Hey dude... they email you the config.. but you might wanna have your
priority numbers correct.
exten = _1NXXNXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN},40,tT)
exten = _1NXXNXX,2,Playback,vm-goodbye
On Mon, 17 Nov 2003, Azher Amin wrote:
voicepulse works fine for me ..
In
Hello,
I am trying to install the cdr-mysql. Information given in the following
kink is what I am trying to follow;
http://www.voip-info.org/wiki-Asterisk+cdr+mysql
I cant figure out where to install the asterisk-addons. Is it in /usr/src or
/usr/src/asterisk ?
Once I create the
Maybe someone here has found a good solution to this problem.
I voulenteer with a local Search And Rescue unit and I was speaking with the
senior members about how they interface the command trailer PBX with the PSTN
or cellular networks when they are on scene at a remote location. Turns out
Sathya Weerasooriya wrote:
Hello,
I am trying to install the cdr-mysql. Information given in the following
kink is what I am trying to follow;
http://www.voip-info.org/wiki-Asterisk+cdr+mysql
I cant figure out where to install the asterisk-addons. Is it in /usr/src or
/usr/src/asterisk ?
Once I
Hi;
Just a little question about SIP.
Is there silence detection with SIP ?
If yes can I suppress it ?
I use asterisk with SJPhone and I think there
silence detection or maybe my ear doesn't hear well :)
Regards
Rattana
Hi Andrew,
I have a similar challenge. I will have to connect a remote location with
PBX to a central location with PBX. While roaming the Internet I came
accross this:
http://www.nokia.com/nokia/0,8764,43170,00.html
Two PSTN/GSM gateways called the Nokia 22 and the Nokia 32. I don't know how
Hi,
I have even now connected to IAXtel at number 1-700-895-5211
when I am in the office, so Asterisk is great.
I just found something strange, which is that if I am already in a
connection with my Grandstream and talking, and a second call comes in,
it rings on the Grandstream.
However, if I
Surprise, people. While geeks may be all in favor of used equipment
(and yes, most of it is probably no worse than new equipment), there are
many customers who are uncomfortable with buying used equipment,
probably because many of them have gotten burned in the past.
I probably shouldn't
I keep getting a error message every time a call comes in via a VOIP
source ..
NOTICE[262161]: File rtp.c, Line 417 (ast_rtp_read): Unknown RTP codec
19 received
Every time I enter a digit asterisk produces on line of this
error/notice, but it is recognizing the numbers correctly though.
WipeOut wrote:
Sathya Weerasooriya wrote:
Hello,
I am trying to install the cdr-mysql. Information given in the following
kink is what I am trying to follow;
http://www.voip-info.org/wiki-Asterisk+cdr+mysql
I cant figure out where to install the asterisk-addons. Is it in
/usr/src or
I voulenteer with a local Search And Rescue unit and I was speaking with the
senior members about how they interface the command trailer PBX with the PSTN
or cellular networks when they are on scene at a remote location. Turns out
they don't. Thus that got me to thinking about how one
Hello,
Did somebody tried to make a script to check e-mail
with POP3?
And pass it to Asterisk Festval ?
Bart.
I wasn't able to figure out why they have two models, but basically, you take your SIM
card out of your GSM phone and put it in the device. It turns your cellphone into a
FXO(?). You'll need a X100p or a channel bank to plug it into your asterisk.
Sucks if you use CDMA...
-
Andrew Thompson
Hi,
...
I won't bother with any of that - purchase a Nokia Premicell (or other
manufacturers similar item). This device takes a normal GSM SIM card and
then presents a normal PSTN line interface - plug that into your normal
Asterisk PSTN line card - job done.
A PCI and/or USB device,
All,
is anyone using Bayonne in conjunction with Asterisk? I'm currently using
only Bayonne, but I'm investigating the possibilities of switching the
telephony frontend over to Asterisk, and have Asterisk route the IVR tasks
to Bayonne through H323.
Anyone care to share his views on this
Hi,
I notice something with asterisk with the System
application.
When I lauch asterisk with -c option the
application System work correctly.
But when I lauch asterisk without option, the
application System doesn't lauch command.
It is normal ?
Regards
Rattana
Dan wrote:
Hi,
...
I won't bother with any of that - purchase a Nokia Premicell (or other
manufacturers similar item). This device takes a normal GSM SIM card and
then presents a normal PSTN line interface - plug that into your normal
Asterisk PSTN line card - job done.
