On Wed, 24 Jun 2009 15:43:14 -0400, Leah Newmark
lnewm...@capalon.com wrote:
I also have noticed odd behavior. When I edit an AGI, the changes aren't
always showing up in the running of the script via asterisk.
May be a total red herring (I'm using an old version of Trixbox)
but if I edit my
On 25/06/2009, David Quinton gna...@bizorg.co.uk wrote:
May be a total red herring (I'm using an old version of Trixbox)
but if I edit my PHPs on a Windows machine and upload using FTP, they
will only run if I fire up Nano and save the file on the Asterisk box.
I haven't used TrixBox, but
Tzafrir Cohen,
if mixing hand-written configs with GUI-configs is not 'good practise',
then how to build a scalable Asterisk IP-PBX where the customer is not
100% dependent of the implementer ?
Like I already said, I got the remark To add a new phone, I do not want
to be forced to call you. And
On Thu, Jun 25, 2009 at 11:02:44AM +0200, jonas kellens wrote:
Tzafrir Cohen,
if mixing hand-written configs with GUI-configs is not 'good practise',
then how to build a scalable Asterisk IP-PBX where the customer is not
100% dependent of the implementer ?
Like I already said, I got the
We have several types of phones,
cisco 7940/7960
aastra 55i/9113i/
grandstream gxp2010
I want to be able to implement hotdesking where an agent will logon to
any phone. I got all of that working, without having to reboot phones,
but then hit a brick wall.
Voicemail.
I still want each phone
If you plan it right from the start, FreePBX can save hell lot of time.
Instead of fixing in include files, you can also create custom contexts from
within the GUI now, i am sure there is a module for that as well. As said
above, either stick fully to GUI or fully to manual configurations. Ugly
I feel a great preference for sticking to manually editing
the .conf-files. But if I define in the contract that changes to the
Asterisk-PBX need to be done by me, I force a maintenance cost towards
the customer and that is not always what is requested...
On Thu, 2009-06-25 at 15:26 +0530,
Asterisk RealTime is perfect for this as you can built them a simple web
interface to make insertions into your sip table.
Ish
jonas kellens wrote:
Tzafrir Cohen,
if mixing hand-written configs with GUI-configs is not 'good
practise', then how to build a scalable Asterisk IP-PBX where the
On Wed, Jun 24, 2009 at 05:01:04PM -0400, Steve Totaro wrote:
In FreePBX there are whatever_custom.conf files that are not touched when
changes are made in the GUI.
A GUI for Asterisk does not necessrily imply FreePBX. There are
certainly other ways to do that.
For instance:
* asterisk-gui:
On Thu, 25 Jun 2009, jonas kellens wrote:
I feel a great preference for sticking to manually editing
the .conf-files.
Then why did you ask for a GUI?
But if I define in the contract that changes to the
Asterisk-PBX need to be done by me, I force a maintenance cost towards
the customer
Hi All;
Did anyone used iaxclient? I would like to know how is the voice quality?
OS to be used Microsoft.
Regards
Bilal
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Julian Lyndon-Smith wrote:
We have several types of phones,
cisco 7940/7960
aastra 55i/9113i/
grandstream gxp2010
I want to be able to implement hotdesking where an agent will logon to
any phone. I got all of that working, without having to reboot phones,
but then hit a brick wall.
jonas kellens wrote:
So, what about this 'iksemel' ??
On RedHat based distros, you can install EPEL and the Dag Wieers repos in order
to get extra things like iksemel.
Leif Madsen.
http://www.leifmadsen.com
http://www.oreilly.com/catalog/asterisk
Why ask for a GUI ?
-- I do not ask for one... my customer does.
Realtime (did some reading) + custom webinterface seems indeed one of
the best solutions.
Then the thing is that I am not a programmer. I am a graduated network
engineer with affection towards VoIP and OpenSource (Asterisk). I do
You have to still have all of the line2 entries in the config file and they
have to be set to .
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Wednesday, June 24, 2009 4:12 PM
To:
Trying to accomplish something that seems simple enough but I've tried
everything I can think of but I cannot get an AGI Transfer to work.
Seems simple enough
$AGI-exec('transfer','SIP/101');
and here's the resultant Debug:
AGI Rx EXEC transfer SIP/101
-- AGI Script Executing Application:
Thanks for the suggestion, but I'm editing directly on the server
I've been doing AGIs for, what, 4 years now? I have never been *this* stumped!
__
On Wed, 24 Jun 2009 15:43:14 -0400, Leah Newmark
lnewmark at capalon.com
Take a look at this:
/var/lib/asterisk/agi-bin/olehphone# head incoming.php displays:
#!/usr/bin/php
?
