-Original Message-
From: Douglas Garstang
Sent: Tuesday, June 27, 2006 11:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Realtime Voicemail
I'm noticing that the documentation on the voip wiki for
voicemail and realtime voicemail
-Original Message-
From: Michiel van Baak [mailto:[EMAIL PROTECTED]
Sent: Tuesday, June 27, 2006 12:21 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Realtime Voicemail
On 12:13, Tue 27 Jun 06, Douglas Garstang wrote:
-Original Message-
From
-Original Message-
From: Douglas Garstang
Sent: Tuesday, June 27, 2006 12:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Realtime Voicemail
-Original Message-
From: Douglas Garstang
Sent: Tuesday, June 27, 2006 11:55
; I've got an older
CVS-HEAD build, pre 1.2, do you think my problems are polycom or asterisk
based?-Ryan
On 6/19/06, Kevin P.
Fleming [EMAIL PROTECTED]
wrote:
-
Douglas Garstang
[EMAIL PROTECTED] wrote: Polycom released their
SIP software version 1.6.6 for th
What's
up with realtime voicemail? I have been going thtough and testing each feature
that can be set as a column in the db, one by one.
Some
do work, and some don't.
Here's
some I have found that do work:
delete
envelope
maxmsg
review
saycid
and
here's some that simply don't work:
to come back, Doug! We miss you! 8)
--Rob
-Original Message-
From: Douglas Garstang [mailto:[EMAIL PROTECTED]
Sent: Monday, 26 June 2006 3:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion;
Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE
Is anyone getting '500 Internal Server' errors back from their Polycom phones
when Asterisk sends a SIP NOTIFY message to them?
I called Polycom tech support, who where utterly useless.
Of course Polycom won't officially support it anyway, as they only support
Asterisk Business Edition. We're
-Original Message-
From: Doug Lytle [mailto:[EMAIL PROTECTED]
Sent: Monday, June 26, 2006 11:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] '500 Internal Server' Error on
SIP NOTIFY
Douglas Garstang wrote:
Is anyone getting
-Original Message-
From: Moises Silva [mailto:[EMAIL PROTECTED]
Sent: Monday, June 26, 2006 2:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] AGI script can not print out
error message
toconsole
what do you mean by could not
Does anyone know of any startups using Asterisk? What about established
companies? Ones that are hiring would be nice :)
Doug.
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update
Cc:
Subject: Re: [Asterisk-Users] Asterisk Startups
Douglas Garstang wrote:
Does anyone know of any startups using Asterisk? What about
established companies? Ones that are hiring would be nice :)
Doug
Well, I hope some more jobs get posted. I took a look tonight, and there was 2
there.
-Original Message-
From: Matt Gibson [mailto:[EMAIL PROTECTED]
Sent: Sun 6/25/2006 11:25 PM
To: asterisk-users@lists.digium.com
Cc:
Subject:
I'm running 1.2.9.1, and I can't get caller id dialplan matching to work.
When calling from 9220370 to 1234, the following does not match.
exten = 9220370/1234,1,NoOp(${CALLERIDNUM})
exten = 9220370/1234,2,Answer
exten = 9220370/1234,3,Playback(tt-weasels)
However, when calling from 9220370 to
should be the second argument based on what works for me
Kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Friday, June 23, 2006 3:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users
Can someone
recommend the best way to view current calls in progress on the Asterisk
console?
Neither the 'show
channels' or 'sip show channels' commands are easy to read.
hestia*CLI show
channelsChannel
Location
State
Application(Data)
SIP/2944093-f9e2
(None)
Up Bridged
Current Calls
Whats wrong with show channels?
On 6/22/06, Douglas Garstang [EMAIL PROTECTED] wrote:
Can someone recommend the best way to view current calls in
progress on the
Asterisk console?
Neither the 'show channels' or 'sip show channels' commands
are easy to
read
the user registration when there is an
invite comming?
On 6/18/06, Aaron Daniel [EMAIL PROTECTED] wrote:
On Sat, 17 Jun 2006, Douglas Garstang wrote:
Good grief I hate Outlook webmail. I can't reply inline.
Switch to thunderbird
I'm using realtime for voicemail users, and for reasons that I don't yet
understand, when it doesn't get used for a while (like overnight), the first
connection attempt of the day will display this on the console.
