server the check out the files on each server. I f I change a file on
server A I can then commit the change to the repository, on the central
server, and then do a svn update on the other 2.
On 6/2/06, Douglas
Garstang [EMAIL PROTECTED]
wrote:
Bruce,
But, if you
You mean voicemail.conf (or the voicemail audio files)? How about Asterisk
realtime for voicemail.conf?
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Friday, June 02, 2006 4:16 PM
To: Asterisk Users List
Subject: [Asterisk-Users] NFS and voicemail
Has
I
believe, that to be able to pick up a ringing line, you need to use shared
appearances, and 'seize' the line. The polycom phones support it, but Asterisk
does not yet.
-Original Message-From: Damon Estep
[mailto:[EMAIL PROTECTED]Sent: Thursday, June 01, 2006
8:21 AMTo:
-Original Message-
From: Jason Bachman [mailto:[EMAIL PROTECTED]
Sent: Thursday, June 01, 2006 7:13 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] AEL #include
I use the goto to jump across contexts with labels all the time.
-Original Message-
From: Michael Collins [mailto:[EMAIL PROTECTED]
Sent: Thursday, June 01, 2006 12:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] AEL #include
I use the goto to jump across contexts with labels all the time.
Anyone know if a way to have voicemail files stored as mp3's?
Thanks,
Doug.
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will continue at the next priority of the current
extension.
Am I being stupid here ?
Julian
Douglas Garstang wrote:
-Original Message-
From: Michael Collins [mailto:[EMAIL PROTECTED]
Sent: Thursday, June 01, 2006 12:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
-Original Message-
From: Kristian Kielhofner [mailto:[EMAIL PROTECTED]
Sent: Thursday, June 01, 2006 1:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Converting Voicemail wav to mp3
Douglas Garstang wrote:
Anyone know if a way
-Original Message-
From: Michael Collins [mailto:[EMAIL PROTECTED]
Sent: Tuesday, May 30, 2006 10:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] AEL #include
How would goto work if all the priorities where n?
...
Example
I just discovered labels in the dialplan.
Maybe someone (hint: the author) could like, do something crazy, and say,
update the unofficial docs on voip-user? There's nothing there about labels in
the pages for extensions.conf OR the Goto() command.
I'm not going to do it. I've realised that
It's
doable if you are only going to be usinga single, non redundant, Asterisk
box. If you intend to use more Asterisk boxes in a 'cluster', your about to
enter a whole world of hurt if you try and get SIP presence to work with
it.
Doug
-Original Message-From: Forrest Beck
Oh Crud. So, if I want to jump to another extension or context, I have to
specify the full context, extension and priority? I can't specify a label? It's
a bit tricky trying to jump to a specific priority in an extension when they're
all called 'n' !
Why is something so simple such a mess...
So,
I've kind of converted my dialplan from:
exten
=
custcare,1,GotoIfTime(8:00-17:00|mon-fri|*|*?acd_one_queue,custcare-open,1)exten
= custcare,2,Goto(custcare-closed,1)
exten
= custcare-open,1
exten
= custcare-open,99
exten
= custcare-closed,1
exten
=
. By exclusively using labels,
everthing is in the one extension and it isn't as easy to read at a glance.
There's also the chance that statements from one section could over-run into
another.
or...
am I missing something?
Doug
-Original Message-From: Douglas Garstang
Sent
in
the presence
group are on the same server.
On Wed, 31 May 2006, Douglas Garstang wrote:
It's doable if you are only going to be using a single, non
redundant, Asterisk box. If you intend to use more Asterisk
boxes in a 'cluster', your about to enter a whole world of
hurt if you try
Yikes! I'm glad I didn't take the plunge into AEL2. Get #include functionality,
but lose cid in the dialplan. Hmmm.
-Original Message-
From: Julian Lyndon-Smith [mailto:[EMAIL PROTECTED]
Sent: Wednesday, May 31, 2006 1:21 PM
To: asterisk-users@lists.digium.com
Subject:
Anyone know if #include works in ael yet?
extensions.ael:
#include inc/pbx/global.conf
context test_context {
};
*CLI ael reload
May 30 13:56:45 NOTICE[8516]: pbx_ael.c:1120 handle_root_token: Unknown root
token '#include'
May 30 13:56:45 WARNING[8516]: pbx.c:3758 ast_merge_contexts_and_delete:
No, only works in the old language, or in AEL2 which is
released in trunk.
