On Sun, 2005-11-20 at 20:52 -0600, Michael Graves wrote:
Would anyone on-list be able to provide me with some sample config
files for the Aastra 480i SIP phone. I'd like to to migrate from
individually hand tweaked to centrally FTP provisioned, but need
somewhere to start. I'm also looking to
Yes, you can use the Eicon Diva Range with 2.6 Kernels
See this page to see how to get an Eicon Diva working with Asterisk.
http://www.voip-info.org/wiki-Asterisk+Eicon+Diva+CAPI+ISDN
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Vlasis
Hatzistavrou -
Hello
I've just installed Asterisk 1.2 onto a FreeBSD system and it is mostly
working well.
But it dies at 2am every morning. Not quite a complete death, but it
seems to loose any ability to communicate with the rest of the world. In
/var/log/messages I just see endless entries like this:
Matt Riddell wrote:
Marc Storck wrote:
Hello,
I would like to know if there is a way in IAX2 and SIP to tell a client
to register at a different server.
For example:
Client tries to register at server B but server B answers with some sort
of redirect to tell the client to register at server
John Biundo [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
I posted the following a couple of days ago. My problem was inbound, but
the workaround might be worth a try:
==
Bug or feature?
Thanks John,
It appears that my outbound supplier has a different default DTMF setting
BJ Weschke wrote:
On 11/19/05, Jean-Denis Girard [EMAIL PROTECTED] wrote:
Hi all,
Is sending text to a conference supported by asterisk-1.2, ie one member
of the conference sends text, it is received by all other members of the
conference (provided their channel supports text of course) ?
I
Matt Riddell [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
Chris Cahill wrote:
The process then goes on to call a few agi scripts, and ends up
creating another file (via agi) in the outgoing directory, this one
being the one that calls the outside world.
Are you *creating* the file
John Martin wrote:
Can anyone recommend a version of mISDN and mISDNuser (dates of CVS or
archive held on someone’s server) that will work with the chan_isdn in
Asterisk 1.2.
I have used the install-misdn script on http://www.beronet.com/download/
and that seems to work..
Hi,
I'm compiling Zaptel1-1.0.9 in Sparc64/Debian and I'm getting these
errors. I compiled asterisk on the same machine and it went ok. I
want to activate the conference feature of asterisk thats why i'm
compiling zaptel. These are the errors:
sip:/usr/local/src/zaptel-1.0.9.1# make
gcc
In article [EMAIL PROTECTED],
Chris Hastie [EMAIL PROTECTED] wrote:
Hello
I've just installed Asterisk 1.2 onto a FreeBSD system and it is mostly
working well.
But it dies at 2am every morning. Not quite a complete death, but it
seems to loose any ability to communicate with the rest of
Hi all!
Someone who can recommend a good E1 gateway for
terminating VoIP traffic. H323 or Sip
Regards
Anders Svensson
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Well that didn't work. When I rebooted MySQL didn't start at allOn 11/21/05, JP Carballo [EMAIL PROTECTED]
wrote:JP Carballo wrote: Eric Bishop wrote: I have:
[EMAIL PROTECTED] ~]# chkconfig --list | grep mysql
mysqld0:off
1:off 2:off
3:on4:off 5:off 6:off [EMAIL PROTECTED] ~]# chkconfig
Title: Message
Hi
I'm looking for
following solution:
Asterisk is
connected to PSTN by Digium or some another card which has Fax
Detection
If incoming call is
a fax I woud like to transfer it to External Fax server by SIP or H323 for
getting a Fax.
If incoming call is
a voice to direct
Anders Svensson wrote:
Someone who can recommend a good E1 gateway for terminating VoIP
traffic. H323 or Sip
Asterisk!
/O
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Title: Message
Hi
I'm looking for
following solution:
Asterisk is
connected to PSTN by Digium or some another card which has Fax
Detection
If incoming call is
a fax I woud like to transfer it to External Fax server by SIP or H323 for
getting a Fax.
If incoming call is
a voice to direct
Well that didn't work. When I rebooted MySQL didn't start at all
The level doesn't set _when_ something starts, just _if_ something
starts. Some daemons should start in single user mode, some not. Some
others should only start when in GUI mode, others not, etc. This is what
level controls.
On Sat, 2005-11-19 at 13:47 -0800, Ben Higley wrote:
[AG] Pocket_PC AT+BRSF=23
[AG] Pocket_PC ERROR
[AG] Pocket_PC AT+CIND=?