Jun 21 07:54:00 ERROR[8112]: cdr_addon_mysql.c:159 mysql_log: cdr_mysql:
Unknown
-Original Message-
From: Patrick [mailto:[EMAIL PROTECTED]
Sent: Tuesday, June 20, 2006 12:05 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Conferencing with multiple servers
On Tue, 2006-06-20 at 15:22 +0100,
All,
Slightly off topic.
Polycom released their SIP software version 1.6.6 for their phones recently. I
was under the impression that this release fixed a previous limitation where
the phones would only watch 7 buddies, ie send 7 sip subscriptions to Asterisk.
I have configured a phone
/17/2006 1:06 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
Subject: Re: [Asterisk-Users] Voicemail with NFS
Douglas Garstang wrote:
I don't think unison is a workable solution. It doesn't scale. The
network
and consequently extremely efficient.
Simon
On 6/17/06, Douglas Garstang [EMAIL PROTECTED] wrote:
Mike,
I don't think unison is a workable solution. It doesn't scale.
The network and system load would
JR,
Are you sure that a ro mounted volume won't behave in the same fashion as a rw
mounted one when the NFS server is abruptly shut down?
Have you tried shutting down the NFS server? Does Asterisk recover from this?
Doug.
-Original Message-
From: JR Richardson [mailto:[EMAIL
balance
On Fri, 16 Jun 2006, Douglas Garstang wrote:
Unless you can guarantee that the system that is currently processing
a call will be the system that handles a transfer request from a phone, are the
same, then transfers will not work.
Incorrect
Other applications can handle it. Don't see why Asterisk can't. Mount the nfs
volume with the -soft option. Do a 'df -k' and you will see that the df command
will time out in a couple of seconds. Why can't Asterisk do the same?
-Original Message-
From: Ira
Anyone else get this while compiling zaptel? I'm guessing I have to modify my
kernel. Neato. :(
Does that mean that the zaptel module (I'm really after ztdummy), or this
xpp_zap thing won't be usable...?
Not that I have zaptel hardware, but it seems Asterisk won't compile itself
without zaptel
-Original Message-
From: Time Bandit [mailto:[EMAIL PROTECTED]
Sent: Thursday, June 15, 2006 4:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Executing a Function from AGI
I'm getting nowhere with this. Is it even possible to set
-Original Message-
From: Stefan Tichy [mailto:[EMAIL PROTECTED]
Sent: Friday, June 16, 2006 7:26 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Re: Executing a Function from AGI
On Thu, Jun 15, 2006 at 03:21:32PM -0600, Douglas Garstang
I have /var/spool/asterisk/voicemail NFS mounted from another server.
Everything is fine, until I simulate an NFS server failure, by shutting down
the NFS server process.
At this point, Asterisk becomes almost non-responsive. It won't even process a
'sip show peers' command correctly. It
PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Voicemail with NFS
you might want to try autofs to drive the nfs functions.
it'll make you
less susceptable as the filesystem won't be mounted full time
Douglas Garstang wrote:
I have /var
-Original Message-
From: Brian Capouch [mailto:[EMAIL PROTECTED]
Sent: Friday, June 16, 2006 3:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Voicemail with NFS
Douglas Garstang wrote:
I'll give this a try, but what happens when
-Original Message-
From: Brian Capouch [mailto:[EMAIL PROTECTED]
Sent: Friday, June 16, 2006 3:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Voicemail with NFS
Douglas Garstang wrote:
I hope someone isn't going to tell me
. However if someone
wants to leave a message, I'm not sure this will work.
On 6/16/06, Douglas Garstang [EMAIL PROTECTED] wrote:
-Original Message-
From: Brian Capouch [mailto:[EMAIL PROTECTED]
Sent: Friday, June 16, 2006 3:40 PM
To: Asterisk Users Mailing List - Non-Commercial
-Original Message-
From: Brian Capouch [mailto:[EMAIL PROTECTED]
Sent: Friday, June 16, 2006 3:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Voicemail with NFS
Douglas Garstang wrote:
I'll give this a try, but what happens when
On Friday 16 June 2006 16:14, Brian Capouch wrote:
Douglas Garstang wrote:
Douglas Garstang wrote:
I hope someone isn't going to tell me that the voicemail
directory going away is going to cause Asterisk to fall in a
heap on the floor.