On Tue, 30 May 2006, Douglas Garstang wrote:
Anyone know if #include works in ael yet?
extensions.ael:
#include inc/pbx/global.conf
context test_context {
};
*CLI ael reload
May 30 13:56:45 NOTICE[8516
Did the following:
svn checkout http://svn.digium.com/svn/asterisk-addons/trunk asterisk-addons
svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk
svn checkout http://svn.digium.com/svn/zaptel/trunk zaptel
svn checkout http://svn.digium.com/svn/libpri/trunk libpri
Compiled and
-Original Message-
From: Joshua Colp [mailto:[EMAIL PROTECTED]
Sent: Tuesday, May 30, 2006 2:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] AEL #include
...[stuff removed]
Not in the 1.2 release series, no. It only receives bug fixes
-Original Message-
From: Kevin P. Fleming [mailto:[EMAIL PROTECTED]
Sent: Tuesday, May 30, 2006 2:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] AEL #include
Douglas Garstang wrote:
In non-developer-speak, that means
: Re: [Asterisk-Users] AEL #include
Actually... it means not in the production release. the subversion
trunk is a release but it is not for the faint at heart. While
generally everything works pretty well, it is expected that you will
find bugs and have issues :)
Sean
Douglas Garstang
Discussion
Subject: Re: [Asterisk-Users] Compiling Asterisk-addons
I believe asterisk-addons won't compile with the latest
trunk. Use 1.2
branch instead if you want asterisk-addons.
-John
Douglas Garstang wrote:
Did the following:
svn checkout
http://svn.digium.com/svn/asterisk
-Original Message-
From: Kevin P. Fleming [mailto:[EMAIL PROTECTED]
Sent: Tuesday, May 30, 2006 3:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Compiling Asterisk-addons
Douglas Garstang wrote:
svn checkout
http
Michael,
Well I don't know if I am missing something or not, but we have various
loops and other things in there. So, we need to use the good old ugly goto().
How would goto work if all the priorities where n?
Doug.
-Original Message-
From: Michael Collins
I have a weird situation. A polycom phone is configured to use system pbx1 as
the primary outgoing 'proxy', followed by systems pbx3 and pbx2. All three
systems have identical sip.conf files. The phone is registered on pbx1.
I shut down the Asterisk application on pbx1. I make a call. The phone
-Original Message-
From: Sean Cook [mailto:[EMAIL PROTECTED]
Sent: Wednesday, May 24, 2006 5:36 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] What and When is the next version of
Asterisk?
Not necessarily... my understanding is that
-Original Message-
From: Kevin P. Fleming [mailto:[EMAIL PROTECTED]
Sent: Thursday, May 25, 2006 9:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Failover Problem
Douglas Garstang wrote:
I shut down the Asterisk application
I just realised that in actual fact Asterisk is displaying on the console
'Ignoring this INVITE request'.
Any ideas why it would be doing that? It doesn't say WHY...
-Original Message-
From: Douglas Garstang
Sent: Thursday, May 25, 2006 10:41 AM
To: Asterisk Users Mailing List
Yes, but it fast becomes a provisioning and management nightmare.
-Original Message-
From: Kerry Garrison [mailto:[EMAIL PROTECTED]
Sent: Thursday, May 25, 2006 12:07 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] FreePBX virtualization
DUNDi isn't really redundant. If any one server goes down, all registrations
will be lost for phones registered with that server. This will remain until the
phons re-register with another Asterisk box. Better make your registration
expiry period raly low. Most people won't accept waiting
Is there an Asterisk Application/Function/Variable that returns the current
number of callers in a given queue?
Thanks,
Doug
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-Original Message-
From: Douglas Garstang
Sent: Tuesday, May 23, 2006 12:12 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Queue Count
Is there an Asterisk Application/Function/Variable that
returns the current number of callers in a given queue
I have a weird situation. A polycom phone is configured to use system pbx1 as
the primary outgoing 'proxy', followed by systems pbx3 and pbx2. All three
systems have identical sip.conf files. The phone is registered on pbx1.
