vel versions, 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_ c
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 bin
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 Polycom
> -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 Garstan
sion.
>If the attempt to jump to another location in the dialplan is not
> successful,
> then the channel will continue at the next priority of the current
> extension.
>
>
> Am I being stupid here ?
>
> Julian
>
> Douglas Garstang wrote:
> >> -Orig
Anyone know if a way to have voicemail files stored as mp3's?
Thanks,
Doug.
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> -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 t
> -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.
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: Aste
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: [Asterisk
ter make sure any phones 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
> bo
d it made sense. 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:
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
=> custcar
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...
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 and get SIP presence to work with
it.
Doug
-Original Message-From: Forrest Beck
[
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 when
> -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?
> ...
> >
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 [mailto
> -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:
>
n-Commercial 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:
> find bugs and have issues :)
>
> Sean
>
> Douglas Garstang wrote:
> > In non-developer-speak, that means, 'not in current
> release', correct?
> >
> >
> >> -Original Message-
> >> From: Aaron Daniel [mailto:[EMAIL P
> -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 n
> -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 fi
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 installe
erisk-Users] AEL #include
>
>
> 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/
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_delet
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 virtualiza
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:
> -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:
>
&
> -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
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
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 30,
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: 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
Is there an Asterisk Application/Function/Variable that returns the current
number of callers in a given queue?
Thanks,
Doug
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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 tran
>
> [Description]
> Not available
> dtw-test-asterisk-001*CLI>
>
> Regards,
> - Brad
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Douglas
> Garstang
> Sent: Thursday, May 18, 2006 6:05 PM
> To
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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of NOT having any customer premise
> equipment other than 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
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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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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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 emph
t then runs through and
> touches/removes however many msg.txt files 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, Do
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 ser
lt;[EMAIL PROTECTED]> wrote:
> > Douglas Garstang wrote:
> > > We are using a backend MySQL database for call flow, not
> user agent
> > > registration info. Just how, exactly, is a backend
> database going to
> > > replicate registration data between Asterisk
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
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
> re
e the
> dialing information.
>
> Regards,
> - Brad
>
> -Original 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-Commer
> -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_loc
> 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
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
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 'extensi
-Users] Dialling a DUNDi Route
>
>
> Douglas Garstang wrote:
> > No... do you have an example of what that looks like? I get more
> > matches on google for 'the early history of hungarian
> cabinet making'
> > than I do for DUNDi examples.
>
>
> [dundi]
3 AM
> To: Asterisk 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.
> >
>
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:
uesday, May 09, 2006 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
0 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 practic
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 12:4
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]
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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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
>
>
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.
_
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
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
>
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.
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 wor
> -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
> -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 Ap
ailing List - Non-Commercial 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]
&
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 acc
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 Ma
m for a Charity
>
>
> Yes its 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
>
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 Wait("SIP
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 'orga
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 'SIP/29
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
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
> Channel
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 acc
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
Ha
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 vari
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 cons
Josue
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
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/3254107&SIP/3254105|") in new
stack
Apr 13 11:06:11 WARNING[9294]: app_page.c:193 page_exec: Incomplete destinati
erisk-Users] Setting Codecs 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
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 b
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 Gars
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()
Folks,
When I have a dial string like this:
Dial(SIP/3254101&SIP/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/3254101&SIP/[EMAIL PROTECTED],20,tr)
What happens in that ca
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.
Dou
> -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:
> > Jos
> -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"
- Non-Commercial Discussion
> 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.
> > _
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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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
match against 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 Messa
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
> -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 Poly
AIL PROTECTED]
Sent: Tue 3/28/2006 5:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
Subject: Re: [Asterisk-Users] Transferring calls - BUG0003710
29 mar 2006 kl. 01.03 skrev Douglas Garstang:
&g
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