Very good. Sangoma cards are great. Get the a101d though. Nice wee review:
http://www.smithonvoip.com/new-voip-products/sangoma-a101d-single-port-t1e1-card/
Voipon are great guys too. We resell for them.
On 07/08/07, Rory Campbell-Lange [EMAIL PROTECTED] wrote:
We will be connecting our
the old a101
though?
Regards
Rory
On 07/08/07, Gavin Henry ([EMAIL PROTECTED]) wrote:
Very good. Sangoma cards are great. Get the a101d though. Nice wee review:
http://www.smithonvoip.com/new-voip-products/sangoma-a101d-single-port-t1e1-card/
Voipon are great guys too. We resell
Please see the official tracker in the Digium buglist:
http://bugs.digium.com/view.php?id=5768
Here are the schemas we did for OpenLDAP:
http://bugs.digium.com/file_download.php?file_id=14842type=bug
http://bugs.digium.com/file_download.php?file_id=14841type=bug
Also, for Novell eDirectory,
.
OK, maybe I need to go and read more about Astirectory.
Regards
Abhishek
On 8/24/07, Gavin Henry [EMAIL PROTECTED] wrote:
Please see the official tracker in the Digium buglist:
http://bugs.digium.com/view.php?id=5768
Here are the schemas we did for OpenLDAP:
http
I see it is res_config_ldap. You'd be much better using the latest
version in the bug tracker.
On 27/08/07, Gavin Henry [EMAIL PROTECTED] wrote:
On 26/08/07, Abhishek M S [EMAIL PROTECTED] wrote:
Dear Mr Galvin,
Gavin ;-)
Thank you for the links. Had gone through the bug tracker before
? or
do I have to do LIBS=-lldap export LIBS ./configure before that? My asterisk
version is 1.2.6.
This Digium version is for 1.4.x, not 1.2
Thanks in advance,
Abhishek
On 8/27/07, Gavin Henry [EMAIL PROTECTED] wrote:
I see it is res_config_ldap. You'd be much better using the latest
,
Abhishek
On 8/27/07, Gavin Henry [EMAIL PROTECTED] wrote:
On 27/08/07, Abhishek M S [EMAIL PROTECTED] wrote:
Dear Mr Galvin,
Gavin! ;-)
As of today I am using the res_config_ldap of Astirectory in my test
Asterisk system to connect to a test LDAP database of my University
No probs.
On 29/08/2007, Abhishek M S [EMAIL PROTECTED] wrote:
Dear Mr Gavin,
Thank you once again. Will have to talk it over with my prof before
upgrading to Asterisk 1.4. The productive system is currently running on
1.2.6.
Thanks
Abhishek
On 8/28/07, Gavin Henry [EMAIL PROTECTED
Dear All,
How long should it take before a exten = h,1,Hangup() kicks in,
versus a exten = s,n,Hangup()
I'm just about to test, but thought I'd ask.
--
http://www.suretecsystems.com/services/openldap/
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and wondered
if this might the problem in hanging up a zap call.
On 8/30/07, Gavin Henry [EMAIL PROTECTED] wrote:
Dear All,
How long should it take before a exten = h,1,Hangup() kicks in,
versus a exten = s,n,Hangup()
I'm just about to test, but thought I'd ask.
--
http
On 17/03/2008, Faraz Khan [EMAIL PROTECTED] wrote:
Good Idea and done. It is now available here:
http://www.voip-info.org/wiki/view/LDAP
The correct LDAP Schema is included:
/asterisk-1.6.0-beta4/contrib/scripts/asterisk.ldap-schema
and
/asterisk-1.6.0-beta4/contrib/scripts/asterisk.ldif
What about using RealTime LDAP in 1.6? That woudl be much faster than a RDBMS.
2008/6/3 Sherwood McGowan [EMAIL PROTECTED]:
Mindaugas Kezys wrote:
Thank you for your opinion.
Then my question would follow: how to build human-friendly system which will
use GUI and lets user use that system
2008/6/4 Tzafrir Cohen [EMAIL PROTECTED]:
On Wed, Jun 04, 2008 at 10:45:13AM +0100, Gavin Henry wrote:
What about using RealTime LDAP in 1.6? That woudl be much faster than a
RDBMS.