A PCI and/or USB
Marc,
This is the typical behavior for call waiting. While you are initiating a
call, people who call your number will get a busy signal until your first
call connects. Once the call connects, the number 2 caller will hear a ring
until you pickup.
If you want to disable callwaiting then put
Does anyone have a good HOWTO on queues Is your computer infected with a virus? Find out with a FREE computer virus scan from McAfee. Take the FreeScan now!
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Has anyone experienced low volume with
the X100P FXO card?
Yes, but, just increase the rxgain in your
zapata.conf and it will likely take care of the issue.
Josh
- Original Message -
From:
Kevin
To: [EMAIL PROTECTED]
Sent: Tuesday, November 18, 2003 9:12
AM
Subject: [Asterisk-Users] Low Volume
X100P
Has
Tried two different Win2k systems and it crashes on load.
-
Andrew Thompson
Your eyes are weary from staring at the CRT. You feel sleepy. Notice how restful it is
to watch the cursor blink. Close your eyes. The opinions stated above are yours. You
cannot imagine why you ever felt otherwise.
Hi, all,
I want to use asterisk SIP as a proxy, and two softphone (Ubiquity SIP Phone) as user agent, like below:
Softphone1--Asterisk SIPSoftphone2
(User Agent) (Proxy) (User Agent)
155.69.xx.xx155.69.yy.yy 155.69.zz.zz
zhoumysipproxy.com Reltec
If I use
Hi, all,
I want to use asterisk SIP as a proxy, and two softphone (Ubiquity SIP Phone) as user agent, like below:
Softphone1--Asterisk SIPSoftphone2
(User Agent) (Proxy) (User Agent)
155.69.xx.xx155.69.yy.yy 155.69.zz.zz
zhoumysipproxy.com Reltec
If I use
Hi,
- Original Message -
From: David Uzzell [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, November 18, 2003 3:54 PM
Subject: Re: [Asterisk-Users] PBX (Asterisk) - Cellular Phone Network
Dan wrote:
Hi,
...
I won't bother with any of that - purchase a Nokia Premicell
Did anybody tried this card with asterisk ?
http://www.digi.com/pdf/prd_mca_datafirequad.pdf
Hi,
- Original Message -
From: Andrew Thompson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, November 18, 2003 4:36 PM
Subject: Re: [Asterisk-Users] Updated iaxComm binaries available for WinXP,
Red Hat 9.0
Tried two different Win2k systems and it crashes on load.
Tried on
Hello All,
I'm about 1/2 way through building an * system for home office. I have
the server built, * installed, but I'm waiting for a Grandstream phone
to arrive to have a reliable client. In the mean time I have a few
short questions.
1. I have the X-Pro soft phone client. It's presently
Hello,
I have set one function key of my IP10s to send the flash event and it
seems to work but only for blind transfert and not for consultative
transfert or call conferencing. How should I do to use the flash key to
do that?
Regards,
Daniel
Florian Overkamp wrote:
Hi,
About once every day my * goes nuts and "asterisk
-r" responds with "broken pipe". All calls are dropped immediately, even
extension 600 (echo). Killing the process and restarting asterisk helps... until
next day. I'm running 0.5.0 release on RH9. Any ideas what's wrong, and what can
I do
We are using a CDMA/PSTN adapter (Motorola 800SC) interconnecting the
cellular network to our pabx. In our case the application is different:
the cell-to-cell calls are cheaper that pstn = cell calls. A fxo
interface in required your pabx.
Jorge
Andrew Thompson wrote:
I wasn't able to figure
Hi there,
here some questions and experiences after playing for one day with 3
Swissvoice ip10s and the latest * CVS:
QUESTIONS:
- what is the user option enter voice mail number good for? It doesn't
appear to be of any practical use
- does anyone have some Swissvoice info that I cannot find
At 8:49 PM +0200 11/17/03, Pertti Pikkarainen wrote:
Some of you have got Wifi600 wireless SIP phone working with Asterisk.
Specially John Todd ( nice review ).
My phones register ok. They can also receive calls from other phones.
But for some reason I can't make them call out ( anybody, ie. SIP
Martin,
As I said in my previous email I was using a SIP phone (SJphone). If you
have a solution please let me know.
Regards,
Jordi
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Martin Pycko
Sent: Friday, November 14, 2003 10:28 PM
To: [EMAIL
Hi,
I have DIVA server BRI with 2 channels and i use
chan_capi drivers. But I only can use 1 channel. I make one call it works, but
if I make a second call asterisk says me = Everyone is busy at this
time.