I shut down the Asterisk application on pbx1. I make a call. The phone
Do you have a 'g' option in your dial command? That will cause the dial plan to
continue executing after they hangup I think.
-Original Message-
From: Tony Mountifield [mailto:[EMAIL PROTECTED]
Sent: Monday, May 22, 2006 8:15 AM
To: asterisk-users@lists.digium.com
Subject:
Hi Ken, Jerry -
Hi, all. I want to have a button on my receptionist's
601 that, when
pressed, will forward her current call to a given extension. Is
there any
way to do that? I've tried to RTFM, but I'm coming up empty.
Uh - If the OP is trying to transfer an existing call,
the SIP hard phones, soft phones, and ATAs, along
with an IP access router with QoS.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Douglas Garstang
Sent: Wednesday, May 17, 2006 10:55 AM
To: Asterisk Users Mailing List - Non
I remember seeing something somewhere that described how I could get SIP header
information with Asterisk. It was a command or a variable. Anyone know what it
is? Thanks.
Doug.
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Asterisk-Users
,
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Thursday, May 18, 2006 6:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] SIP Header Info
I remember seeing something
Does anyone know if/how I can get Asterisk to listen for SIP/RTP/IAX/whatever
on multiple network interfaces?
Thanks,
Doug
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Just curious...
Does anyone know if any companies using Asterisk on embedded hardware (out at
the customer premisis), such as the Soekris Net4801, to provide VOIP service?
Doug.
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I've been working with Asterisk for a little while now, and have been looking
recently at my next career opportunity. It seems from searching the various job
sites that the predominant VOIP technology is not the applications-based open
source approach we took, but Cisco, with a really heavy
-Original Message-
From: Leif Madsen [mailto:[EMAIL PROTECTED]
Sent: Friday, May 12, 2006 12:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Dialling a DUNDi Route
On 5/12/06, Florian Overkamp [EMAIL PROTECTED] wrote:
Douglas
Ugh. We thought we'd fixed some problems by using regexten and DUNDi. Guess not.
We have a configuration with three Asterisk boxes. Phones register with a
single, primary asterisk box under normal conditions. For voicemail deposit,
retrieval, we trunk the calls over to our asterisk voicemail
to match up
with the number
of voicemails. Works like a charm, and you don't have to replicate
registration :) Like someone else said, think outside the box :)
On Fri, 12 May 2006, Douglas Garstang wrote:
Ugh. We thought we'd fixed some problems by using regexten
and DUNDi. Guess
I'm using DUNDi.
My lookup returns 'IAX2' for the tech, and 'dundi:[EMAIL PROTECTED]/3254101'
for the destination.
How do I dial this?
I've tried dialling it with:
Dial IAX2/dundi:[EMAIL PROTECTED]/3254101
passed from my AGI script, but the other endpoint (xxx.187.142.204) is
returning:
Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Dialling a DUNDi Route
Did you set up a dundi iax user in iax.conf?
On Thu, 11 May 2006, Douglas Garstang wrote:
I'm using DUNDi.
My lookup returns 'IAX2' for the tech, and
'dundi:[EMAIL PROTECTED]/3254101
system, not the first. This is a big problem. :(
-Original Message-
From: Florian Overkamp [mailto:[EMAIL PROTECTED]
Sent: Thursday, May 11, 2006 10:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Dialling a DUNDi Route
Douglas
I hope I have this wrong, but when I have a bunch of priority 1 NoOp's created
from regexten in sip.conf, and I do an 'extensions reload', I lose all the
priority 1 NoOps! This can't be right... this means that in a production
environment, if you make a change to your dialplan and do an
I'm also seeing that re-registrations from the phones are not recreating the
priority 1 NoOP's I have to completely restart Asterisk, and they come
back. I assume they're being repopulated from astdb. Good grief.