If performance is such a major issue, why not use explicit queries?
realtime has overhead even in extensions
What model in the Polycom or Aastra range is the 360 level with?
2008/6/6 Chris Bagnall [EMAIL PROTECTED]:
When I pushed some vendors for prices there was only a tiny gap between
the 300 and 360. Would suggest looking hard at the 360 always...
Interesting... here in the UK the price
2008/6/7 Gavin Henry [EMAIL PROTECTED]:
What model in the Polycom or Aastra range is the 360 level with?
Probably the IP601:
http://www.voipon.co.uk/polycom-soundpoint-ip601-p-121.html
and 57i:
http://www.voipon.co.uk/aastra-57i-ip-phone-p-420.html
Snom 360:
http://www.voipon.co.uk/snom-360
Hi All,
Is this still the cause in 1.4 and 1.6 as per:
http://www.voip-info.org/wiki/view/Asterisk+and+OpenSER+integration.
Do people recommend OpenSER in front for deployments bigger than 300 end points?
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2008/6/12 Syed Nasruddin [EMAIL PROTECTED]:
HI,
I am using TDM800P Digium Card with Asterisk 1.4.* version. I have fair
command over Asterisk up till now and have run it in different scenarios
such as Call Center Solution, PBX solution.
There is a requirement to use Asterisk only as
2008/6/13 Mark Hamilton [EMAIL PROTECTED]:
Hi,
How can I rotate /var/log/asterisk/cdr-custom/Master.csv nightly by date?
Logrotate on a *nix box.
--
http://www.suretecsystems.com/services/openldap/
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months later. :)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gavin
Henry
Sent: June 13, 2008 4:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] cdr-custom/Master.csv rotation
2008/6/13 Mark
the same question on the list 3
months later. :)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gavin
Henry
Sent: June 13, 2008 4:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] cdr-custom/Master.csv
2008/6/16 Syed Nasruddin [EMAIL PROTECTED]:
Thanks for the link. I think I will be using this product.
It's very, very good. You can hook it up to MySQL instead of sqlite if
needed, just e-mail support.
--
http://www.suretecsystems.com/services/openldap/
LDAP for account and Mysql for
extensions/queues.
Quoting Gavin Henry [EMAIL PROTECTED]:
On 17/03/2008, Faraz Khan [EMAIL PROTECTED] wrote:
Good Idea and done. It is now available here:
http://www.voip-info.org/wiki/view/LDAP
The correct LDAP Schema is included:
/asterisk-1.6.0-beta4
Google Apps version might.
2008/6/25 Marc Smith [EMAIL PROTECTED]:
Hi,
Anyone using Asterisk IMAP voicemail storage with Google Apps / GMail
IMAP? If so, does their IMAP implementation support any kind of
master user (Dovecot) abililty? Good? Bad?
--Marc
What did you do to setup a button for alerts?
Thanks.
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To
2008/7/2 Loic Didelot [EMAIL PROTECTED]:
Depends on the phone.
On many devices you can setup buttons to call a url. Thats what I did.
Ah, yes. Would be a good thing to implement here. Then you can do
anything, like a support ticket etc.
Cheers.
___
We do as do Gradwell.com
--
http://voip.suretecsystems.com/
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AstriCon 2008 - September 22 - 25 Phoenix, Arizona
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asterisk-users mailing
Dear All,
What setup would you recommend for making VoIP calls whilst bringing
latency down between offices at:
* Edinburgh
* Kuala Lumpur
* Singapore
* Tokyo
* Seoul
* Beijing
* San Francisco
Some of the Asia offices are 300ms some 200ms.
Any advice greatly apreciated.
Thanks.
Thanks all for your suggestions.
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To UNSUBSCRIBE or update
Or provide both solutions - let the offices call each other via VoIP, but
if too laggy, fall-back to VoIP - PSTN... (- VoIP)
How can you test for this precall?
Cheers.
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That looks cool. Will have a play.
On 10/18/08, Ming Yong [EMAIL PROTECTED] wrote:
Anael,
You should take a look at Druid (Open Source Unified Communications)
Project based on Asterisk that has complete LDAP backend and Zimbra
connector.