How can I configure it ?
Best regards
Rattana
On Mon, 17 Nov 2003 20:30:09 -0500 (EST), Bob Bevins [EMAIL PROTECTED]
wrote:
-- Playing 'beep' (language 'en')
-- x=0, open writing:
/var/spool/asterisk/voicemail/local/31/INBOX/msg0011 format: wav49,
0x80de8b8
-- x=1, open writing:
/var/spool/asterisk/voicemail/local/31/INBOX/msg0011
At 09:45 18-11-2003 -0500, you wrote:
And yes, they can run fine together(I'm not using VOIP, just a T1 out of
Asterisk to Bayonne to test and see if it would work). The IVR application
that I currently still have running on Bayonne is only still on Bayonne
because it can never go down, and
Title: ISDN Card Types for Europe
What types of ISDN BRI cards work well in Europe (Guadeloupe, Martinique and France) ? For example: AVM C2 or AVM C4 or eicon Diva server 4 BRI? Any others? Which driver is appropriate?
Ray Burkholder
[EMAIL PROTECTED]
http://www.oneunified.net
704 576
Hi,
- Original Message -
From: Michael Van Donselaar [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, November 18, 2003 5:11 PM
Subject: Re: [Asterisk-Users] Updated iaxComm binaries available for WinXP,
Red Hat 9.0
On Tue, 18 Nov 2003 09:36:23 -0500, you wrote:
Tried two
Thanks John,
Can you check what version you are using ?
I can start with the very same ( once I get it ).
I have sent a request to BCM but haven't got any reply yet.
-- Pertti
John Todd wrote:
At 8:49 PM +0200 11/17/03, Pertti Pikkarainen wrote:
Some of you have got Wifi600 wireless SIP
On Tue, 18 Nov 2003 17:02:42 +0200, Dan [EMAIL PROTECTED] wrote:
Hi,
Tried on WinXP Pro and it loads, but in the background (no window).
There is something needed from the wxWindows package to just run the
executable?
Nothing needed from the wxWindows package. I think it's because it can't
Hi, I want to correct an error, in my figure, the softphone2's name is Raytec, not Reltec. As the figure below shows:
Thanks!
Softphone1Asterisk SIPSoftphone2(User Agent)(Proxy) (User Agent)155.69.xx.xx 155.69.yy.yy 155.69.zz.zz zhoumysipproxy.comRaytec
Title: DIAX - Can place a call, but can't be called?!
Greetings,
DIAX seems to work well placing calls, but I can't actually receive a call . Here, DIAX (x305) registers, then I use a sip phone to place a call to DIAX (which definitely is not in use by me at debug time, but it is idle on
Title: RE: [Asterisk-Users] Transfer directly to voicemail?
Does anyone have a setup ware a user can pickup voice messages from a group type voice box and from his own one in one go to the voice mail?
or to notify the user that hi have a message in his (sales) group mail box
Many Thanks
On Tue, 2003-11-18 at 09:53, Florian Overkamp wrote:
At 09:45 18-11-2003 -0500, you wrote:
And yes, they can run fine together(I'm not using VOIP, just a T1 out of
Asterisk to Bayonne to test and see if it would work). The IVR application
that I currently still have running on Bayonne is only
On Tue, 2003-11-18 at 08:53, Florian Overkamp wrote:
Actually in this light it might be cute to have an 'uptime' counter inside
asterisk (maybe a lastlog that can also show the reason of the last restart
- was it a stop gracefully or did it just crash?) *grin*
Have you tried show uptime from
Hi guys,
I'm new to the telco game and still pretty new to Asterisk, although
I've been using it for a couple of months now and like most of what I
see. At my office, we've got a small two extension setup w/ two Digium
cards for a single FXO line and three FXS extensions, but I'm also
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Jason A. Pattie wrote:
| I have this exact same problem as well. We have a scenario in which we
| are not using any analog extensions, just SIP and IAX software based
| phones (DIAX, X-Lite, gnophone, (trying to use) linphone, etc.) with a
| single
On Mon, 2003-11-17 at 08:34, Steve Murphy wrote:
Hello--
I've been asked an interesting question, and I'm too ignorant to answer
it authoritatively (yet). Can anyone help me?