[AG] Pocket_PC ERROR
[AG] Pocket_PC AT+CIND?
[AG] Pocket_PC ERROR
[AG] Pocket_PC AT+CMER=3,0,0,1
[AG] Pocket_PC ERROR
[AG] Pocket_PC
Hi all,
i've a problem on a PSTN line that i've connected to Asterisk server with a
Diginum Card. A diagram:
+--[Fax Machine]
|
[PSTN]+
|
+--[Asterisk]
The problem is if i pick up the phone on Fax Machine i get no Dialtone.
On 21 Nov 2005, at 00:38, Luki wrote:LD_ASSUME_KERNEL 2.4.1 ... will make the kernel do old-styleprocess-perthread posix threads. I don't have this anywhere in the startup script on 2.6.12-1.1372_FC3and still have only one process in ps:Sorry, I wasn't clear, if you _do_ have LD_ASSUME_KERNEL
Hello Enky,
We have encountered similar problems with various Ericsson Nokia
phones. We couldn't get the channel driver to work 100%. However, we
cannot actually tell whether it was our mistake or whether there was a
problem with the channel driver. We tried to contact the driver's
Hi all!
I'm trying to play some music from asterisk, and when I call to the PBX
from a GSM mobile phone, the more I speak while hearing the music, the
worst is the quality of the music I hear... My audio is at 8Khz,
16bits/sample.
I've tried different codecs for asterisk, but results
I dont think its a good idea to put an * in Bosnia when we are in Sweden.
Anders
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olle E.
Johansson
Sent: den 21 november 2005 10:42
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Hi,
I am trying to configure the LCS with the
Asterisk1.0.3
Is that needed to modify some code in
configuration files or it needed to modify in the source code too.
Can anyone please suggest how to configure
the LCS?
Regards,
Bidyut
Wipro Technologies
Confidentiality
[EMAIL PROTECTED] wrote:
I dont think its a good idea to put an * in Bosnia when we are in
Sweden.
Why not?
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I tried this dial command to get a sound to play to the caller on answer.
I have even tried to use the LIMIT_CONNECT_FILE option with no success.
As can be seen below the start_sound variable shows 'UNDEF'.
Are there some other settings I have missed out, eg. file location, type etc.
The
On Mon, 21 Nov 2005, Tony Mountifield [EMAIL PROTECTED] wrote:
In article [EMAIL PROTECTED],
Chris Hastie [EMAIL PROTECTED] wrote:
Hello
I've just installed Asterisk 1.2 onto a FreeBSD system and it is mostly
working well.
But it dies at 2am every morning. Not quite a complete death, but it
On Mon, 21 Nov 2005, Tony Mountifield [EMAIL PROTECTED] wrote:
In article [EMAIL PROTECTED],
Chris Hastie [EMAIL PROTECTED] wrote:
Hello
I've just installed Asterisk 1.2 onto a FreeBSD system and it is mostly
working well.
But it dies at 2am every morning. Not quite a complete death, but it
-Original Message-
From: Senad Jordanovic [mailto:[EMAIL PROTECTED]
Sent: Monday, November 21, 2005 5:42 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] E1 Gateway
[EMAIL PROTECTED] wrote:
I dont think its a good idea to put an * in
I was able to register Portrait with our Asterisk box, but no audio, no
signaling at all.
Played a while with different codecs but no success.
Did anybody make it really work with asterisk?
Any hints, configs etc.
Regards
Guido Hecken
I've also had some luck with Microsoft Portrait
Guido
Im looking for SIP phone
*** REPLY SEPARATOR ***
On 11/20/2005 at 8:01 AM Time Bandit wrote:
Hi there,
is there any free softphone that i can customize accoring to my needs ??
You could use IaxComm : http://iaxclient.sourceforge.net/iaxcomm/index.html
hth
On Saturday 19 November 2005 19:12, dolcicbe wrote:
Hello
can you explain me that more exactly.
Thank you
Bernhard
I think what is meant is that SuSe already come with zaptel modules in
/lib/modules/`uname -r`/extra
The zaptel build process puts the its modules in
Hello,
I try to compile zaptel .
I installed kernel-sources but when i run :
make linux26
/
serveur1:/usr/local/src/ASTERISK/zaptel-1.2.0# make
linux26
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE
-DSTANDALONE_ZAPATA
harry gaillac wrote:
Hello,
I try to compile zaptel .