-Original Message-
From: Douglas Garstang
Sent: Thursday, May 11
Douglas Garstang wrote:
We're doing all of our call routing from a database accessed from
AGI. When we trunk calls from one asterisk system over to
another via
IAX to terminate the call, the dialling parameters are defined by
what's in the dial command on the second system
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Thursday, May 11, 2006 10:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] 'extensions reload' clears Regextens
As a note, if you don't create the dundi_local
Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Thursday, May 11, 2006 1:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] 'extensions reload' clears Regextens
-Original Message-
From
Cc:
Subject: Re: [Asterisk-Users] Dialling a DUNDi Route
Douglas Garstang wrote:
What am I trying to achieve? Uhm... a carrier grade, highly redundant
(ie multiple servers), VOIP solution with advanced business(not
residential
a DUNDi Route
On Thu, 2006-05-11 at 10:33 -0600, Douglas Garstang wrote:
[snip]
When you IAX trunk a call from Asterisk A to Asterisk B, you can't
pass the ring time and ring options of the original SIP call between servers.
Iirc you can pass
Has anyone gotten around the general problem where you have multiple Asterisk
servers in a cluster, any of which may take a call. If you transfer a call from
one Asterisk system to another, the second has no idea of the first call, and
the first refuses to release the call and logs:
May 5
Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Transferring calls between two Asterisk
Servers
Douglas Garstang wrote:
I know there's bugs open on this.
This is not a bug. There is no practical way to handle a SIP
client who
tries to transfer a call
8:40 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Transferring calls between two Asterisk
Servers
Douglas Garstang wrote:
I know there's bugs open on this.
This is not a bug. There is no practical way to handle a SIP
client who
tries
Can
anyone tell me if they know if it's possible to pass/copy sip subscriptions from
one Asterisk system to another? Can IAX do this? What about regexten?
Doug.
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If you run Asterisk as non root, you may have problems installing G729
licenses. The digium registration utility has certain hard coded stuff, and
doesn't behave well when things aren't installed in the standard location.
Doug.
-Original Message-
From: Luki [mailto:[EMAIL PROTECTED]
I have a tricky situation. I have a polycom phone with number 3254103. I have
configured the phone to forward to a new number, 1805999.
Here's my dialplan:
exten = 3254103,1,Dial(SIP/3254103,10,tr)
exten = 1805999,1,Dial(SIP/[EMAIL PROTECTED],40,tr)
When Asterisk dials 3254103, here's
Has anyone tried to use this?
I have:
exten = 2944000,1,Queue(some_q)
exten = 2944000,2,Set(QUEUE_PRIO=10)
exten = 2944000,3,Queue(some_q)
When the user enters the queue again, they are being put at the back of the
queue. It seems this new variable does not work.
Doug.
of 1 doesn't seem to work.
-Original Message-
From: Kevin P. Fleming [mailto:[EMAIL PROTECTED]
Sent: Wednesday, May 03, 2006 3:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Setting QUEUE_PRIO
Douglas Garstang wrote:
When the user
I remember a thread about this exact problem a few weeks ago.
You need to upgrade the phone XML config files to the ones that come with the
version of firmware you are using.
Doug.
-Original Message-
From: Kenneth Shaw [mailto:[EMAIL PROTECTED]
Sent: Monday, May 01, 2006 3:24 PM
To:
Ok, anyone used Sphinx with Asterisk? The docs are great at telling me how the
internals of the damn thing work, but now how to USE it. I can't find a single
example of how to run 'decode' in command line mode, without specifying a
billion options!
Doug.
possible, just create different contexts for each
organisation.
Bails
Michiel van Baak wrote:
On 13:40, Thu 20 Apr 06, Douglas Garstang wrote:
Does AMP also let you split up each charity so that each
only has access to manage their own content? That seems to me
to be a pretty big
You can't use round robin DNS. Round robin DNS will cause every SIP packet to
potentially go through a different static path, which will break things.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Saturday, April 22, 2006 5:27 AM
To: 'Asterisk Users
I'd like to increase the speed of the Asterisk sound files. Miss Alison talks a
bit slow.