Question: If I'm going to implement a somewhat small (10-80) phone
system, and I have a choice of using VOIP phoneset
The version of the WiFi software that I am running that is confirmed
to work with Asterisk is wb000_d.img
JT
Thanks John,
Can you check what version you are using ?
I can start with the very same ( once I get it ).
I have sent a request to BCM but haven't got any reply yet.
-- Pertti
John
ohhh , thnx. btwit was my typo mistake :)
AzherBrian West [EMAIL PROTECTED] wrote:
Hey dude... they email you the config.. but you might wanna have yourpriority numbers correct.exten = _1NXXNXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN},40,tT)exten = _1NXXNXX,2,Playback,vm-goodbyeOn Mon, 17
On Tue, 2003-11-18 at 11:04, john lawler wrote:
Hi guys,
working on designing a larger installation for a customer which will
involve ~16 analog handsets that I'll be running through a Rhino
Equipment channel bank to a Digium T1. My question is all about what
type of phone service would
At 1:20 PM + 11/18/03, Linus Surguy wrote:
I voulenteer with a local Search And Rescue unit and I was speaking with
the
senior members about how they interface the command trailer PBX with the
PSTN
or cellular networks when they are on scene at a remote location. Turns
out
they don't.
If anyone could help me with this, I'd appreciate it!
I've got an Asterisk deployment where I'd like to use an existing external Octel voicemail system. I've been trying to define an extension that if the call isn't answered in a few rings, to dial our external voicemail number. That voicemail
Title: Message
We
deploy the Eicon Diva Server range of cards for production systems as they have
onboard echo cancellation, and work very well with chan_capi and
asterisk.
Tan
www.voiptalk.org
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
interface the command trailer PBX with the PSTN
or cellular networks when they are on scene at a remote location. Turns
out
they don't. Thus that got me to thinking about how one would get Asterisk
to
interface with a cell phone directly, or what hardware out there works
well
for the task.
Some of the older cell phones use to expose either a 2-wire or 4-wire
interface via a connector on the phone (don't know about transmission
I won't bother with any of that - purchase a Nokia Premicell (or other
manufacturers similar item). This device takes a normal GSM SIM card and
I'm trying to clean up some notices/warnings that are repeatedly logged
in *.Any Help would be appreciated as I'm not sure of the cause
/solution.
Here are the errors:
Nov 17 15:53:38 WARNING[1217602880]: File chan_zap.c, Line 1321
(zt_call): cidspill already exists??
Hi,
- Original Message -
From: Todd Taylor [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, November 18, 2003 6:22 PM
Subject: [Asterisk-Users] DIAX - Can place a call, but can't be called?!
Greetings,
DIAX seems to work well placing calls, but I can't actually receive a
call
--- Andrew Thompson [EMAIL PROTECTED] wrote:
Tried two different Win2k systems and it crashes on load.
Doesn't crash for me, just don't get anything. Continues to run in the background but
no interface. I have to ctrl-alt-del to end it.
Kevin
Tried two different Win2k systems and it crashes on load.
You currently have to run it from the directory it is located in. If you make a
shortcut, make sure that Run In: has the correct directory.
iaxComm needs to see the resources in ${cwd}/rc
Ok, I tried 'running it from the directory it is
Hi,
- Original Message -
From: Kevin Bockman [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, November 18, 2003 8:26 PM
Subject: Re: [Asterisk-Users] Updated iaxComm binaries available for WinXP,
Red Hat 9.0
--- Andrew Thompson [EMAIL PROTECTED] wrote:
Tried two different Win2k
Hi,
Do you know if we can use AGI or other script to handle the
asterisk events by using the existing asterisk manager process ?
Please advise.
Thanks
George
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Ah ha. That's *almost* got it. It will now load and * will run. The
only big gotcha is it won't pick up or dial out on a POTS line. ztcfg
shows both channels configured OK, as does 'zap show channels.' If I
try to dial out I get:
-- Executing Goto(SIP/3063-74d0, outside|9555|1) in new
SIP Express Router have radius support. Look there for hints on how to get
Radius support for VOIP.
http://iptel.org/ser/
/O
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How to set up MusicOnHold insted of Ringng
tone?
Bart.
On Tue, 2003-11-18 at 11:56, [EMAIL PROTECTED] wrote:
If anyone could help me with this, I'd appreciate it!
I've got an Asterisk deployment where I'd like to use an existing
external Octel voicemail system. I've been trying to define an
extension that if the call isn't answered in a few
Smells like a -dev type discussion.