I installed kernel-sources but when i run :
make linux26
/
serveur1:/usr/local/src/ASTERISK/zaptel-1.2.0# make
linux26
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE
Once upon a time Sunday 20 November 2005 10:38 pm, JP Carballo wrote:
JP Carballo wrote:
Eric Bishop wrote:
I have:
[EMAIL PROTECTED] ~]# chkconfig --list | grep mysql
mysqld 0:off 1:off 2:off 3:on4:off 5:off 6:off
[EMAIL PROTECTED] ~]# chkconfig --list | grep
Once upon a time Sunday 20 November 2005 8:39 pm, Matt Riddell wrote:
Eric Bishop wrote:
Hi All,
I am running Asterisk (1.0.9.) on CentOS 4 with CDR recording being
output to MySQL. However whenever the system boots up after a reboot I
am needing to manually restart Asterisk because
Hi! I'm new to asterisk and I'm trying to develope an application that
allows the caller to input an Id and then the system redirects to an
operator.
What I did so far is to create an agi script that receives the call,
allow the input, query a DB to check thah the Id is valid and then
transfer.
I
Hi all,
i've a problem in my Asterisk system. We have
around 30 SIP phones connected to an asterisk system, and sometimes some SIP
channel (associated to an extension) gets busy all the time, even whenthat
extensionisn't in use.
We have a workaround for this, as we can't restart
asterisk
Salut Harry, plus de nouvelles de toi :(
Serais tu faché?
Olivier
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de harry gaillac
Envoyé : lundi 21 novembre 2005 13:34
À : asterisk-users@lists.digium.com
Objet : [Asterisk-Users] Can not build zaptel with
Hello David,
I rewrote the Makefile so I can compile the modules .
However I got the same problems with kernel 2.4.I
fixed some variables which was not found .
Is it a problem with my debian installation
!!???
Regards
Harry
PS: I like to set ! for Mr Pascal :-)
--- David Uzzell
Andrew Kohlsmith [EMAIL PROTECTED] wrote:
It doesn't really matter whether you buy it (my explanation) or not -- if your
specific echo is greater than what the software and/or hardware are designed
to handle, it will work poorly. It's called a misapplication of the
technology.
Two products
Folks,
I have several SIP providers that work fine. But I just added a
Broadvoice account and all I seem to get is Your call cannot be
completed at this time.
Broadvoice is registered and receives incoming calls.
Dial plan (identical to other external SIP providers) is passing call
to
In article [EMAIL PROTECTED],
Chris Hastie [EMAIL PROTECTED] wrote:
On Mon, 21 Nov 2005, Tony Mountifield [EMAIL PROTECTED] wrote:
Next, examine the cron jobs that happen at 2am, to see if any of them
could explain anything.
I did look at that. Nothing seems to run at 2am that doesn't
This are the facts:
after a couple of days running, everything appears to run very well..
asterisk is alive, no bad lines in log..
But actuallu th oh323 channel disappears
if tou type at the console oh323 TAB no helps is given
oh323: no such command !!!
help: nothiong about oh323 !!!
Hello Everyone,
I have finished up work on what will (hopefully) become AstLinux 0.3.0.
AstLinux 0.2.9 has been released as a test release, and includes the
following changes:
- Asterisk 1.2.0
- Zaptel 1.2.0
- libpri 1.2.0
- Sangoma wanrouter beta1-2.3.4
- Linux kernel 2.6.13.3
- improved
I've got an HT486 on my network which was configured to send DTMF over
RTP. _Was_ being the operative word because post my recent upgrade to
1.2, RFC2833 for DTMF just stopped working!
Works fine on my GXP2000, but no longer on the HT486. Got it going again
by configuring sip info mode.
Hi,
I'm trying to get the 'xfersound' working with v1.2. I enabled all in
features.conf (like: xfersound = beep), but I can't get the beep when
transferring a call.
I'm trying this with * v1.2, the bristuff-version, but I'm not sure if
that matters? (does it only work with SIP-to-SIP
Hi All
I have configured asterisk with the addons and setup my config files so that i
can pull sip extensions (phones) from a mysql database.
I have followed all the docs and have editted my extconfig.conf res_mysql.conf
and sip.conf to contain all that is advised.
From the CLI i can see
Jesus Bermudez Riquelme - Pcmur Soluciones Informaticas wrote:
Hi all,
i've a problem in my Asterisk system. We have around 30 SIP phones
connected to an asterisk system, and sometimes some SIP channel
(associated to an extension) gets busy all the time, even when that
extension isn't in use.
Hi Kristian,
Excellent thanks..