It's an open source project we are looking for
The LDIF needs updating as it's not a working example. I'll have one
next week. I'll release an updated schema too.
Gavin.
On 10/18/08, Tilghman Lesher [EMAIL PROTECTED] wrote:
On Saturday 18 October 2008 02:30:16 Anael DIAZ wrote:
I need help in implementing Asterisk with LDAP. I' ve
Dear All,
Is it possible to install * in front of a Avaya IP 406 system via a T
connector E1 tap so it's external to the Avaya system?
We would like to record upto 60 channels (2 * ISDN30e). This may increase
later.
Also, could the calls go into the cdr for retrieval/browsing later?
What
-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Gavin Henry
Sent: April 20, 2007 9:07 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Passive E1 Pri Tap for Voice Recording
Dear All,
Is it possible to install * in front of a Avaya IP 406 system
:-)
Good luck,
David
On 4/20/07, Gavin Henry [EMAIL PROTECTED] wrote:
Dear All,
Is it possible to install * in front of a Avaya IP 406 system via a T
connector E1 tap so it's external to the Avaya system?
We would like to record upto 60 channels (2 * ISDN30e). This may increase
later.
Also
On 21/04/07, Leo Ann Boon [EMAIL PROTECTED] wrote:
Gavin Henry wrote:
Dear All,
Is it possible to install * in front of a Avaya IP 406 system via a T
connector E1 tap so it's external to the Avaya system?
Voicetronix has an open sourced solution using their OpenPRI in Hi-Z mode.
http
On 24/04/07, sravana [EMAIL PROTECTED] wrote:
Hi all,
I have installed Asterisk in my PC. I am running one LDAP server. I
could not get enough documents which would help me to intergrate the
existing user Database. Say I have a LDAP directory which has all the
numbers and user details I should
Dear All,
Why does my configure fail like so:
checking for pg_config... /usr/local/pgsql/8.2.4/bin/pg_config
checking for PQexec in -lpq... no
configure: ***
configure: *** The PostgreSQL installation on this system appears to be broken.
configure: *** Either correct the installation, or run
Dear All,
Where can I find a res_pgsql.conf and some sql to insert for tables etc.?
Are all db res things to be done via odbc now?
Why was this included with no docs or sample conf?
Thanks,
Gavin.
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PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gavin Henry
Sent: 05 May 2007 22:31
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk 1.4.4 and Custom Postgres 8.2.4
(checkingfor PQexec in -lpq... no)
Dear All,
Why does my configure fail like so:
checking for pg_config... /usr
Hi All,
Can anyone recommend any test kit that you can hook up your Pri/Bri
cards to without having actual ISDN in your office. For example
testing an * system before it goes to a clients office.
Thanks,
Gavin.
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,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Gavin Henry
Envoyé : mercredi 9 mai 2007 09:40
À : asterisk-users@lists.digium.com
Objet : [asterisk-users] Testing ISDN T1/E1 Bri and Pri etc.
Hi All,
Can anyone recommend any test kit that you
] Behalf Of Gavin Henry
Sent: Wednesday, May 09, 2007 2:09 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: RE : [asterisk-users] Testing ISDN T1/E1 Bri and Pri etc.
On 09/05/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi Gavin,
A second
Hi All,
What do you recommend? I was looking at:
http://www.voipon.co.uk/sangoma-a200-fxo-fxs-analogue-card-pci-express-p-393.html
But it will be 3 PCI slots.
Thanks,
Gavin.
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asterisk-users
:[EMAIL PROTECTED] On Behalf Of Gavin
Henry
Sent: Wednesday, May 09, 2007 3:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] 10 FXS - Channel Bank or PCI Card?
Hi All,
What do you recommend? I was looking at:
http://www.voipon.co.uk/sangoma-a200-fxo-fxs
On 09/05/07, Robert Hajime Lanning [EMAIL PROTECTED] wrote:
I would look into one of these:
http://www.digium.com/en/products/hardware/analogcards.php
I've seen those too ;-)
quote who=Gavin Henry
Hi All,
What do you recommend? I was looking at:
http://www.voipon.co.uk/sangoma-a200
On 09/05/07, cb [EMAIL PROTECTED] wrote:
On May 9, 2007, at 3:45 PM, Gavin Henry wrote:
http://www.voipon.co.uk/sangoma-a200-fxo-fxs-analogue-card-pci-
express-p-393.html
But it will be 3 PCI slots.