I can use sox to increase the speed, but then the pitch changes and she starts
to sound like a chipmunk. Any audio experts out there know how I can increase
the speed a little bit, and change the pitch
Discussion
Subject: RE: [Asterisk-Users] Connecting to a cluster of SIP servers
How about using LVS?
http://www.ultramonkey.org/3/topologies/lb-overview.html
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Douglas Garstang
Sent: lunes
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Monday, April 24, 2006 10:06 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Connecting to a cluster of SIP servers
On Mon, 24 Apr 2006, Douglas Garstang wrote
-Original Message-
From: Jon-o Addleman [mailto:[EMAIL PROTECTED]
Sent: Monday, April 24, 2006 10:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Faster Sound Files
On Mon, Apr 24, 2006 at 09:41:40AM -0600, Douglas Garstang
spake
I think a 'sip reload' will keep your sip subscriptions.
-Original Message-
From: Colin Anderson [mailto:[EMAIL PROTECTED]
Sent: Monday, April 24, 2006 1:24 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] HINTS with Polycom stops working
Does AMP also let you split up each charity so that each only has access to
manage their own content? That seems to me to be a pretty big limitation of all
the Asterisk management software out there. It's designed to be used by one
company to manage their own config, not to be used by many
I'm having some issues with Background() and Read() commands.
See the example below. This is when I wait for Background to finish playing the
sound file, before entering '12345#'.
All works fine.
hestia*CLI
-- Executing Answer(SIP/2944093-3366, ) in new stack
-- Executing
I'm using Asterisk 1.2.7.1. Has the callerid number matching functionality in
extensions.conf changed recently?
exten = ,1,NoOp(${CALLERID})
hestia*CLI
-- Executing NoOp(SIP/2944093-d24d, Cletus the Slaw Jawed Yokel
2944093) in new stack
== Auto fallthrough, channel
Is this with Asterisk in the RTP stream? Is it doing any transcoding?
-Original Message-
From: JR Richardson [mailto:[EMAIL PROTECTED]
Sent: Tuesday, April 18, 2006 9:34 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asterisk Performance 350 Concurrent
I don't think Asterisk supports SIP MESSAGE, does it?
-Original Message-
From: João Paulo Antunes [mailto:[EMAIL PROTECTED]
Sent: Tuesday, April 18, 2006 10:26 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] eyeBeam + ASterisk 1.2.7.1 + Instant Message
Hi,
I need
to manually set certain CDR fields.
1).
Callers are allowed to call someone within the same organisation by using a 4
digit extension. A database lookup maps the 4 digit extension to the real
number. However, a CDR for this call shows the original 4 digit extension still.
What
Jeez.
Why does every startup in the universe have to be in the bay area.
:(
-Original Message-From: Dean Collins
[mailto:[EMAIL PROTECTED]Sent: Monday, April 17, 2006 1:00
PMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: [Asterisk-Users] Orative
Southern California would make me happy, maybe the north west.
:)
-Original Message-From: Dean Collins
[mailto:[EMAIL PROTECTED]Sent: Monday, April 17, 2006 1:49
PMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: RE: [Asterisk-Users]
Orative
Easy
Well... did you tell him your services where not free and come to a
financial arrangement before you started?
-Original Message-From: Voce Lavoce
[mailto:[EMAIL PROTECTED]Sent: Friday, April 14, 2006 3:14
PMTo: asterisk-users@lists.digium.comSubject:
[Asterisk-Users] My
I just upgraded to Asterisk 1.2.7 from 1.2.5.
Page() is behaving differently.
I'm getting an error - Incomplete destination '' supplied.
-- Executing Page(SIP/2944093-5999, SIP/3254107SIP/3254105|) in new
stack
Apr 13 11:06:11 WARNING[9294]: app_page.c:193 page_exec: Incomplete destination
2006/4/13, Kevin P. Fleming [EMAIL PROTECTED]:
Douglas
Garstang wrote: I just upgraded to Asterisk 1.2.7 from
1.2.5. Page() is behaving differently. I'm getting an error
- Incomplete destination '' supplied.This was a bug introduced in
1.2.7. I have just fixed it in Subversion
Does anyone know if it's possible to set the codecs for a number via an
Asterisk command?