On 11/21/05, Kristian Kielhofner [EMAIL PROTECTED] wrote:
Hello Everyone,
I have finished up work on what will (hopefully) become AstLinux
0.3.0.
AstLinux 0.2.9 has been released as a test release, and includes the
following changes:
-
Hi,
I'm experiencing some problems with my Asterisk 1.0.9. When a customer
tries to use transfer method sometimes Asterisk crashes. The following
message appears in /var/log/asterisk/messages
Nov 17 15:56:35 WARNING[759]: No path to translate from
SIP/12.34.56.78-3aef(1) to
Hi all,
for h323 to h323 can asterixk *proxy **call* with *signal **only* ( no rtp
go through asterisk )
thanks in advance
best regards!
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Aastra Telecom has released SIP v1.3 firmware for the Aastra/Sayson range of IP
phones.
This is a major update compared to firmware 1.2.x with many bugfixes and
Asterisk(tm) interop limitations fixed.
The firmware, updated manuals and release notes are available for download at:
Does anyone know is the zyxel p2000w has call waiting? I
hear noise when a second call comes in but cannot find any documentation.
Thanks,
Chip
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Hi Are and Matt -
We have a user (our CEO) who has phones in two different offices, and
we'd like him to be able to get all his VM in either office,
regardless of which office was originally called.
My idea instead is to use externnotify to run some kind of script to
forward the vm to
Yes. The biggest challenge is putting together a mux device that
mixes the text frames out to all of the user/channel threads in the
conference.
On 11/21/05, Olle E. Johansson [EMAIL PROTECTED] wrote:
BJ Weschke wrote:
On 11/19/05, Jean-Denis Girard [EMAIL PROTECTED] wrote:
Hi all,
Is
On Monday 21 November 2005 07:53, Doug Meredith wrote:
Two products are both intended to eliminate echo, and product A, due
to it's design, can't eliminate some of the echos that product B can.
It seems quite fair to say that B is a better product than A.
It depends on your specific needs.
If
Hello Olivier,
Non je ne suis pas fâché !
Alors ce *b2bua ?
En fait je cherche une solution pour intègrer
SER+Asterisk sur la même machine.
Ser est un bon proxy asterisk un bon ipbx.
Je souhaite utilisé ser pour le routage sip avec
asterisk et pour fournir les service de téléponie
d'entreprise
Did you see the mysql.log file?
I was having a similar problem, and i saw a problem with an update in a
mysql table when a user was trying to register a phone.
Sixto
- Original Message -
From: scott [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, November 21, 2005
Pavel Siderov wrote:
Hi,
I'm experiencing some problems with my Asterisk 1.0.9. When a customer
tries to use transfer method sometimes Asterisk crashes. The following
message appears in /var/log/asterisk/messages
Nov 17 15:56:35 WARNING[759]: No path to translate from
asterisk1*CLI sip show users
UsernameSecret Accountcode Def.Context ACL NAT
205 testfrom-internal No No
204 testfrom-internal No No
203 testfrom-internal No No
202 020 from-internal No No
201
I'd like to begin messing around with realtime and mysql, but have never
done anything with either before. Can anyone point me to any form of
document that would help me understand the installation/config process?
Been around * for a couple of years and linux for more then ten years,
just never
Andrew Kohlsmith [EMAIL PROTECTED] wrote:
I took exception to your painting the
Digium hardware echo can module and the software echo cans in zaptel as
trash, as they work very well for many people. They clearly aren't
sufficient for your specific needs, and thus the Orion Telecom echo
Hi,
I'm using an ATL IP400 phone and cant get it to register, it fails with:
chan_sip.c:9405 handle_request_register: Registration from 'xx sip:[EMAIL
PROTECTED]'
failed for 'x.x.x.x'
looking at the register request i notice two things:
Authorization: Digest
yes
On 11/21/05, Javier Oviedo [EMAIL PROTECTED] wrote:
Hi all,
for h323 to h323 can asterixk *proxy **call* with *signal **only* ( no rtp
go through asterisk )
thanks in advance
best regards!
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On Monday 21 November 2005 09:23, Doug Meredith wrote:
That wasn't me.
hahaha you're quite right. I wasn't paying attention to who replied. My
apologies.
I feel that my points still apply, though.
-A.
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Hello,
I checked out the asterisk version from the CVS. But I dont seem to have
the addmailbox script.
How can I setup a mailbox without this utility.