Brian Capouch wrote:
You never give up
Garstang wrote:
Douglas Garstang wrote:
I hope someone isn't going to tell me that the voicemail
directory going away is going to cause Asterisk to fall in a
heap on the floor.
Brian Capouch wrote:
You never give up
Unless you can guarantee that the system that is currently processing a call
will be the system that handles a transfer request from a phone, are the same,
then transfers will not work.
Round robin DNS won't work at all. Every time you send out a SIP message, your
going to be sending it to a
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Wednesday, June 14, 2006 7:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] DUNDi Not Able to Handle Complex
FailoverSituations
On Wed, 14 Jun 2006, Douglas
-Original Message-
From: Watkins, Bradley
[mailto:[EMAIL PROTECTED] Behalf Of Watkins,
Bradley
Sent: Thursday, June 15, 2006 2:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] DUNDi Not Able to Handle Complex
FailoverSituations
Kevin
Fleming has said on numerous ocassions that this is known not to work, and is
not supported.
-Original Message-From: Benjamin Stocker
[mailto:[EMAIL PROTECTED]Sent: Tuesday, June 06, 2006 4:31
AMTo: asterisk-users@lists.digium.comSubject:
[Asterisk-Users] Asterisk
It
seems that I am having a heck of a time explaining my attempts at distributing
ACD Queues to the list. Here's one little problem, that's only a piece of the
puzzle.
dundi.conf:
180q
=
global_dundi_q_pbx1,100,IAX,dundi1:[EMAIL PROTECTED]/${NUMBER},nopartial180q
=
We need to make sure that all queue applications run on the correct system that
the user agents that own the queue application are registered to. So when a
server fails and the user agents register with their secondary server (which
will always be configured to be the same server for those
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Thursday, June 15, 2006 9:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] DUNDi Not Able to Handle
ComplexFailoverSituations
On Thu, 15 Jun 2006, Douglas
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Thursday, June 15, 2006 9:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] DUNDi Not Able to Handle
ComplexFailoverSituations
On Thu, 15 Jun 2006, Douglas
-Original Message-
From: Watkins, Bradley [mailto:[EMAIL PROTECTED]
Sent: Thursday, June 15, 2006 10:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] DUNDi Not Able to
HandleComplexFailoverSituations
Is it possible for you to
-Original Message-
From: Stephen Davies [mailto:[EMAIL PROTECTED]
Sent: Thursday, June 15, 2006 11:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] DUNDi Not Able to Handle
ComplexFailoverSituations
On 15/06/06, Douglas Garstang
The DundiLookup() application command seems to have been replaced by the
DUNDILOOKUP application function.
I'm wondering why, because the DUNDILOOKUP function doesn't set the TECH and
DEST variables. I edited the code and added a WEIGHT variable to the variables
set, but the DUNDILOOKUP
-Original Message-
From: Douglas Garstang
Sent: Thursday, June 15, 2006 12:51 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] DUNDi Not Able to Handle
ComplexFailoverSituations
-Original Message-
From: Stephen Davies
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Thursday, June 15, 2006 12:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] DUNDi Not Able to
HandleComplexFailoverSituations
On Thu, 15 Jun 2006, Douglas
-Original Message-
From: Douglas Garstang
Sent: Thursday, June 15, 2006 1:23 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] DUNDi Not Able to
HandleComplexFailoverSituations
-Original Message-
From: Aaron Daniel [mailto
Discussion
Subject: RE: [Asterisk-Users] DUNDi Not Able to
HandleComplexFailoverSituations
On Thu, 15 Jun 2006, Douglas Garstang wrote:
No... this last bit doesnt. My dundi.conf has:
180q =
global_dundi_q_pbx1,100,IAX,dundi1:[EMAIL PROTECTED]/${NUMBE
R},nopartial
180q
Hmmm. Not having much luck with this. I'm trying to call the DUNDILOOKUP
function and assign it to a variable in an AGI script.
I've tried setting with EXEC CMD and with SET VARIABLE. In both cases, it's
treating DUNDILOOKUP literally, rather than calling a funciton.
