Ajey Gore wrote:
I figured that asterisk can do conferencing if we have zap interface. The
BRI cards I have they do not have Zaptel. How do I enable conferencing on my
server?
What do you mean when you describe cards as having or not having Zaptel?
--
Alex Balashov
Evariste Systems
Web:
Hi!
Anyone got a patch for call deflection for Zaptel/libpri drivers?
Thanks!
/hanna
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Hi Ajey,
which kind of BRI are you using?
Giorgio Incantalupo
Ajey Gore wrote:
I figured that asterisk can do conferencing if we have zap interface. The
BRI cards I have they do not have Zaptel. How do I enable conferencing on my
server?
Regards
Ajey
You can do conferencing without the zap interface. just modprobe
ztdummy. Its good for small conferences.
On Mon, 2008-04-14 at 23:39 -0500, Ajey Gore wrote:
I figured that asterisk can do conferencing if we have zap interface. The
BRI cards I have they do not have Zaptel. How do I enable
Hi,
I was able to store voicemail following the tutorial
http://www.voip-info.org/wiki/view/Asterisk+Voicemail+ODBC+storage
i would just like to inquire how can i create a web interface (will use php)
to play the voicemail stored in the database.
the field in the database is recording longblob
Please keep us updated on your progress.
I am considering putting several of these boxes in
and I would love to hear how this comes out.
Wish I had something to suggest.
Ex Vito wrote:
Hi list,
After a lot of testing + troubleshooting, I guess I'm observing
what I am now calling a
Quote
We had a master source location.with a master image
We cloned the hard drive with linux dd copy of master image
Did the dd to clone it actually work on RAID devices
Mike Trest - On Travel wrote:
-Original Message-
I'd be interested in sections like Rolling
Pete,
Have a look at:
http://bugs.digium.com/view.php?id=11196
- Original Message -
From: Pete Kay
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Tuesday, April 01, 2008 12:31 PM
Subject: [asterisk-users] Realtime MOH
Hi all,
I want to allow different
On 15/04/2008, Rudolf Ladyzhenskii [EMAIL PROTECTED] wrote:
Hi, all
I have SPA3000 (in Linksys reincarnation) and it has very annoying problem.
Sometimes, incoming PSTN call drops the moment one picks up analog
phone on FXO port.
Most of the times it works, other times phone on FXS
Hello,
I have switched on DND on a SNOM 360. When I call this phone, I get the
following output:
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/user4-0821b0e8,
SIP/user3|20|tr) in new stack
-- Called user3
-- Got SIP response 480 Do Not Disturb back from 192.168.0.34
--
14 apr 2008 kl. 16.19 skrev Al lists:
I have seen this issue on both 1.2 and 1.4, was not able to
reproduce to find a cause or bug.
I have seen this after power failure boot up.
show sip peer command shows most of peers, except one or two (in my
cases trunk) .
if i issue a sip reload
Two use-cases where autofill=no is desirable:
1) If it's important that you answer your callers in strict order (i.e.
in order to meet estimated wait time commitments etc).
2) If your queue members/agents are local channels (as local channels
are always available, so call attempts will be
15 apr 2008 kl. 12.06 skrev Stefan Guenther:
Hello,
I have switched on DND on a SNOM 360. When I call this phone, I get
the
following output:
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/user4-0821b0e8,
SIP/user3|20|tr) in new stack
-- Called user3
-- Got SIP response 480 Do
On Mon, 14 Apr 2008, Ajey Gore wrote:
I figured that asterisk can do conferencing if we have zap interface.
It can do conferencing without a zap interface too.
The
BRI cards I have they do not have Zaptel.
And?
How do I enable conferencing on my server?
Well you could start by reading
With regard to (1), yes, very good point there and certainly reason
enough to leave it alone. I had completely forgotten about a use case
like that.
With regard to (2), I'm pretty sure there's been work done in the
recent past to make chan_local more state aware so that this might not
be
We are using Asterisk and SER with Polycom 550 phones running SIP version
2.2.2.0084. The phones register to SER. If an AOR appears on more than one
phone when a call arrives for that AOR one, some or all of the Polycom phones
reboot. I can't seem to find the source of this problem. Has
Johansson Olle E schreef:
15 apr 2008 kl. 12.06 skrev Stefan Guenther:
Hello,
I have switched on DND on a SNOM 360. When I call this phone, I get
the
following output:
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/user4-0821b0e8,
SIP/user3|20|tr) in new stack
-- Called user3
--
Hi List,
I know that this question has been asked before so please forgive me. I have a
client that wants a box with 24 extensions that will have on it 6 conference
rooms. All the phones will be using uLaw.