Some of these messages could be removed for your use, or just change
what you want to see in your logs.
On Tue, 2003-11-18 at 12:18, Andy Hester wrote:
I'm trying to clean up some notices/warnings that are repeatedly logged
in *.Any Help would be
No and there is absolutely no need for it to. RADIUS is not anything
that should have ever been deployed in a VoIP environment.
sarcasm
I have no need for H.323 or Skinny or SIP; IAX works fine...
H.323 is something that should never ever have been invented...
/sarcasm
I fully agree
On Tue, 2003-11-18 at 12:37, George Lin wrote:
Hi,
Do you know if we can use AGI or other script to handle the
asterisk events by using the existing asterisk manager process ?
AGI is for handling calls. AGI is to phone calls like CGI is to web page
requests.
There is a perl module to use
I guess I should have been more specific. When I said
Running 'sip debug' does not solve this problem.
What I meant was that SIP debug doesn't show me
anything.
It turns out it was a dumb codec error on my part.
Problem solved.
--- Andrew Thompson [EMAIL PROTECTED] wrote:
- Original
Hello,
I have finished my basic polishing of the Asterisk GUI client I have been
writing in Perl/TK and have released a first beta version on sourceforge:
http://sourceforge.net/projects/astguiclient/
I am still working on a user manual for the application, but the code works
and we have been
Hi,
Is it possible to pick the context of a call from chan_sip based on the
domain of the To: header of the INVUTE? I've had a quick look throught he code
and can't see anything, I want to use the voicemail virtual hosting with
chan_sip. Can the sip domain be picked out with a global in
hi.
Il mar, 2003-11-18 alle 20:33, Andrew Kohlsmith ha scritto:
sarcasm
I have no need for H.323 or Skinny or SIP; IAX works fine...
H.323 is something that should never ever have been invented...
/sarcasm
I feel the same way about SIP. What a nasty protocol. :-(
the really big
mattf wrote:
Hello,
I have finished my basic polishing of the Asterisk GUI client I have been
writing in Perl/TK and have released a first beta version on sourceforge:
http://sourceforge.net/projects/astguiclient/
I am still working on a user manual for the application, but the code works
and we
/var/log/asterisk/event_log should have the last time the server restarted.
This should closely match "show uptime" result.
Jared Smith wrote:
On Tue, 2003-11-18 at 08:53, Florian Overkamp wrote:
Actually in this light it might be cute to have an 'uptime' counter inside
asterisk
At 8:00 PM + 11/18/03, Tristan 'Minty' Colgate wrote:
From: Tristan 'Minty' Colgate [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] SIP Context from domain?
Reply-To: [EMAIL PROTECTED]
Date: Tue, 18 Nov 2003 20:00:55 +
Hi,
Is it possible to pick the context of a call
At 09:52 18-11-2003 -0700, you wrote:
On Tue, 2003-11-18 at 08:53, Florian Overkamp wrote:
Actually in this light it might be cute to have an 'uptime' counter inside
asterisk (maybe a lastlog that can also show the reason of the last
restart
- was it a stop gracefully or did it just crash?)
We're still having problems with DTMF detection on our X100P cards.
Incoming callers that hold down the 1 button for too long are being
connected to extension 11. One would think fat fingers were uncommon,
but it happens to alot of people.
I suspected this was related to our having to increase
Anybody knows about a FREE or GPL G723.1 capable softphone for windows?
thanks anyone.
Hector.
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I seem to be having a problem with transcoding and/or agreeing on a
valid codec. I am running a new image pulled from CVS at 1:30 PM CST.
The issue occurs when I try to make a call to a toll-free number over
sipphone.com.
Here's what I see in the console:
NOTICE[1259545280]: File channel.c,
Hello,
The reason I used MySQL is for simplicity on the client end, there is a perl
module Net::MySQL that requires no extra libraries to be installed on the
machine, unlike Postgres module, and the Database routines used by this
program are hardly database intensive. It's easy enough to install
Steven Critchfield wrote:
On Tue, 2003-11-18 at 12:37, George Lin wrote:
Hi,
Do you know if we can use AGI or other script to handle the
asterisk events by using the existing asterisk manager process ?
AGI is for handling calls. AGI is to phone calls like CGI is to web page
requests.
There is
Hi Matt,
After a first look to the screen shoot, it sounds incredible...
Good work ;P
Will it work without Zaptel interface ?