Ie, yes you can set the codecs in sip.conf for a user, but I'd like to have a
command that can set the same thing so that it can be done without having to
change sip.conf.
Essentially I want the user to
on the Fly
Douglas Garstang wrote:
Does anyone know if it's possible to set the codecs for a
number via an Asterisk command?
Ie, yes you can set the codecs in sip.conf for a user, but
I'd like to have a command that can set the same thing so
that it can be done without having
I'm
wondering if the page application is broken in 1.2.5
The
following:
exten
= 2001,1,Page(SIP/3254105)
does
strange stuff. The caller's phone immediately drops into the call, while the
callee's phone is still ringing. I'd think it was a SIP messaging issue, except
that the Dial()
Point taken.
-Original Message-
From: Kevin P. Fleming [mailto:[EMAIL PROTECTED]
Sent: Monday, April 10, 2006 10:17 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] App Page() in 1.2.5
Douglas Garstang wrote:
exten = 2001,1,Page
Folks,
When I have a dial string like this:
Dial(SIP/3254101SIP/3254102,20,tr)
and I want to check the ${DIALSTATUS} variable after the dial, how do I know
which number I am getting the variable for?
And, what about this?
Dial(SIP/3254101SIP/[EMAIL PROTECTED],20,tr)
What happens in that
Wow. If Asterisk could return SIP response codes that would be AWESOME.
-Original Message-
From: Joshua Colp [mailto:[EMAIL PROTECTED]
Sent: Monday, April 03, 2006 10:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SIP Responsecodes
The 'sip show channels' and 'show channels' command aren't exactly easy to
interpret, especially if one of the numbers has pic codes and rate centers
inserted (the rest is truncated on the output), or you have a proxy involved in
the call. Wish someone with some C knowledge would fix that.
Ok, this is highly confusing.
hestia*CLI sip show users
Username Secret Accountcode Def.Context
ACL NAT
2944030 2944030 oneeighty_start
No RFC3581
2944035 2944035
Is there any way to get Asterisk to reload the /var/lib/asterisk/astdb file?
It seems to only read it on startup.
Thanks.
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Subject: Re: [Asterisk-Users] Reload astdb?
Douglas Garstang wrote:
Is there any way to get Asterisk to reload the
/var/lib/asterisk/astdb file?
It seems to only read it on startup.
Thanks.
___
--Bandwidth and Colocation provided
-Original Message-
From: Kevin P. Fleming [mailto:[EMAIL PROTECTED]
Sent: Thursday, March 30, 2006 1:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Reload astdb?
Joshua Colp wrote:
It depends what you mean by reload. The file
-Original Message-
From: Joshua Colp [mailto:[EMAIL PROTECTED]
Sent: Thursday, March 30, 2006 1:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Reload astdb?
Douglas Garstang wrote:
Joshua,
I'd like issue a command
-Original Message-
From: mustardman29 [mailto:[EMAIL PROTECTED]
Sent: Wednesday, March 29, 2006 4:08 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Receptionist Phones
Could you please explain this limitation. Why would Polycom
I've been going in circles for a few weeks now with Realtime SIP.
My extconfig.conf has:
sipusers = mysql,dbname,ast_sip_users
sippeers = mysql,dbname,ast_sip_users
When I do a 'sip show peers' I see all my phones. When I do a 'sip show users'
I only see a few of them. I can't work out why
a type=peer entry anyway? I thought a peer was for
outgoing calls only???
Is this the way it's going to work in some future release of Asterisk?
Btw, I tried setting the phones to peer because I don't know what the frig I'm
doing.
Doug
-Original Message-
From: Douglas
I made a call from 3254102 to 2944093. I then tried to do a transfer to
3254107.
IP addresses have been changed to protect the innocent.
It appears this related to bug 3710. It's unclear from the bug if the problem
has been fixed or not. If it hasn't, then this seems pretty serious and would I
Do a trace with ngrep or ethereal and see if the phone is sending back a sip
DECLINE or similar. If it is, I'd say you have an Asterisk bug.
Doug.
-Original Message-
From: Stephen Kratzer [mailto:[EMAIL PROTECTED]
Sent: Tuesday, March 28, 2006 7:51 AM
To:
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