Thanks.
Dovid___
-- Bandwidth and
Hi Faraz,
yes, you can use ztdummy but it cannot completely replace Digium cards.
It depends from your hardwareI had troubles with some kind of
serversso beware.
Giorgio.
Faraz R. Khan wrote:
You can do conferencing without the zap interface. just modprobe
ztdummy. Its good for small
I didn't see any mention of this contest on the mailing list - is anyone
using Lypp with their asterisk server for anything funky?
http://deancollinsblog.blogspot.com/2008/04/lypp37-signals-mashup-contes
t-or-why-i.html
http://blog.lypp.com/2008/04/14/37signals-voip-mashup-with-lypp/
Hey,
I just found out today that it doesn't work on Asterisk 1.4.19 (at
least for realtime queues) if you have autofill=yes in queues.conf.
However it works if you add it in queue settings for each queue, for
realtime that would be
ALTER TABLE queue_table ADD COLUMN autofill TINYINT(1) UNSIGNED
Hi,
I have a big issue during transfers (using Polycom phones, but I don't think
that's relevent) with Asterisk 1.14.19. Basically, the value contained in
${CDR(accountcode)} dissapears.
Here is the relevant code snippet:
--
exten =
But I want my polycom to attempt g729 on SIPPEER-SIPPEER calls, I only want
ulaw used when SIPPEER-ZAP is the case.
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darryl Dunkin
Sent: Monday, April 14, 2008 9:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
As far as I noticed - this issue is not 1.4.19 only. Same thing happens on
all Asterisk versions.
Set your own variable before transfer:
Exten = , Set(__MYACC=${CDR(accountcode)})
And use ${MYACC} in other (transfered) calls.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
The PSTN only allows ulaw or alaw (depending on your location). You
CANNOT send calls in any other codec over a PSTN line. Generally, if
you want to use G729 then you must buy a G729 license (with a few
exceptions).
Jeremy Mann wrote:
But I want my polycom to attempt g729 on SIPPEER-SIPPEER
So in other words, if I have G729 enabled on the phones, I must get G729
licenses to use Zap channels. Otherwise I have to use ULAW for everything?
I fail to understand why it'd be difficult to do codec negotiation on SIP-ZAP
calls, Zap sends that it only supports ulaw, if the phone doesn't
You need to clarify what you mean by:
1. pick up phone on FXO port (Where is this phone attached? are you
branching the incoming PSTN where one goes to SPA and one to a normal
phone?)
2. Phone on FXS? the FXS on the SPA itself? How does that work? do you
have call routing setup that way in
If you are talking between two g729 endpoints, the Asterisk overhead is
very small.
Jeremy Mann wrote:
So in other words, if I have G729 enabled on the phones, I must get G729
licenses to use Zap channels. Otherwise I have to use ULAW for everything?
I fail to understand why it'd be
On Tue, 15 Apr 2008, Dovid B wrote:
I know that this question has been asked before so please forgive me.
Easier to ask for forgiveness than permission?
I have a client that wants a box with 24 extensions that will have on it
6 conference rooms. All the phones will be using uLaw.
I don't
I have tried to install an Asterisk server on a CentOS EC2 image. The
install went ok. I was able to connect with X-lite to the instance and the
instance apparently played back SayDigits(123) (see below)
Connected to Asterisk 1.4.19 currently running on domU-12-31-38-00-91-42
(pid = 13114)
Asterisk is reporting the following error:
[Apr 15 16:58:32] WARNING[14759] ast_expr2.fl: ast_yyerror(): syntax error:
syntax error, unexpected ':', expecting $end; Input:
: Always
^
here is the dialplan:
exten = OUT,1,Gotoif($[$[${DB(AMPUSER/${ARG1}/recording)} :
out=([^|]+)] = Always]?r,1)
On 15/04/2008, at 10:34 PM, Tzafrir Cohen wrote:
On Tue, Apr 15, 2008 at 10:33:43AM +0800, Jeremy Malcolm wrote:
On 14/04/2008, at 5:58 PM, Tzafrir Cohen wrote:
In the Asterisk CLI, what happens when you run:
This is Asterisk 1.2:
unload chan_zap.so
load chan_zap.so
Yeah, I mentioned in
On Tuesday 15 April 2008 07:40:47 Jeremy Mann wrote:
But I want my polycom to attempt g729 on SIPPEER-SIPPEER calls, I only want
ulaw used when SIPPEER-ZAP is the case.