Just to clarify in case you didn't already realize it. It doesn't
actually *use* 3 PCI
Dear All,
I have a tiny dial plan like:
[testing]
exten = 454,s,Ringing()
exten = 454,n,Wait(4)
exten = 454,n,Dial(SIP/[EMAIL PROTECTED]:5605,10)
exten = 454,n,Hangup
This connects fine when I dial 454 from any extension in my system,
but there is never any audio?
Where can I start to look
On 23/05/07, Alex Balashov [EMAIL PROTECTED] wrote:
Gavin,
Hi.
Does the Asterisk server's route to 192.168.45.183 traverse a firewall or
router that may be blocking non-SIP ports that are dynamically allocated?
Nope, all internal.
SDP -- part of the SIP INVITE transaction
a softphone does, in
the SDP session.
Gavin Henry wrote:
Dear All,
I have a tiny dial plan like:
[testing]
exten = 454,s,Ringing()
exten = 454,n,Wait(4)
exten = 454,n,Dial(SIP/[EMAIL PROTECTED]:5605,10)
exten = 454,n,Hangup
This connects fine when I dial 454 from any extension in my
allow=g711
in sip.conf that it finally started working for me.
That may not be your exact problem, but my guess would be a CODEC issue if
it's not your firewall.
I'll check this out, thanks.
-- Nick
On Wed, 23 May 2007, Gavin Henry wrote:
Dear All,
I have a tiny dial plan like
Dear All,
With the standard Voicemail system, is it possible to have your
Busy/Unavailable messages only apply during say 9-5, then another
message saying you've gone home after that time?
It might be just a case of user training, that they change their
message if they need this feature.
A
Hi,
This contacted call has no audio, any ideas?
The conference suite from another provider on internal IP is waiting
for an ACK on port 5605, but * is sending it back to port 2289
Internal between Asterisk and another Conference suite:
* SIP Call
Direction: Outgoing
Call-ID:
This is what is shown when the call connects with:
sip show channel
The conference suite from another provider on internal IP is waiting
for an ACK on port 5605, but * is sending it back to port 2289
Internal between Asterisk and another Conference suite:
* SIP Call
Direction:
Dear all,
I think this is common, or at least how it is supposed to be, but
whening dialing over a ZAP channel, it's taking around 5~ seconds to
ring on the over end, likewise inbound.
This is just with a normal Dial command.
Are there any ways to tweak this?
Thanks,
Gavin.
On 01/06/07, Gordon Henderson [EMAIL PROTECTED] wrote:
On Fri, 1 Jun 2007, Gavin Henry wrote:
Dear all,
I think this is common, or at least how it is supposed to be, but
whening dialing over a ZAP channel, it's taking around 5~ seconds to
ring on the over end, likewise inbound
On 04/06/07, Caio Zanolla [EMAIL PROTECTED] wrote:
Hi everyone,
in ldap realtime sip peers i need fullcontact set to
sip:[EMAIL PROTECTED] for asterisk to correctly match the peers (at least
for the natted peers to reach them)...
anyway, how do I populate fullcontact on the fly with
Dear all,
We seem to be getting phantom calls when a inbound caller via the
legacy pbx hangups before
the SIP handsets have answered. The extensions also seem to hear
ringing on the lines too sometimes.
SIP Inbound
On 07/06/07, Stephen Bosch [EMAIL PROTECTED] wrote:
Gavin Henry wrote:
Dear all,
We seem to be getting phantom calls when a inbound caller via the
legacy pbx hangups before
the SIP handsets have answered. The extensions also seem to hear
ringing on the lines too sometimes
Hi all,
In a pure VoIP env, what is the current state of do's and don't s of
virtualizing * in order to provide multiple separate instances, say
for hosting lots of Asterisk-gui/FreePBX/a-n-other gui?