I've tried this:
EXEC Set
Python... but it doesn't matter. The examples I pasted where what I am sending
to stdout, so the scripting application shouldn't be an issue.
-Original Message-
From: Alexander Lopez [mailto:[EMAIL PROTECTED]
Sent: Thursday, June 15, 2006 3:31 PM
To: Asterisk Users Mailing List -
I'm getting nowhere with this. Is it even possible to set a variable to the
result of a function call in AGI???
-Original Message-
From: Douglas Garstang
Sent: Thursday, June 15, 2006 3:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users
-Original Message-
From: Martin Joseph [mailto:[EMAIL PROTECTED]
Sent: Tuesday, June 13, 2006 10:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] OPENSER / SER and Asterisk
On Jun 13, 2006, at 8:29 PM, Douglas Garstang wrote
the ser.cfg or
somethjing .. then it would be great.
Regards
Santosh Rao
Martin Joseph wrote:
On Jun 13, 2006, at 8:29 PM, Douglas Garstang wrote:
If you do this, and not have Asterisk in the call setup path, your
going to lose the ability to do a lot of features. What about
black
-Original Message-
From: Jean-Michel Hiver [mailto:[EMAIL PROTECTED]
Sent: Wednesday, June 14, 2006 1:47 AM
To: Santosh Rao; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] OPENSER / SER and Asterisk
Santosh Rao a écrit :
asterisk has a
Does anyone know where I can find some good DUNDi docs?
The ones are dundi.org are absolutely horrible.
The examples in dundi.conf are pretty much useless.
I still can't figure out why Digium can't write some good documentation. It's
their 'baby' after all. This really drives me nuts and pisses
I have three Asterisk boxes.
Each has the following in dundi.conf:
180net = dundi_local,0,IAX,dundi:[EMAIL PROTECTED]/${NUMBER},nopartial
180q = dundi_q_pbx1,1,IAX,dundi:[EMAIL PROTECTED]/${NUMBER},nopartial
180q = dundi_q_pbx2,2,IAX,dundi:[EMAIL PROTECTED]/${NUMBER},nopartial
180q =
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Wednesday, June 14, 2006 9:34 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] DUNDi Docs
On Wed, 14 Jun 2006, Douglas Garstang wrote:
The examples in dundi.conf
-Original Message-
From: Watkins, Bradley [mailto:[EMAIL PROTECTED]
Sent: Wednesday, June 14, 2006 10:17 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] DUNDi Docs
Yes, what is it you attempting? I use DUNDi extensively,
though you
It has also just become glaringly apparent to me that a 'reload' does not
always reload the DUNDi configuation.
How can I reload DUNDi without stopping/starting Asterisk?
-Original Message-
From: Douglas Garstang
Sent: Wednesday, June 14, 2006 11:00 AM
To: Asterisk Users Mailing
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Wednesday, June 14, 2006 12:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] DUNDi Users
If you do a reload pbx_dundi.so, it'll reload the dundi
configuration.
All,
Is there a way I can perform a lookup to see if a given extension exists within
a given context, on the local system? I could call Dial() and check the result
of $DIALRESULT, but I'm thinking there should be a better way.
Note, that I don't want to use ChanIsAvail(). That's only for
Worked it out...
ChanIsAvail(Local/[EMAIL PROTECTED])
-Original Message-
From: Douglas Garstang
Sent: Wednesday, June 14, 2006 2:38 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Determining if extension exists
All,
Is there a way I can perform
This is driving me nuts.
Why doesn't the DUNDILOOKUP function return the weight of a path to a number?
The CLI 'dundi lookup' command does. What about the mac address and expiry
period? The CLI command returns those, but the DUNDILOOKUP function does not.
Why?
We absolutely need this in order
dialplan function... I'm working on some other projects right now, but
I'm sure the Digium folks would welcome a patch from you if you really want it.
On 6/14/06, Douglas Garstang [EMAIL PROTECTED] wrote:
This is driving me nuts
Unless it's changed recently, Asterik doesn't support the SIP 'MESSAGE' command.
Doug.
-Original Message-
From: Attilla De Groot [mailto:[EMAIL PROTECTED]
Sent: Tuesday, June 13, 2006 2:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users]
function
Hi Doug,
I didn't knew this.