Set(_SIP_CODEC=ulaw)
Dial(Zap/g0/...)
--
Tilghman
___
-- Bandwidth and
That would work just spiffy if you are calling another SIP device, but
by the time the call gets to that point in the dialplan the codec of the
originating device has already been chosen and set in stone.
Tilghman Lesher wrote:
On Tuesday 15 April 2008 07:40:47 Jeremy Mann wrote:
But I want
If this is posted in the wrong place, I apologize profusely in advance.
Please advise me as to the correct place to post it!
Ifbyphone Inc. is a fully funded startup located in Skokie, IL (Chicago
suburb) that provides a platform of Software as a Service functions to
bridge the web to the
Sadly you are correct:
-- Executing [EMAIL PROTECTED]:4] Set(SIP/156-083514c0,
_SIP_CODEC=ulaw) in new stack
-- Executing [EMAIL PROTECTED]:5] NoOp(SIP/156-083514c0, 4) in new stack
-- Executing [EMAIL PROTECTED]:6] NoOp(SIP/156-083514c0, ) in new stack
-- Executing [EMAIL
Hi all,
I have a polycom 501 phone that I rebooted today. It stopped working...
Normally the screen shows New call, Forward and that is all...
Now the screen shows New call, Forward, MyStat, Buddies.
It no longer accepts incoming calls nor can I make outgoing calls.
I have reloaded factory
Randy, I think you are simplifying the issue by saying it's because of
asterisk that they are no longer required.
They also had a very important use when sending bulk international faxes
in that they provided store and forward functionality.
Eg you send a fax to japan but you would fax to a
מוישי ברעוודה wrote:
Asterisk is reporting the following error:
[Apr 15 16:58:32] WARNING[14759] ast_expr2.fl: ast_yyerror(): syntax
error: syntax error, unexpected ':', expecting $end; Input:
: Always
^
here is the dialplan:
exten = OUT,1,Gotoif($[$[${DB(AMPUSER/${ARG1}/recording)} :
15 apr 2008 kl. 13.38 skrev Ron Arts:
Johansson Olle E schreef:
15 apr 2008 kl. 12.06 skrev Stefan Guenther:
Hello,
I have switched on DND on a SNOM 360. When I call this phone, I get
the
following output:
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/user4-0821b0e8,
SIP/user3|20|tr) in
I may of removed it for testing just prior to send the email - i get the
same error when its there:
exten = IN,1,Gotoif($[$[${DB(AMPUSER/${ARG1}/recording)} : in=([^|]+)]
= Always]?r,1)
On Tue, Apr 15, 2008 at 6:07 PM, Jason Parker [EMAIL PROTECTED] wrote:
מוישי ברעוודה wrote:
Asterisk is
try rebotting the phone. at boot up. hold down 4,8,6,* - that should reset
the local config for the phone
On Tue, Apr 15, 2008 at 5:58 PM, Jerry Geis [EMAIL PROTECTED] wrote:
Hi all,
I have a polycom 501 phone that I rebooted today. It stopped working...
Normally the screen shows New call,
If you want to get a G729 call to go via Zap you must purchase a G729
license. No amount of discussion is going to change that.
Jeremy Mann wrote:
Sadly you are correct:
-- Executing [EMAIL PROTECTED]:4] Set(SIP/156-083514c0,
_SIP_CODEC=ulaw) in new stack
-- Executing [EMAIL
I guess that's my frustration, I don't want it g729, I want it ulaw, I just
wish Zap did codec negotiation from the client. It'd be a nice option instead
of automatically trying to translate if it's not ulaw. Could save some
processor overhead(obviously at the expense of bandwidth).
try rebotting the phone. at boot up. hold down 4,8,6,* - that should reset
the local config for the phone
On Tue, Apr 15, 2008 at 5:58 PM, Jerry Geis geisj at pagestation.com
http://lists.digium.com/mailman/listinfo/asterisk-users wrote:
/ Hi all,
//
// I have a polycom 501 phone that I
Two use-cases where autofill=no is desirable:
1) If it's important that you answer your callers in strict order (i.e.
in order to meet estimated wait time commitments etc).