I've read lots of threads going back to 2007 and I'm in the general
option that kvm is the way
2009/2/27 John Todd jt...@digium.com:
On Feb 26, 2009, at 6:01 PM, Senad Jordanovic wrote:
Gavin Henry wrote:
Hi all,
In a pure VoIP env, what is the current state of do's and don't s of
virtualizing * in order to provide multiple separate instances, say
for hosting lots of Asterisk-gui
Just transfer them to your meetme extension after you've called them.
Just like you would transfer someone who has called you.
* will then put them into that conference.
Thanks.
On 08/03/2009, Sven Geggus use...@fuchsschwanzdomain.de wrote:
Hello,
setting up Meetme was very easy. I jut added
Hi All,
We've got msidn configured:
Port 1: TE-mode BRI S/T interface line (for phone lines)
- Protocol: DSS1 (Euro ISDN)
- childcnt: 2
mISDN_close: fid(3) isize(131072) inbuf(0x8fd5060) irp(0x8fd5060)
iend(0x8fd5060)
and running on Asterisk 1.4.21.2:
pbx*CLI misdn show stacks
2009/3/12 Paulo Santos paulo.r.san...@sapo.pt:
Gavin Henry wrote:
Hi All,
We've got msidn configured:
Port 1: TE-mode BRI S/T interface line (for phone lines)
- Protocol: DSS1 (Euro ISDN)
- childcnt: 2
I don't know if it depends on the card, but in my case I need to set
2009/3/12 Giorgio Incantalupo gincantal...@fgasoftware.com:
Hi Gavin,
if you can make and receive calls it works...do not worry if your line
is shown as DOWN, some telco turns it off but it works without problem.
Remember to ask your telco for the right signalling and set it the right
way
Dear all,
I'm currently researching options for a MT asterisk gui/system for a
small business centre that will have 12 units in it. Each unit will be
configured for one extension.
The system there will have a max of 12 concurrent calls to PSTN
provided via an ADSL/SDSL link to our VoIP provider
2009/3/17 Gordon Henderson gordon+aster...@drogon.net:
On Mon, 16 Mar 2009, Gavin Henry wrote:
Dear all,
I'm currently researching options for a MT asterisk gui/system for a
small business centre that will have 12 units in it. Each unit will be
configured for one extension.
The system
2009/3/17 Gordon Henderson gordon+aster...@drogon.net:
On Tue, 17 Mar 2009, Geraint Lee wrote:
We can put about 9/10 calls using SIP/gsm through our BT Business Network
ADSL package connection (832kbit upstream, £65/month) before you notice
the
quality starting to drop, but you could always
A2billing is a good fit for that then. Yeah, voipon. Thanks for the
input Gordon. Maybe worth hooking up offline if we're doing similar
stuff.
Gavin.
On 17/03/2009, Gordon Henderson gordon+aster...@drogon.net wrote:
On Tue, 17 Mar 2009, Gavin Henry wrote:
2009/3/17 Gordon Henderson gordon
Yeah, I've experienced that. But what can you do other than stick woth
a fat codec.
On 17/03/2009, Gordon Henderson gordon+aster...@drogon.net wrote:
On Tue, 17 Mar 2009, Gavin Henry wrote:
2009/3/17 Gordon Henderson gordon+aster...@drogon.net:
On Tue, 17 Mar 2009, Geraint Lee wrote:
I know
Hi all,
Has anyone put * in between an Avaya and Cisco system to connect two
offices together?
I was thinking about adding a SIP trunk on each side and getting
Asterisk to pass calls between them. There is a leased line for
bandwidth.
Any tips/ideas on whether this is possible or dumb?
Thanks.
BTW, what's the recommended production version of Asterisk source
you'd recommend, the latest 1.4 or 1.6?
In fact, nevermind. This is asked so many times I'll hit the archives.
Cheers.
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2009/4/3 John Todd jt...@digium.com:
On Apr 3, 2009, at 7:40 AM, Gavin Henry wrote:
Hi all,
Has anyone put * in between an Avaya and Cisco system to connect two
offices together?
I was thinking about adding a SIP trunk on each side and getting
Asterisk to pass calls between them
Hi all,
What other open source tools are people using for this? I was looking
at Openfire and their asterisk plugin.
Is it easy to roll your own with res_jabber.so ??