Regards,
Rajesh Golani
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Can someone tell me what problem I am having with Zaptel on
a Suse 10 distribution?
cc -I. -O4 -g -Wall
-DBUILDING_TONEZONE -DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o gendigits.o gendigits.c
cc -o gendigits gendigits.o -lm
./gendigits
cc -I. -O4 -g -Wall
Try out http://astbill.com
AstBill is an Open Source Web Based Billing, Routing and Management
Software for Asterisk and MYSQL.
It is using 100% Realtime and there is active support in the forum.
http://astbill.com/forum-- Are Casillahttp://astartelecom.com - Independent VOIP Telecoms Broker.
You have error: Username/auth name mismatch
So there is clearly and issue with the content in your table.
In our setup the column name and username have the same value = 114
the fromuser and authuser column = NULL
If this is not helping send your table definition and the content of your record
How do you install AMP? I downloaded it and tried to run
make or install and it doesnt work. Is there some trick to this?
Thank.s
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Friends in the Asterisk community,
There have been a lot of questions about Asterisk version numbers on the
mailing lists. Here's a clarification:
* Executive summary
---
- Asterisk 1.2 = RELEASE version (previously called stable)
Asterisk 1.2.0 = First release of 1.2
Hi
Thank you for your reply.
I have tried various definitions in the sipusers table but none seem to be
working :-(
I have attached mey structure and content export below for your attention.
Many thanks
Scott Pinhorne
--
-- Table structure for table `sip_users`
--
CREATE TABLE `sip_users`
It looks like you do not have the kernel source code installed. Go to
'Yast' and 'Install Software'. Look for the package called
'kernel-source'. It will install the source for your kernel. Then run
the 'Update Software' to make sure the kernel and the kernel source are
the same version. Then
Hiya,
anyone have an idea what I need to do to fix this, I have a TDM400P and
asterisk 1.2, when I make a call to the system asterisk see the phone
ringing and looks like it picks it up from the console, but the phone
actually just continues to ring.
I am thinking I have something silly in the
Angelito Manansala wrote:
yes
On 11/21/05, Javier Oviedo [EMAIL PROTECTED] wrote:
Hi all,
for h323 to h323 can asterixk *proxy **call* with *signal **only* ( no rtp
go through asterisk )
thanks in advance
best regards!
___
--Bandwidth and
Hello,
Several of us were told that there would be a 1.0.10 release as the
final release of Asterisk 1.0 tree. There are several serious bugs in
the 1.0 tree that have been fixed in v1-0 cvs and it would be nice to
have this packaged as a release before the tree stops being accessible
on the CVS
On Monday 21 November 2005 17:12, Goran Donev wrote:
How do you install AMP? I downloaded it and tried to run make or install
and it doesn't work. Is there some trick to this?
The trick is to run the install script and read the documentation. Just not
in that order...
--
Cheers
Wayne
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Téléchargez cette version sur http://fr.messenger.yahoo.com---BeginMessage---
Hello,
Here is my config :
Angelito Manansala wrote:
yes
On 11/21/05, Javier Oviedo [EMAIL PROTECTED] wrote:
Hi all,
for h323 to h323 can asterixk *proxy **call* with *signal **only* ( no rtp
go through asterisk )
thanks in advance
best regards!
Hello,
As far as I know Asterisk cannot disentangle RTP from
You can download a new SIP firmware and force the Cisco IP phone to use it.
Some interesting links about it:
http://www.aarnet.edu.au/events/conferences/2004/apan-questnet/sipworkshop/uas/cisco7960/cisco7960.html
http://www.voip-info.org/wiki/index.php?page=Asterisk+phone+cisco+79xx
Joao
On Mon, Nov 21, 2005 at 08:18:09PM +0530, Rajesh Golani wrote:
Hello,
I checked out the asterisk version from the CVS. But I dont seem to have
the addmailbox script.
Because it is no longer needed
How can I setup a mailbox without this utility.
app_voicemail does that for you. No need to
VoicePulse
IAX.cc
BroadVoice
Teliax
Kerry Garrison
Publisher - GeekGazette.com - VOIPSpek.net
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com
Hi,
Can anyone recommend a good reputable VoIP gateway service provider that I
can use with my Asterisk server in wa.us? All I
Hi,
It's not possible to provide log due to the reason that system is in
production and there are many current calls. Crash happens on 1-2 weeks
once. I cannot simulate and get the same result with x-lite, cisco ata
and sipura 3000 when trying transfer. But some of the customers some
way
this would be very beneficial to me as well.. I have the S518 ADSL card in
my Linux system as well..
I was looking at going to an ASTLINUX solution.
Hi Kristian,
Excellent thanks..