Not always the case. Let's look at multiple queue assignment where agents
have skills (logged in) to multiple queues.
Originally posted by: mailto:
Hi,
I've been looking into the one bad thing about * which is there's no
practical solution to running a console. You know the kind where
you have rows of buttons each representing an extension. You press
the button of the extension you want to transfer the call
You mentioned that you removed sip.cfg. Do you have the file with the
way it was before you made any changes?
If so, try placing it back where it's supposed to be, then reboot the
server, then reboot the phone. (p.s. I don't have an *, I don't know
anything about your problem! this is just
Originally posted by: mailto:
Hi all
Now I'm making IVR sequance that is customised [mainmanu].
I wish to notify invaid command like a following
exten = i,1,playback('your command is ...')
exten = i,2,playback(${EXTEN}) ; Say 'i' oops! ;-(
exten = i,3,playback(' is incorrect! please
Originally posted by: mailto:
Hi all
a funny tool for your * server and portal site.
http://www.dairiten.com:81/asterisk_online/indicator.php
enjoy :-)
# please notify me if this icon design has problem.
---
Masakazu Nakano.
dairiten.com - an VoIP and Ubiquitus Portal site in Japan.
Thought I'd let everyone know I've released app_swift v1.6.1 which is
entirely based off of Will Orton's work he's placed in the public
domain.
Works great with Asterisk v1.6.0-beta7.1.
In any case, can be downloaded from my site at:
http://www.darrensessions.com
Go easy on me, this is my
Hi,
maybe someone can give me a hint to solve the following issue. I want to
limit the calls to a specific SIP-destination. Disabling callwaiting at
the phones is not an option, because it should be configured via the *
database.
My solution uses GROUP_COUNT, which works fine most of the time.
What is app_swift ?
Zoa
Darren Sessions wrote:
Thought I'd let everyone know I've released app_swift v1.6.1 which is
entirely based off of Will Orton's work he's placed in the public
domain.
Works great with Asterisk v1.6.0-beta7.1.
In any case, can be downloaded from my site at:
Ex Vito,
[comments inline]
Ex Vito wrote:
Hi list,
After a lot of testing + troubleshooting, I guess I'm observing
what I am now calling a regression with zaptel 1.4.10 (is it?)
As such I call for peer feedback, before either asking Digium
install support or filing a bug.
Asterisk builds two channels and bridges them together. If the codecs
mis-match then it must transcode, the negotiation on the Zap end is done
seperately from the SIP end, so it does not care what your handset
decided on.
If you want ulaw, use ulaw, not g729 (on any call leg). You won't be
able
you could try to set a var to the exten maybe.. and then use that var ..
since when in exten = i , well i will be the exten..
On Tue, Apr 15, 2008 at 11:52 AM, Anonymous [EMAIL PROTECTED] wrote:
Originally posted by: mailto:
Hi all
Now I'm making IVR sequance that is customised
FOP works for us, no need for X:
http://www.asternic.org
If you need to avoid using a mouse, you can use the Polycom attendant
console instead:
http://www.polycom.com/usa/en/products/voice/desktop/soundpoint_ip/sound
point_ip_attendant_console.html
-Original Message-
From: [EMAIL
Correct, but if I have two sip peers, one with G729ulaw, the other with
gsmulaw, they will negotiate before trying to send audio.
With ZAP, it tries to transcode whatever it receives into ulaw, period. No
negotiation to even tell the client to send ulaw if capable.
With no call level
Dovid B wrote:
Hi List,
I know that this question has been asked before so please forgive me. I
have a client that wants a box with 24 extensions that will have on it 6
conference rooms. All the phones will be using uLaw.
And how many total users on the system, including outside
Correct, those are two peers talking direct, one call leg (SIP-SIP).
In this case, you have two call legs which are then bridged:
SIP - Asterisk
Asterisk - Zap
You've already negotiated g729 before Asterisk notices that the call is
going out Zap (via your dialplan). At this point, you have to
Your stack trace appears to possibly be stack corruption.
