15 jan 2010 kl. 08.23 skrev Yuji Kondo:
I have two questions for Asterisk feature.
1. Can Asterisk support presence feature ?
Asterisk is a telephony PBX and supports presence subscriptions for extension
states - if a phone line is busy or not, over a few different SIP presence
Ok this has Probably been put to bed several time but never mind.
Elastix, Trixbox, or AsterixNow, or DIY (ie Ubuntu or whatever
installed with OpenPBX, Asterix etc by hand)
I've got a new server to run Asterix on and want to get it working
quickly and yet be configurable in the future with out
Ishfaq Malik wrote:
Ishfaq Malik wrote:
Hi
We run a hosted VoIP service for multiple customers off the same server
and I'm having an odd issue with just one customer in particular. We're
using realtime in a MySQL DB and this is their dialplan
*** 1. row
hi, all
i try to confiture realtime queue, but not work, details as below:
Insert into queue_table(name)value('95040654321');
INSERT INTO queue_member_table VALUES ('', 'Zhang Shukun',
'95040654321', 'SIP/1001', 2, 1);
INSERT INTO queue_member_table VALUES ('', 'Li Aiwei', '95040654321',
Szasz Szabolcs wrote:
Hi,
How can I strip + from the front of the caller ID?
I have tried this:
exten = s/_+X.,1,Set(CALLERID(name)=${CALLERID(name):1})
Hi,
what about:
exten = _+[1-9].,1,Dial(Local/${EXTEN:1...@context
to proceed your call without leading + in your dialplan?
Cheers
- Paul Scott p...@cpanel.net wrote:
Yeah sounds like you wanna use NVFaxDetect
it would allow you to add something like exten = fax,1,Swift(number
has changed); to your inbound call part of your dialplan
On Jan 14, 2010, at 11:45 AM, Juan C. Villa wrote:
Could you use
On 01/15/10 17:54, randall wrote:
hi all,
i noticed that a lot of VOIP phones have a double network interface
allowing you to use only 1 LAN cable for both the phone and your
desktop, a really nice feature that can save a lot of cable, but most
are 10/100 connections while i have a
- --[ UxBoD ]-- ux...@splatnix.net wrote:
- Paul Scott p...@cpanel.net wrote:
Yeah sounds like you wanna use NVFaxDetect
it would allow you to add something like exten =
fax,1,Swift(number
has changed); to your inbound call part of your dialplan
On Jan 14,
Hi, I have a question about jitterbuffer and PLC.
I use Asterisk 1.6.2.0 and 1.6.0.20 or older.
I use uLaw.
My system map:
=
[ asterisk 2 ] -- # LOSS # -- # A # -- [ asterisk 1 ] -- # B # -- [ X-lite ]
On 01/15/2010 12:41 PM, Rob Hillis wrote:
On 01/15/10 17:54, randall wrote:
hi all,
i noticed that a lot of VOIP phones have a double network interface
allowing you to use only 1 LAN cable for both the phone and your
desktop, a really nice feature that can save a lot of cable, but most
Hi,
is someone able to provide inbound DID for South America, at least
Bolivia, Colombia, Panama and Venezuela.
Please contact me of list, thanks
Regards
--
Daniel
--
_
-- Bandwidth and Colocation Provided by
- Original Message -
From: randall
To: asterisk-users@lists.digium.com
Sent: Friday, January 15, 2010 7:54 AM
Subject: [asterisk-users] 10/100 voip phones and gigabit connection
hi all,
just subscribed to the list and first mail, nice to be here.
Hopefully i'm in
On 01/15/2010 02:00 PM, Leif Neland wrote:
- Original Message -
*From:* randall mailto:rand...@songshu.org
*To:* asterisk-users@lists.digium.com
mailto:asterisk-users@lists.digium.com
*Sent:* Friday, January 15, 2010 7:54 AM
*Subject:* [asterisk-users] 10/100
- Original Message -
From: Zhang Shukun
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Friday, January 15, 2010 11:48 AM
Subject: [asterisk-users] Realtime queue not work
hi, all
i try to confiture realtime queue, but not work, details as below:
- Original Message -
From: randall
To: asterisk-users@lists.digium.com
Sent: Friday, January 15, 2010 2:11 PM
Subject: Re: [asterisk-users] 10/100 voip phones and gigabit connection
On 01/15/2010 02:00 PM, Leif Neland wrote:
- Original Message -
2010/1/15 Leif Neland le...@neland.dk:
- Original Message -
From: Zhang Shukun
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Friday, January 15, 2010 11:48 AM
Subject: [asterisk-users] Realtime queue not work
hi, all
i try to confiture realtime queue, but not
comprobado este mensaje.