Thank you.
Regards,
Attilla
On Jun 13, 2006, at 4:52 PM, Douglas Garstang wrote:
Unless it's changed recently, Asterik doesn't support the SIP
'MESSAGE' command.
Doug.
-Original Message-
From: Attilla De Groot [mailto
-Original Message-
From: Moises Silva [mailto:[EMAIL PROTECTED]
Sent: Tuesday, June 13, 2006 1:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] extensions.conf
No limit in code imposed. Not sure about performance penalty for a
file
Has anyone integrated Asterisk Queues with Polycom phones?
What I'd like to do is display the agent status next to their appearance. I
don't see much discussion about this.
This is not the same thing as setting bw1/bw against the appearance in the
phone directory.
Thanks
Doug.
If you do this, and not have Asterisk in the call setup path, your going to
lose the ability to do a lot of features. What about black/white lists, rate
centers, pic codes, intra company extension dialling and other advanced
features?
Sure, you might be able to do them with SER but good luck
Does anyone know how I can get stderr from AGI to be sent to somewhere other
than the console? It seems that this is the only place it can go. Changing
logger.conf has no effect.
If you want to see errors from AGI scripts, you have to run the Asterisk
console, which isn't viable.
Doug.
Oh yeah, I also won't get time/date stamps if I redirect stderr to a file like
that
-Original Message-
From: Douglas Garstang
Sent: Monday, June 12, 2006 8:51 AM
To: 'Frederic Jean'
Subject: RE: [Asterisk-Users] AGI Stderr
Frederic,
Thanks, but that's
Two
solutions...
1. Set
OpenSER to to receive registrations from phones. OpenSER 'fans out' the
registrations to multiple Asterisk boxes with the send() command. This will
break things like call transfer however unless you can guarantee that a
transferred call goes back to the same
Somewhat off topic.
We upgraded a Polycom phone from SIP v1.6.3 to v1.6.6
The phone will no longer send SIP subscription messages for buddies to
Asterisk. I have broken the directory file down to make it as simple as
possible.
Here is what it contains.
?xml version=1.0 encoding=UTF-8
-Original Message-
From: Martin Joseph [mailto:[EMAIL PROTECTED]
Sent: Thursday, June 08, 2006 1:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] RE: IAX Passing Variables
On Jun 8, 2006, at 11:04 AM, Douglas Garstang wrote
I have
only seen Asterisk send stdout to the console, which is _extremely_ annoying. If
your running a system in production mode, and your having a problem, you have to
1)
shut Asterisk down
2)
restart the Asterisk console
3)
reproduce the problem
4)
shut asterisk down again and
5)
Well, this kinda sux.
We have three Asterisk servers. Phones register to a single,
primary server.
When a phone on one wants to reach a phone on another, we use
DUNDi to discover the destination pbx and IAX to transfer the
call to the other Asterisk box. This seems to be a fairly
common
Well, this kinda sux.
We have three Asterisk servers. Phones register to a single,
primary server.
When a phone on one wants to reach a phone on another, we use
DUNDi to discover the destination pbx and IAX to transfer the
call to the other Asterisk box. This seems to be a fairly
common
I don't know about g.729, but this will work for wav - g711.
sox file.wav file.ul
Doug.
-Original Message-
From: Matthew Crocker [mailto:[EMAIL PROTECTED]
Sent: Tuesday, June 06, 2006 12:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users]
certified to be able to purchase phones from them, but once you are certified
with Polycom you can actually download the firmware from their extranet.
On 6/5/06,
Douglas Garstang [EMAIL PROTECTED]
wrote:
Off topic. Anyone know where I can get
Polycom SIP
I thought I saw a guide at voip-info that described how to set up and asterisk
to run in a chrooted environment. Now, I can't seem to find it. Anyone know
where such a guide may be?
Doug
___
--Bandwidth and Colocation provided by Easynews.com --
If by database you are referring to an external database, such as MySQL, you
have to address failover, redundancy and performance issues if you go in that
direction.