Could you try either this branch:
http://svn.digium.com/view/zaptel/team/mattf/zaptel-1.4-stackcleanup/
Or with a kernel that does not have 4K stacks enabled? You can check if
your installed kernel does with the following
Or with a kernel that does not have 4K stacks enabled? You can check if
your installed kernel does with the following command.
$ cat /boot/config-`uname -r` | grep 4K
# CONFIG_4KSTACKS is not set
Opps, forgot to feedback: yes this kernel seems
to have CONFIG_4KSTACKS
An app to invoke the Cepstral text-to-speech engine.
On Tue, Apr 15, 2008 at 11:46 AM, Zoa [EMAIL PROTECTED] wrote:
What is app_swift ?
Zoa
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To
Kai-Uwe Jensen wrote:
An app to invoke the Cepstral text-to-speech engine.
On Tue, Apr 15, 2008 at 11:46 AM, Zoa [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
What is app_swift ?
Zoa
I've written an AGI wrapper for it as well, in case you don't want to
re-compile
to
exvito - I know it is a pain in the cahoonkus - but would you consider
sharing the OTHER Digium board issues you are having , the recommended
steps you were given by Digium to troubleshoot them, and the results ?
I think this real-wold experience wold be invaluable to the list.
THX in Advance
Darryl Dunkin wrote:
FOP works for us, no need for X:
http://www.asternic.org
If you need to avoid using a mouse, you can use the Polycom attendant
console instead:
http://www.polycom.com/usa/en/products/voice/desktop/soundpoint_ip/sound
point_ip_attendant_console.html
We recently
Shaun - Could you clarify your post a bit ?
1 - Is the 4 K stacks a Known Problem ?
a) If so is it known to be problem on any specific Linux distro ?
b) Should ALL installation Check for this PRIOR to doing an
Asterisk Install ?
2) The branch you mention below - are fixes from it in
On Tue, 2008-04-15 at 16:22 +0800, nhadie ramos wrote:
i would just like to inquire how can i create a web interface (will
use php) to play the voicemail stored in the database.
This really isn't the proper venue for that type of question... but
searching Google for PHP BLOB returns a large
actually I screw up a lot as i changed something for testing. here is the
correct error/dialplan:
[Apr 15 16:58:32] WARNING[14759] ast_expr2.fl: ast_yyerror(): syntax error:
syntax error, unexpected '=', expecting $end; Input:
= Always
^
here is the dialplan:
exten =
It seems the standard for Analog DID (at least around here) is wink start,
does the Rhino cards work with this or do I need to have the telco
immediately send the DTMF tones?
On Wed, Feb 13, 2008 at 12:33 PM, James Finstrom
[EMAIL PROTECTED] wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash:
The only solution that I found for this is to use Asterisk 1.4 with
devstate backport
(http://svncommunity.digium.com/svn/russell/asterisk-1.4/func_devstate-1.4/)
and use the hints and to determine if it's inuse (or any other status)
before the dialing - in order to generate a proper reply. I
Hi list,
I've been reading the archives and I know that transfers are unimplemented
in the CDR (
http://lists.digium.com/pipermail/asterisk-users/2007-June/189902.html)
In my case I'm running Asterisk 1.4.17 as an Office PBX. In this setup just
a small group of users are able to make long
What is the default username/password. In the Maestro forum's it only says
it's hardcoded, but doesn't say the actual username/password.
Best Regards,
On Tue, Apr 15, 2008 at 4:43 PM, Lee Jenkins [EMAIL PROTECTED] wrote:
Darryl Dunkin wrote:
FOP works for us, no need for X:
Hi,
i'm new in asterisk programming.
Maybe my question was posted thousand times but i found nothing using google.
I'm looking for a method to limit the total simultaneous calls
(inbound and outbound) that pass from internal phones to 2 SIP
providers.
I found the calllimit option but it works
Gang,
I know some of you like to keep up-to-date on various VoIP-ish
happenings. Here's an interesting little article about FreeSWITCH that
also mentions Asterisk:
http://digg.com/software/Freeswitch_Poised_to_Shake_Up_the_Open_Source_V
oIP_Scene
The author guesstimates that Asterisk has
In the sip peer definition,
disallow=all
allow=g729
allow=ulaw
SHOULD work. Asterisk can't transcode g729, so it should fall on ulaw
for the ZAP calls. But, when your polycoms talk with each other, as
long as all necessary REINVITEs happen, they should use the 729 codec I
believe. Remember
On Wed, Apr 16, 2008 at 12:46 AM, Vinz486 [EMAIL PROTECTED] wrote:
Hi,
i'm new in asterisk programming.