Thanks.
--
Sent from my mobile device
http://www.suretecsystems.com/services/openldap/
2009/4/20 jonas kellens jonas.kell...@telenet.be:
Please, is there anyone who can help me with this zaptel -- Dahdi -problem
??
Will chan_dahdi.conf work with zaptel.conf ? Will Asterisk be able to
communicate with the Digium TDM pci-card ?
Or do I need to compile dahdi and recompile
2009/4/23 Matt Riddell li...@venturevoip.com:
On 18/04/2009 2:28 a.m., Gavin Henry wrote:
Hi all,
What other open source tools are people using for this? I was looking
at Openfire and their asterisk plugin.
Is it easy to roll your own with res_jabber.so ??
I used openfire in the past
Is your box on a public ip or via nat? If eth0 isn't the ip you set it
to bind on it will ignore it.
I mean, is your * box on an internal address?
On 02/05/2009, jonas kellens jonas.kell...@telenet.be wrote:
I have connected my Asterisk-box directly to my internetconnection. I
have disabled my
Why not use OpenSIPS or Kamailio in stateful mode?
You will need to look at how media is handled though, but a SIP proxy
will work easily.
On 13/05/2009, Adrian Marsh adrian.ma...@ubiquisys.com wrote:
Hi David,
Thanks for the reply. That's pretty much what I've already tried, but
with no
Is there any document on the reasons for the 1.6.0 and 1.6.1 branches?
I remember reading something but can't find it again.
Was it stability versus new features?
I'm currently playing with 1.6.1
Gavin.
On 19/05/2009, Benny Amorsen benny+use...@amorsen.dk wrote:
Miguel Molina
Where do they currently change their password? If it's somewhere you
control, why not add some to create the realmed password?
Gavin.
On 02/06/2009, John A. Sullivan III jsulli...@opensourcedevel.com wrote:
Hello, all. I'm afraid I've been dropped into the deep end even though
I am an
It also depends where you are registering your users. If merely using
Asterisk for a media server, do the auth via LDAP in Kamailio, which
will just use the userPassword attribute (or however the Kamailio LDAP
module binds to check auth or what you script it to do) then a normal
password change
Sorry, lastly I defined it as auxilary to do exactly that; add it to
any existing entry.
Thanks.
On 02/06/2009, John A. Sullivan III jsulli...@opensourcedevel.com wrote:
Hello, all. I'm afraid I've been dropped into the deep end even though
I am an Asterisk novice. I've set up a few tiny,
One last thing ;-) use OpenLDAP!
On 02/06/2009, John A. Sullivan III jsulli...@opensourcedevel.com wrote:
Hello, all. I'm afraid I've been dropped into the deep end even though
I am an Asterisk novice. I've set up a few tiny, tiny systems in the
past and have now been asked to pull together
2009/6/2 John A. Sullivan III jsulli...@opensourcedevel.com:
Most of the desktops are KDE and they use the KDE change password
facility. It works via pam I believe. Is there an Asterisk interface
with pam that would cause it to simultaneously change the Asterisk SIP
realm password? If there
2009/6/2 John A. Sullivan III jsulli...@opensourcedevel.com:
grin OpenLDAP isn't an option. And thanks very much for all the
responses. I've not had a chance to mock it up yet and see how it works
hands on. I am planning that the users ultimately interface SIP to
Kamailio and use Asterisk
2009/6/2 John A. Sullivan III jsulli...@opensourcedevel.com:
Thanks. I do appreciate the input as I am jumping into the deep end as
I said :)
On Tue, 2009-06-02 at 21:43 +0100, Gavin Henry wrote:
2009/6/2 John A. Sullivan III jsulli...@opensourcedevel.com:
grin OpenLDAP isn't an option
Hi,
Has anyone ever experienced crosstalk with sangoma analogue cards on 64bit?
We have exhausted every test to try and replicate this and find a
solution with Sangoma tech support, but we can not fix it.
We are about to try the card and four *seperate* UK BT lines in a 32bit system.
The
Hi,
Has anyone ever experienced crosstalk with sangoma analogue cards on 64bit?
We have exhausted every test to try and replicate this and find a
solution with Sangoma tech support, but we can not fix it.
