, 2009 3:54 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Newbie in Cisco Phone
Hello all
I have used some low end cisco
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Newbie in Cisco Phone
The 7936G/ 7937G Data Sheet says SCCP only which is a shame. It really is a
great sounding phone. I have several customers with them as SCCP.
http://www.cisco.com/en/US/prod/collateral/voicesw
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Aarons
(US)
Sent: Friday, January 23, 2009 5:10 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Newbie
List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Newbie in Cisco Phone
Asterisk's Skinny support is very rudimentary and doesn't include the
CCM provisioning stuff.
Short answer - not really. Not unless you want to go through a *whole*
lot of work.
Sam Tam wrote:
Yes I know too
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sam Tam
Sent: Thursday, January 22, 2009 3:54 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Newbie in Cisco Phone
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michiel van
Baak
Sent: Friday, January 23, 2009 3:35 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Newbie in Cisco Phone
On 05:39, Fri 23 Jan 09, Sam Tam wrote:
Yes I know too.
Is there anyway to make it work with asterisk
Lee, John (Sydney) wrote:
Calling all Polycom gurus:
I am using Polycom IP601 phones with Asterisk 1.4.21.2
In all Polycom phones, I set the following in sip.cfg.
dialplan dialplan.impossibleMatchHandling=2
/dialplan
(I leave the digitmap unchanged because I thought setting
Calling all Polycom gurus:
I am using Polycom IP601 phones with Asterisk 1.4.21.2
In all Polycom phones, I set the following in sip.cfg.
dialplan dialplan.impossibleMatchHandling=2
/dialplan
(I leave the digitmap unchanged because I thought setting
impossibleMatchHandling will ignore the
on the console.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Steve Murphy
Sent: Thursday, 11 September 2008 2:13 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Newbie AEL2: Syntax for Hint
On Thu, 2008-09-11 at 17:41 +1000, Lee, John (Sydney) wrote:
Steve, I downloaded the latest Asterisk version (see below).
*CLI core show version
Asterisk 1.4.21.2 built by root @ machine1 on a i686 running Linux on
2008-09-11 06:10:06 UTC
If I code:
Hint(Custom:light1)
It will pass
context BLF {
hint(Sip/1000) 1000 = NoOp();
};
Works for me
Thanks Eric.
I did not experience any problem in hint with SIP. The problem is if you use
it with Custom.
winmail.dat___
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Lee, John (Sydney) wrote:
I am struggling to find out how to code hint in AEL2.
I did hint(Custom:light1) and it keeps complaining about the : (colon).
It works fine for SIP device like hint(SIP/439).
Anyone who has tried it before?
I just whipped this up to test and it works for me in
I am struggling to find out how to code hint in AEL2.
I did hint(Custom:light1) and it keeps complaining about the : (colon).
It works fine for SIP device like hint(SIP/439).
Anyone who has tried it before?
___
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On Wed, 2008-09-10 at 18:10 +1000, Lee, John (Sydney) wrote:
I am struggling to find out how to code hint in AEL2.
I did hint(Custom:light1) and it keeps complaining about the : (colon).
It works fine for SIP device like hint(SIP/439).
Anyone who has tried it before?
Yes, a while back I
: Thursday, 11 September 2008 2:13 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Newbie AEL2: Syntax for Hint
On Wed, 2008-09-10 at 18:10 +1000, Lee, John (Sydney) wrote:
I am struggling to find out how to code hint in AEL2.
I did hint(Custom:light1
Hello!
Sorry, I'm sure it's stupid. but I've got a simple ISDN line and a simple
ISDN-card, now finally running. :-)
I'm using application Jack and asterisk (CLI) only to do my bidding. Now I
can make calls. But how ca I setup my extensions.conf to receive a call? I've
had an example like
I have been coding my own IVR for ACD (aka queue) using Polycom phones
using AEL2. In particular, I have coded my own AgentCallbackLogin
because a) cmd AgentCallbackLogin() is buggy and will not be supported
by dev anymore b) I can put in features like hotdesking and additional
validation like
I played with the Polycom login/logout function about a year ago, and it
looked brilliant.
I could never get it to work, but at the time I had both Polycom and
Digium agree that it would be worth getting running.
I ran out of time on that project, and have never re-visited it. But it
would
?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Faraz Khan
Sent: August 13, 2008 8:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Newbie: Queue and CDR Reporter and Analyser
queuemetrics
Lee, John
Doesn't Queuemetrics run on a license basis?
Anything else that's probably open source and free?
Does anyone have any comments/experience about using asteriskguru queue
statistics?
http://www.asteriskguru.com/tutorials/installation_guide.html
___
One of the Asterisk people down here in Melb set it up for the company
they used to work for, and I played with it once and it seemed to be usable.
PaulH
Lee, John (Sydney) wrote:
Doesn't Queuemetrics run on a license basis?
Anything else that's probably open source and free?
Does
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