Maybe my question was posted thousand times but i found nothing using google.
I'm looking for a method to limit the total simultaneous calls
(inbound and outbound) that pass from
Vieri wrote:
How can I tell the make system in 1.4.19 that ilbc is
already on the system and that it should link to
/usr/lib/libilbc.a?
Shouldn't the configure script do that?
No; the Asterisk build system has never had support for using a
system-provided version of the iLBC library.
Guilherme Loch Waltrick Góes wrote:
What is the default username/password. In the Maestro forum's it only
says it's hardcoded, but doesn't say the actual username/password.
Guilherme,
The username is leebo and the password is 123. You can see it by going to:
Admin Users Edit Users
and
On Tue, Apr 15, 2008 at 8:37 PM, Al Baker [EMAIL PROTECTED] wrote:
exvito - I know it is a pain in the cahoonkus - but would you consider
sharing the OTHER Digium board issues you are having , the recommended
steps you were given by Digium to troubleshoot them, and the results ?
I think
Greetings,
This may have already been asked many times, but I cannot seem to find a
satisfactory and consistent answer anywhere.
I have an X100P card (from x100p.com) installed in a Dell PowerEdge 2850
or 2650 (cannot recall):
00:00.0 Host bridge: Broadcom CMIC-WS Host Bridge (GC-LE chipset)
On Tue, Apr 15, 2008 at 11:37 PM, Mojo with Horan Company, LLC
[EMAIL PROTECTED] wrote:
In the sip peer definition,
disallow=all
allow=g729
allow=ulaw
SHOULD work. Asterisk can't transcode g729, so it should fall on ulaw
for the ZAP calls. But, when your polycoms talk with each
Hi There,
We have some users using x-lite as their SIP phone... but im wondering
how to get the Calls Contacts to show as being available (Or if it
can be done at all?). Is this what Presence is?
Thanks
Simon
___
-- Bandwidth and Colocation Provided
Yup, I am using realtime queue. Do you mean the global setting in
queue.conf is useless and you have to set every thing in each queue to
activate the settings? If it is true, does it also apply to other
realtime settings?
On Tue, Apr 15, 2008 at 8:21 PM, Atis Lezdins [EMAIL PROTECTED] wrote:
Configure the extension as a softphone using the format
extension@asterisk.ip.address.
Works fine for me - and works even better for agents!
Simon wrote:
Hi There,
We have some users using x-lite as their SIP phone... but im wondering
how to get the Calls Contacts to show as being
No progress at all. Version from Debian/Lenny repository still crashes and I'm
not able to compile AGX. It gives out a long list of error messages. Some
unsatisfied dependencies...?
I Can't experiment for a while after unwanted night-time visit of fire-fighters
:-( I have to let everything dry
Thanks for the reply.. Sorry for the lame question.. Do i do that in
X-Lite or Asterisk?
On Wed, Apr 16, 2008 at 2:07 PM, Rob Hillis [EMAIL PROTECTED] wrote:
Configure the extension as a softphone using the format
extension@asterisk.ip.address.
Works fine for me - and works even better for
X-Lite. Of course, Asterisk will need a hint configured for that
extension as well...
Simon wrote:
Thanks for the reply.. Sorry for the lame question.. Do i do that in
X-Lite or Asterisk?
On Wed, Apr 16, 2008 at 2:07 PM, Rob Hillis [EMAIL PROTECTED] wrote:
Configure the extension as a
Thanks again!.. Right. I have it working now, it shows the users
statuses as online or offline and changes them when someone closes
their app. But not free/busy type changes.. Any idea why here?
Simon
On Wed, Apr 16, 2008 at 3:21 PM, Rob Hillis [EMAIL PROTECTED] wrote:
X-Lite. Of course,
How does one apply the patch file that is on the site (downloads.digium.com).
Thanks
J Werh
[EMAIL PROTECTED]
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
IIRC Asterisk doesn't support the full presence publishing spec so you
won't get the full range of possible status types, however you should at
least get free/busy. I vaguely recall having to change the presence
type from peer-to-peer to something else - that's done in the SIP
configuration
91 matches
Mail list logo