http://www.eset.com
__ Información de ESET NOD32 Antivirus, versión de la base de firmas de
virus 4774 (20100115) __
ESET NOD32 Antivirus ha comprobado este mensaje.
http://www.eset.com
__ Información de ESET NOD32 Antivirus, versión de la
Leif Neland wrote:
- Original Message -
*From:* Zhang Shukun mailto:bit...@gmail.com
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
mailto:asterisk-users@lists.digium.com
*Sent:* Friday, January 15, 2010 11:48 AM
*Subject:* [asterisk-users]
On 01/15/2010 02:19 PM, Leif Neland wrote:
- Original Message -
*From:* randall mailto:rand...@songshu.org
*To:* asterisk-users@lists.digium.com
mailto:asterisk-users@lists.digium.com
*Sent:* Friday, January 15, 2010 2:11 PM
*Subject:* Re: [asterisk-users] 10/100
On Fri, 15 Jan 2010, randall wrote:
Sure. My point was just that IF you only got one connection in the wall,
its cheaper to get a switch than getting a phone with dual 1Gbit ports.
Leif
OK, point taken.
but i have 6xisdn2 and already 2x24 gigabit switches (will need to replace
one
Zhang Shukun wrote:
2010/1/15 Leif Neland le...@neland.dk:
- Original Message -
From: Zhang Shukun
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Friday, January 15, 2010 11:48 AM
Subject: [asterisk-users] Realtime queue not work
hi, all
i try to confiture
2010/1/15 Robert Broyles bahj...@gmail.com:
Leif Neland wrote:
- Original Message -
*From:* Zhang Shukun mailto:bit...@gmail.com
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
mailto:asterisk-users@lists.digium.com
*Sent:* Friday, January 15, 2010
On 01/15/2010 02:54 PM, Jeff LaCoursiere wrote:
On Fri, 15 Jan 2010, randall wrote:
Sure. My point was just that IF you only got one connection in the wall,
its cheaper to get a switch than getting a phone with dual 1Gbit ports.
Leif
OK, point taken.
but i have 6xisdn2 and
2010/1/15 Robert Broyles bahj...@gmail.com:
Zhang Shukun wrote:
2010/1/15 Leif Neland le...@neland.dk:
- Original Message -
From: Zhang Shukun
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Friday, January 15, 2010 11:48 AM
Subject: [asterisk-users] Realtime
2010/1/12 Jeff LaCoursiere j...@jeff.net
That is so not true. FreePBX has hooks in a million places to do custom
dialplan stuff - I do it all the time. I also link in custom AGI/AMI
applications, custom provisioning, custom LCR, and am even working with
one customer that has mastered making
2010/1/15 Peter Childs pchi...@bcs.org
Elastix, Trixbox, or AsterixNow, or DIY (ie Ubuntu or whatever
installed with OpenPBX, Asterix etc by hand)
I've got a new server to run Asterix on and want to get it working
quickly and yet be configurable in the future with out having to
reisntall
HI All,
Can you please help me with the Dail plan for the following case:
I want to use Asterik as a B2B,Transport proxy.
I dont want to do regisration of UAC or UAS.
All I will do is the dail plan update with the routing information.
is it possible to do.
Asterik will be used between 2
I have a dialplan entry that takes a did, and sends it to a group of
stations Dial(Sip/ExtSip/ExtSip/Ext) etc.
However, cdr only shows dst = 5000 (given) and lastdata shows the dial
context, however I see no cdr entry for who actually answered the phone. , I
can see dstchannel as
Stephen Davies wrote:
Decide if you are going to be a zealot for your preferred approach
That's a little harsh, wouldn't you say? Do whatever your most
comfortable with. But, to call me and those like me a zealot, for
offering advice that was asked for is a little off, in my opinion.