We are about to try the card and four *seperate* UK BT lines in a 32bit system.
The
Every call as soon as the sangoma card is live.
Speak to Konrad on your techdesk for more info.
Thanks.
On 06/06/2009, Moises Silva moises.si...@gmail.com wrote:
Currently we have put in a temp OpenVOX tdm400 card and it works
perfectly. As soon as we swap that and use Sangoma via wanrouter
That is correct. That is the first test we did.
On 07/06/2009, Moises Silva moises.si...@gmail.com wrote:
On Sat, Jun 6, 2009 at 3:18 PM, Gavin Henrygavin.he...@gmail.com wrote:
Every call as soon as the sangoma card is live.
Speak to Konrad on your techdesk for more info.
Thanks.
I'll
quote who=Sharon Lim
hi there,
i am wondering can meetme.conf able to support diffferent context. Cause
currently, it has [rooms] context. ]
is it possible to have same conference number with different context?
thanks
Try it and see ;-)
--
Kind Regards,
Gavin Henry.
Open Source. Open
Hi All,
Has anyone hooked up * as an extension/trunk of an Avaya system that has
around 2 ISDN30e's.
Trying to add 100 extensions to one of our systems, but not sure where to
start reading.
Thanks.
--
Kind Regards,
Gavin Henry.
___
--Bandwidth
There a realtime LDAP driver now in 1.6beta2
On 23/01/2008, Cavalera Claudio Luigi [EMAIL PROTECTED] wrote:
Hello,
I've found this information about asterisk and LDAP:
http://www.voip-info.org/wiki/index.php?page=Asterisk+LDAP
which can be out of date.
I'm trying this
This has to be an Asterisk based appliance no?
http://www.truecall.co.uk/acatalog/trueCall_Features.html
Looks pretty easy to setup using AstLinux or similar.
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asterisk-users
Exactly. I was thinking that a similar service would be a good addon
as an option to an ITSP.
Gavin.
On 18/07/2009, Steve Totaro stot...@totarotechnologies.com wrote:
On Sat, Jul 18, 2009 at 4:36 AM, Alan Lord (News)
alansli...@gmail.comwrote:
On 18/07/09 00:35, Gavin Henry wrote:
This has
Yeah, and the fxs port too.
On 18/07/2009, Alan Lord (News) alansli...@gmail.com wrote:
On 18/07/09 00:35, Gavin Henry wrote:
This has to be an Asterisk based appliance no?
http://www.truecall.co.uk/acatalog/trueCall_Features.html
I saw this on the TV the other night. Couldn't believe how
Hi All,
Has anyone passed the tests using Asterisk:
http://www.btwholesale.com/pages/static/Products/Converged_Voice/IP_Exchange.html
I presume the same rules apply for scaling and possibly have
OpenSIPS/Kamailio on the front?
Thanks.
--
http://www.suretecsystems.com/services/openldap/
2009/7/31 Gordon Henderson gordon+aster...@drogon.net:
On Fri, 31 Jul 2009, Gavin Henry wrote:
Hi All,
Has anyone passed the tests using Asterisk:
http://www.btwholesale.com/pages/static/Products/Converged_Voice/IP_Exchange.html
Intersting. Looks like BT trying to become an ITSP
2009/7/31 Gordon Henderson gordon+aster...@drogon.net:
On Fri, 31 Jul 2009, Gavin Henry wrote:
Hi All,
Has anyone passed the tests using Asterisk:
http://www.btwholesale.com/pages/static/Products/Converged_Voice/IP_Exchange.html
Intersting. Looks like BT trying to become an ITSP
2009/7/31 Steve Howes st...@geekinter.net:
On 31 Jul 2009, at 08:22, Gavin Henry wrote:
Has anyone passed the tests using Asterisk:
BT guy we spoke to said yes : )
Good to know!
--
http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com
2009/7/31 Gordon Henderson gordon+aster...@drogon.net:
On Fri, 31 Jul 2009, --[ UxBoD ]-- wrote:
Gordon,
Cast your mind back as I had a similar issue ... changing the cable sorted
it for me!
Cursiously enough, I thought about that - but these were 2 brand new
cables out of packets and I
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