Aresk
On Tue, 2003-11-18 at 20:03, mattf wrote:
Hello,
I have finished my basic polishing of the Asterisk GUI client I have been
writing in Perl/TK and have
Ive tried that, all it seems to do
is distort the audio quality and add much more echo. I see others have had this same problem and I
was wondering if it was resolved or just living with it.
-Original Message-
From: Josh J. Zuerner
[mailto:[EMAIL PROTECTED]
Sent: Tuesday,
On 17-11 16:33, Jeremy McNamara wrote:
Sebastian Nocetti wrote:
Does Asterisk support Radius accounting?
No and there is absolutely no need for it to. RADIUS is not anything
that should have ever been deployed in a VoIP environment.
You would be surprised how many people
Hi,
- Original Message -
From: Hcqm [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, November 18, 2003 10:02 PM
Subject: [Asterisk-Users] G723.1 Softphone for windows
Anybody knows about a FREE or GPL G723.1 capable softphone for windows?
thanks anyone.
Hector.
Netmeeting can
Hello,
Thanks :)
Sorry, it will not work out of the box without a zaptel device, you will
have to go in and tinker with the code a bit if you want it to work. Also,
some of the functions only work with Zap devices(call recording). Everything
I do with Asterisk is with SIP devices and T1s through
We still have a few of the 3com phones in use at our company but we do
not support them with our
SIP products. The 3com phone was meant to be a PBX feature phone as you
stated and as a result
the flash ROM and RAM was not beefy enough to support the SIP protocol
as it matured. The last
ROM
Hcqm wrote:
Anybody knows about a FREE or GPL G723.1 capable softphone for windows?
thanks anyone.
No and there won't be until the G723.1 patents expire (sometime in 2008
I think).
--Eric
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Hello,
If there's enough demand I can put a configuration variable into the
client/server apps to have it switchable from MySQL to PostgreSQL. That
should be fairly easy to do. I just need to know which Perl module for
postgreSQL is the most acceptable one to use.
MATT---
-Original
callwaiting=no is not supported by chan_sip. Call waiting
enabling/disabling is a function of SIP phones. Unfortunately, GS does not
support disabling call waiting as yet, so I've had to put in a patch to
overcome the problem. Look under
http://bugs.digium.com/bug_view_page.php?bug_id=408. You
If there's enough demand I can put a configuration variable into the
client/server apps to have it switchable from MySQL to PostgreSQL. That
should be fairly easy to do. I just need to know which Perl module for
postgreSQL is the most acceptable one to use.
DBI is the most acceptable, IMO.
Can you put up some screenshots
On Tue, 18 Nov 2003, mattf wrote:
Hello,
If there's enough demand I can put a configuration variable into the
client/server apps to have it switchable from MySQL to PostgreSQL. That
should be fairly easy to do. I just need to know which Perl module for
On Wed, Nov 19, 2003 at 08:27:31AM +1100, Paul Liew wrote:
callwaiting=no is not supported by chan_sip. Call waiting
enabling/disabling is a function of SIP phones. Unfortunately, GS does not
support disabling call waiting as yet, so I've had to put in a patch to
overcome the problem. Look
Where can I buy the Wifi600 phones ??
Or does anyone know of other Wifi SIP phones ??
Any help would be appreciated
Regards Mick
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[EMAIL PROTECTED]
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The original post had them, here they are again:
Here are the screen shots of the same application running on Linux and
Windows:
http://www.freedomphones.net/astguiclient_linux.gif
http://www.freedomphones.net/astguiclient_windows.gif
MATT---
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[EMAIL PROTECTED] wrote:
I seem to be having a problem with transcoding and/or agreeing on a
valid codec. I am running a new image pulled from CVS at 1:30 PM
CST.
The issue occurs when I try to make a call to a toll-free number
over
sipphone.com.
Here's what I see in the console:
http://www.pulverinnovations.com/
On Tue, 2003-11-18 at 21:57, [EMAIL PROTECTED] wrote:
Where can I buy the Wifi600 phones ??
Or does anyone know of other Wifi SIP phones ??
Any help would be appreciated
Regards Mick
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Asterisk-Users
anyone knows if this phones now support auth with sid ??? my school
wireless lan needs the auth
Miguel
On Tue, 2003-11-18 at 21:57, [EMAIL PROTECTED] wrote:
Where can I buy the Wifi600 phones ??
Or does anyone know of other Wifi SIP phones ??
Any help would be appreciated
Regards Mick
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