Doug
The Asterisk Development Team has announced the release of Asterisk 1.4.29.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.4.29 resolved several issues reported by the
community, and would have not been possible
The Asterisk Development Team has announced the release of Asterisk 1.6.0.21.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.6.0.21 resolved several issues reported by the
community, and would have not been
The Asterisk Development Team has announced the release of Asterisk 1.6.1.13.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.6.1.13 resolved several issues reported by the
community, and would have not been
The Asterisk Development Team has announced the release of Asterisk 1.6.2.1.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.6.2.1 resolved several issues reported by the
community, and would have not been
2010/1/15 Doug Lytle supp...@drdos.info
Decide if you are going to be a zealot for your preferred approach
That's a little harsh, wouldn't you say? Do whatever your most
comfortable with. But, to call me and those like me a zealot, for
offering advice that was asked for is a little off,
On Fri, Jan 15, 2010 at 1:54 AM, randall rand...@songshu.org wrote:
does anybody know of another solution to this or is my conclusion above
simply all the choice there is?
So let me get this straight.
You're planning on buying multiple Gigabit, PoE switches, and you're
quibbling over the price
Have an intersting issue whem migrating a site from Zap on 1.3 to DAHDI on
1.4.. Nothing special about the hardware - older TDM400 card, 2 red
modules fitted...
Both channels work fine under 1.2/Zaptel. With 1.4/DAHDI both channels
still work OK, but only for one line - the 2nd line causes it
Most important thing is to PLAN your solution out.. flowcharts,
understanding where calls go, etc.
Project planning, and good ideas on how the calls should be handled, and
coming up with testing scenarios, to make sure everything flows correctly.
From:
Is there a way I can change the ring cadence on FXS lines on a system using a
Digium Wildcard TDM2400 card? I recently deployed a new phone system, and the
customer has a few POTS phones in areas where they don't have data network
services, so we're using the FXS lines to provide dialtone at
On 01/15/2010 05:01 PM, David Backeberg wrote:
On Fri, Jan 15, 2010 at 1:54 AM, randallrand...@songshu.org wrote:
does anybody know of another solution to this or is my conclusion above
simply all the choice there is?
So let me get this straight.
You're planning on buying
I don't know how many of you are going to be at ITEXPO/Digium Asterisk
World in Miami next week - I hope to see as many of you as possible,
though.
There has been an interesting change in the line-up for the show, that
I think bears mentioning here since it possibly will help quite a few
Hi all,
I've an asterisk ver 1.4.22.
As in object I have an extension beloning to a queue.
I need that for an external incoming call, the extension recieve the
call waiting signal/tone, whereas for internal incoming call, the
extension appear busy.
Is it possible? Could someone let me know the
Hi Guys,
Other than than yum repository (which fails when installing freepbx with it)
are there any automated install scripts out there that would install
Asterisk 1.6 or 1.4 onto a CentOS LAMP system?
If the script install FreePBX that would be a BONUS.
Thanks,
Bruce
--
I am having a problem with configuring TE410P card. I am using C2SBX+
motherboard from supermicro with fc11 os
After installing the dahdi drivers and running the command
/etc/init.d/dahdi start, the output of /proc/interrupts shows only 2
interrupts generated (I have enabled only 2 spans in
On Fri, Jan 15, 2010 at 12:27 PM, Bruce Nik brucev...@gmail.com wrote:
Hi Guys,
Other than than yum repository (which fails when installing freepbx with it)
are there any automated install scripts out there that would install
Asterisk 1.6 or 1.4 onto a CentOS LAMP system?
If the script
Provided there is no comprehensive install guides (or is there?) yes I would
like to see an easy install script which can install it all.
On Fri, Jan 15, 2010 at 12:27 PM, Bruce Nik brucev...@gmail.com wrote:
Hi Guys,
Other than than yum repository (which fails when installing freepbx with
On Fri, Jan 15, 2010 at 3:15 PM, Bruce Nik brucev...@gmail.com wrote:
Provided there is no comprehensive install guides (or is there?) yes I would
like to see an easy install script which can install it all.
tar xvzf
./configure
make
(optional, do a 'make menuconfig')
make install
But the
On Fri, 2010-01-15 at 14:11 +0100, randall wrote:
its not the network switch that i'm worried about, its the build in
switch of the phones with the double network card
--
Hi,
Don't think you'll find phone's with an internally gbit switch.