Running it shows this:
/var/lib/asterisk/agi-bin/olehphone# ./incoming.php
#!/usr/bin/php5 -q
Is that normal behavior if php5 is the library installed? Seems very odd
to me.
LN
The script runs fine command line.
I have edited in the past to try as /usr/bin/php -q and it didn't help.
Right now, it's not even reading the changes. I must be missing
something very obvious...
LN
asterisk-users-requ...@lists.digium.com wrote:
Message: 16
Date: Wed, 24 Jun 2009 17:17:59
Asterisk-server behind Endian-firewall: SIP-aware, 5060 + RTP-ports
opened and 5060 forwarded to Asterisk (192.168.2.2)
Can someone see why SIP-registration fails ??
register = 092779077:x...@85.119.188.3
[3starsnet]
type=peer
host=85.119.188.3
username=092779077
secret=
fromuser=092779077
Mike,
1. Remove the 'line 2' entries completely from the SEPXX.XML file.
2. Change the 'Version' tag in the SEPXX.XML file. You need only change
one digit; I usually just increment the last digit.
(version1.0.0.0-9/version).
3. Restart the phone (Settings - **#**).
4. This
David's directions will work on a 7941/7961, not the 7940/7960. You do have
to keep the line configuration for the 79x0 series phones in the
SIP${MAC}.cnf file.. I have not tested setting them to , but I know if
you telnet into the phone they will show UNPROVISIONED as the setting.
You can also
Hi all,
On my *NOW beta box, I was able to assign an IVR an extension (801, etc.)
With the new 1.5 *NOW and freepbx, I can create an IVR, but how do I assign
it an extension, so I can dial or transfer users to that IVR? Thanks in
advance for the help. I checked the freepbx docs but didn't find any
Probably unrelated, but its bad practice to use short tags ? ? /
?=$var;? use ?php ? / ?php echo $var;? instead.
Incase short_open_tag = Off in php.ini
On Thu, Jun 25, 2009 at 11:08 AM, Leah Newmarklnewm...@capalon.com wrote:
Take a look at this:
/var/lib/asterisk/agi-bin/olehphone# head
Thanks...completely irrelevant in my case though. I installed and set up
php on my end, so I have short_open_tag to be on :)
Any other ideas anyone?
Leah Newmark
VoIP Programmer
Capalon Communications
_
Probably unrelated, but
I think I found the part of my AGI the script is stuck at.
The #!/usr/bin/php command was fine. What the agi debug I believe is
displaying is the output of this:
$in = fopen(php://stdin,r);
Which explains what I thought was cached
-- the same #!/usr/bin/php5 -q command repeatedly failing was
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I have a license for Allison-8kHz and two concurrent port licenses that
I purchased from Cepstral at the end of last year. I just got around to
installing to my * 1.6.0.10 machine.
I've decided that the best way for me to integrate the two would be
SIP-registration errors are solved by restarting the Asterisk-server.
But I expect them to return in time...
I can make call now, but the other end does not hear me. So problem with
RTP-flow...
Can someone guide me to the solution ?
On the Asterisk-server I have this (iptables):
-A
Hi all,
I'm testing the persistent dynamic queue members functionality on
1.6.0.10. The queue members are agents defined in the agents.conf file.
When I issue an asterisk restart and check the queue members again on
the CLI, all of them are listed as /invalid/ and there is no way to
change
Barry L. Kline wrote:
I'm going to end up buying more ports from Digium but I'd like to also
use the existing voice/port licenses that I currently have. Is this
possible? Is there anyway to migrate the licenses to the Digium
implementation of Cepstral?
There is no need; your existing
Miguel Molina escribió:
Hi all,
I'm testing the persistent dynamic queue members functionality on
1.6.0.10. The queue members are agents defined in the agents.conf
file. When I issue an asterisk restart and check the queue members
again on the CLI, all of them are listed as /invalid/ and
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Kevin P. Fleming wrote:
There is no need; your existing Cepstral-supplied licenses will continue
to operate, and will be added to any Digium-supplied licenses you
purchase and activate.
Thanks Kevin.
So I shouldn't worry about this?
Hi,
I am using a call file formated like this:
Channel: local/12125557...@outbound/n
Callerid: 12125551212
Context: detect
Extension: s
Priority: 1
This sends the call into the dialplan at the [outbound] context. In
[outbound], I have:
[outbound]
exten = _1.,1,Dial(SIP/${ext...@flowroute,43)
If
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