-Original Message-
From: Moises Silva [mailto:[EMAIL PROTECTED]
Sent: Monday, June 05, 2006 10:24 AM
To: Asterisk
-Original Message-
From: Michiel van Baak [mailto:[EMAIL PROTECTED]
Sent: Monday, June 05, 2006 8:03 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Config Revision Control
On 09:41, Mon 05 Jun 06, Andrew Kohlsmith wrote:
On Saturday 03 June 2006 02:47,
On Mon, 2006-06-05 at 10:44 -0600, Douglas Garstang wrote:
I thought I saw a guide at voip-info that described how to
set up and asterisk to run in a chrooted environment. Now, I
can't seem to find it. Anyone know where such a guide may be?
http://www.voip-info.org/wiki-Asterisk+non-root
-Original Message-
From: Michiel van Baak [mailto:[EMAIL PROTECTED]
Sent: Monday, June 05, 2006 8:03 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Config Revision Control
On 09:41, Mon 05 Jun 06, Andrew Kohlsmith wrote:
On Saturday 03 June 2006 02:47,
Off topic. Anyone know where I can get Polycom SIP software v1.6.6,
unofficially?
Not that Polycom is analy retentive, or anything like that...
Doug
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE
Well, this kinda sux.
We have three Asterisk servers. Phones register to a single, primary server.
When a phone on one wants to reach a phone on another, we use DUNDi to discover
the destination pbx and IAX to transfer the call to the other Asterisk box.
This seems to be a fairly common
Oh
sweet.
-Original Message-From: Rob McKrill
[mailto:[EMAIL PROTECTED]Sent: Friday, June 02, 2006 11:25
AMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: Re: [Asterisk-Users] Polycom-Asterisk
hints/presence
According to the release notes for
Has
anyone got any neat solutions for Asterisk .conf file revision
control?
We
have multiple Asterisk boxes here, that we'd like to maintain a _mostly_ common
set of conf files on. They aren't all the same though. There's subtle
differences. For example,in sip.conf, iax.conf etc, the
, but it works great here.
On 6/2/06, Douglas
Garstang [EMAIL PROTECTED]
wrote:
Has anyone got any neat
solutions for Asterisk .conf file revision control?
We have multiple Asterisk
boxes here, that we'd like to maintain a _mostly_ common set of conf files
Title: Message
Brad,
Not
sure if #include statments will help. For that to work, there would have to be a
separate directory structure for each server. I'd like to keep it as common as
possible.
If we
had, on our first pbx server...
[general]context=frompstn_startallowguest=yes
here.
On 6/2/06, Douglas
Garstang [EMAIL PROTECTED]
wrote:
Has anyone got any neat
solutions for Asterisk .conf file revision control?
We have multiple Asterisk
boxes here, that we'd like to maintain a _mostly_ common set of conf files
on. They aren't
Title: Message
Ok,
does anyone know if anyone has already created a guide for using subversion with
Asterisk?
I've
hit a wall already, where the subversion docs say that your files _must_ go into
a directory called trunk(huh? What's with that?). That's going to break
Asterisk, who
be downloaded into /etc/asterisk by running svn up inside
the directory.
Might need to get your brakes checked if you keep hitting walls :)
On Fri, 2 Jun 2006, Douglas Garstang wrote:
Ok, does anyone know if anyone has already created a guide
for using subversion with Asterisk
control at
the file level, so for each file in the library there is a version history.
On 6/2/06, Douglas
Garstang [EMAIL PROTECTED]
wrote:
Bruce,
Do you run a subversion
client on every Asterisk box, and get the files directly, or do run
-Original Message-
From: Hadley Rich [mailto:[EMAIL PROTECTED]
Sent: Friday, June 02, 2006 3:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Config Revision Control
On Saturday 03 June 2006 09:37, Douglas Garstang wrote:
Aaron
stated you can check or co the trunk to any folder.
On 6/2/06, Douglas
Garstang [EMAIL PROTECTED]
wrote:
Aaron,I'm
trying to check-in (is that the right term?) the files for the first time.
There's nothing in the repository yet.Doug.
-Original Message- From: Aaron Da
in there about using trunk, it's just a
suggestion.
Ours is split out by server name inside a configs folder.
On Fri, 2 Jun 2006, Douglas Garstang wrote:
Aaron,
I'm trying to check-in (is that the right term?) the files
for the first time. There's nothing in the repository yet.
Doug
601 - 700 of 1210 matches
Mail list logo