As for voip it is not needed.
If you connect
If the phone is first, then it slightly limits the PC and rebooting the
phone causes loss of contact with the PC.
If the PC is first you have to have dual ports on it (a few bucks of
hardware, plus configuration costs), then rebooting the PC causes the phone
to loose contact with the world. Not
Here is the 1.4.x version on centos 5 walk through.
http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce Nik
Sent: Friday, January 15, 2010 3:15 PM
On Fri, Jan 15, 2010 at 04:06:54PM +, Gordon Henderson wrote:
Have an intersting issue whem migrating a site from Zap on 1.3 to DAHDI on
1.4.. Nothing special about the hardware - older TDM400 card, 2 red
modules fitted...
Both channels work fine under 1.2/Zaptel. With 1.4/DAHDI both
On Sat, 16 Jan 2010, Tzafrir Cohen wrote:
On Fri, Jan 15, 2010 at 04:06:54PM +, Gordon Henderson wrote:
Have an intersting issue whem migrating a site from Zap on 1.3 to DAHDI on
1.4.. Nothing special about the hardware - older TDM400 card, 2 red
modules fitted...
Both channels work
I have NVFaxDetect working 100% with Asterisk 1.6. Check out this
article in my blog on details of how I got it working:
http://cloudsconnected.com/?p=57
Good luck!
On Fri, 2010-01-15 at 11:28 +, --[ UxBoD ]-- wrote:
- Paul Scott p...@cpanel.net wrote:
Yeah sounds like you wanna use
We are using Polycom 550 and 650 phones and OSLEC echo cancellation software
with Asterisk. Occasionally, we get echo on our PRI phone calls. The echo is
always from our voice echoing back to us. How can I fix this echo? I have
tried installing the VPMADT032 module on our TE121 card, but
Peter Childs wrote:
Ok this has Probably been put to bed several time but never mind.
Elastix, Trixbox, or AsterixNow, or DIY (ie Ubuntu or whatever
installed with OpenPBX, Asterix etc by hand)
I've got a new server to run Asterix on and want to get it working
quickly and yet be
Noah I. Engelberth wrote:
Is there a way I can change the ring cadence on FXS lines on a system
using a Digium Wildcard TDM2400 card? I recently deployed a new phone
system, and the customer has a few POTS phones in areas where they
don't have data network services, so we're using the FXS
hin lee wrote:
We are using Polycom 550 and 650 phones and OSLEC echo cancellation
software with Asterisk. Occasionally, we get echo on our PRI phone
calls. The echo is always from our voice echoing back to us. How can
I fix this echo? I have tried installing the VPMADT032 module on our
On Fri, 15 Jan 2010, Hans Witvliet wrote:
If you connect your pc with GB-lan card to an dual-ported ip-phone, you
and up with an 100Mbps lan connection to your pc.
Only way to avoid that, is to insert a cheap second lan-card in your pc,
and connect your phone to the second lan, so your pc
Windows, yes, but used to be through 3rd party software. Doubt this
has changed as Windows has no focus on any useful network anything.
Linux, yes, and it's definitely not complicated. Probably take 2
minutes to setup if you already had bridge utils installed, maybe 5 if
you had to install the
Hello,
Does the cached values for realtime peers expire automatically?
I have rtcachefriends=yes in sip.conf. When the peer registers for the first
time it is cached. After the first registration if I modify the peer in the
database the new values are not used until I do a 'sip prune realtime
Looking at all the docs I can find Asterisks looks like it should be
able to do the job and a whole lot more.
This is for a small call centre so ideally we want all the features of
an average call centre, ACD, Call Recording, Queue's etc etc.
Any pointers on how to get started would be most
2010/1/15 --[ UxBoD ]-- ux...@splatnix.net
- --[ UxBoD ]-- ux...@splatnix.net wrote:
- Paul Scott p...@cpanel.net wrote:
Yeah sounds like you wanna use NVFaxDetect
it would allow you to add something like exten =
fax,1,Swift(number
has changed); to your inbound call
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