On 11/21/05, Kristian Kielhofner [EMAIL PROTECTED] wrote:
Hello Everyone,
I have finished up
Matt Florell wrote:
Hello,
Several of us were told that there would be a 1.0.10 release as the
final release of Asterisk 1.0 tree. There are several serious bugs in
the 1.0 tree that have been fixed in v1-0 cvs and it would be nice to
have this packaged as a release before the tree stops
Hello.
I have sucessfully installed chan_bluetooth with my asterisk system.
However I wasn't able to get to that until I completed a few other steps..
1) using the sdptool - start up the services that the Audiovox is looking
to pair with
'sdptool hs'
'sdptool hf'
this allowed me to start the
Pavel Siderov wrote:
Hi,
It's not possible to provide log due to the reason that system is in
production and there are many current calls. Crash happens on 1-2 weeks
once. I cannot simulate and get the same result with x-lite, cisco ata
and sipura 3000 when trying transfer. But some of the
I'll try that tonight...
On Sat, 2005-11-19 at 13:47 -0800, Ben Higley wrote:
[AG] Pocket_PC AT+BRSF=23
[AG] Pocket_PC ERROR
[AG] Pocket_PC AT+CIND=?
[AG] Pocket_PC ERROR
[AG] Pocket_PC AT+CIND?
[AG] Pocket_PC ERROR
[AG] Pocket_PC AT+CMER=3,0,0,1
[AG] Pocket_PC
Based on a discussion on the IRC a long time ago (several days) I've
created a patch for 1.2 in the bug tracker that allows you to see if a
parking lot is occupied or not - provided you use the Flash panel or SIP
subscriptions.
What you do:
* Patch the 1.2 source with the patch in
I'm runing [EMAIL PROTECTED] beta6 and I have a problem with registration of SIP phone.
I can't find/replicate when exactly its happends but sometimes after server restart or phone restart one of the phone can't register and I get this in the server:
Transmitting (no NAT) to
Just picked up two of these puppies from my parcelforce depot.
Man, they are smart phones. They look the business. I installed one
within seconds, fantastic web configuration - much like the SPA3000 box.
Speakerphone sounds good, handset feels and sounds good.
I'll be using this heavily over
Am I correct in assuming that if I am not running Realtime on my
asterisk 1.2 server, the proper way to disable it is to remove the
following 2 files:
/usr/lib/asterisk/modules/pbx_realtime.so
/usr/lib/asterisk/modules/app_realtime.so
I am just testing out the default installation and am getting
Does it hold state information for any channel? Even ZAP, IAX,
etc!!!
If it does, Olle, you have just placed us one step closer to being able
to emulate a Key system!!!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Olle E. Johansson
AFAIK there were some known issues preventing call transfer from H323
terminals, at least with Innovaphone ones.
Yours
l.
On Fri, 18 Nov 2005 17:45:39 +0100, Santosh Rao
[EMAIL PROTECTED] wrote:
Hello list,
We have asterisk v1.2.0 CVS head and ooh323 in place. calls
can be
We have a review of it at http://voipspeak.net, I personally really like it.
Kerry Garrison
Publisher - GeekGazette.com - VOIPSpek.net
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
It is a better practice to use a noload option in
modules.conf. That way if and when you upgrade you wont need to remove them
again they will just continue to not load
Alex
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
PedroSent: Monday, November 21, 2005
Well, this is interesting - is anybody actually using app_icd out there?
:-)
l.
On Thu, 17 Nov 2005 00:54:56 +0100, Tyler [EMAIL PROTECTED] wrote:
Anyone using app_icd? I need to use some of the advanced features that
the regular asterisk Queue() application won't provide. Anyone have
On 11/21/05, Alexander Lopez [EMAIL PROTECTED] wrote:
Does it hold state information for any channel? Even ZAP, IAX,
etc!!!
If it does, Olle, you have just placed us one step closer to being able
to emulate a Key system!!!
-Original Message-
From: [EMAIL PROTECTED]
Could you please advice me how to create log all calls or only for
those using Bye/Also. I've made some researche using google and found
that SJPhone use this method -
http://www.sjlabs.com/doc/SJphone%20Profiles.pdf .
Thanks in advance,
Pavel
Olle E. Johansson wrote:
Pavel Siderov wrote:
I have a customer who is running fairly large conferences (between 5
and 30 participants) on their Asterisk box. It uses SIP to talk to a
PSTN provider.
They are complaining that under some circumstances they experience
echo of one or more participants. On listening in to one of their
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