Hi
U can visit the http://iaxclient.sf.net for some opensource underdevelopment
softphones.
Take Care
Obaid Amin Syed
From: Chris Albertson [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Good W2K softphone
Date: Fri, 3 Oct 2003 23:00:13 -0700
I think ODBC is the way to go. There is really nothing to write.
You can't anymore MySQL was ripped from Asterisk because the client
libs
are GPL.
I would be more than happy to help write a DB Virtualization
function for *.
I *love* the way it works in Java, but that's not a real
I am using Audacity to record some voice prompts.
The .wav files I'm producing are of stellar quality. However, once I
turn them into .gsm, they sound buzzy and muffled.
I know that some of this comes with the territory, but I wonder if there
is anyone out there who does this routinely, and
- Original Message -
From: Brian Capouch [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, October 06, 2003 5:21 PM
Subject: [Asterisk-Users] Anyone else use Audacity for prompts?
I am using Audacity to record some voice prompts.
The .wav files I'm producing are of stellar
Shaun Ewing wrote:
I know that some of this comes with the territory, but I wonder if there
is anyone out there who does this ( wav - gsm) routinely, and who can advise me as to
the MO I could use that results in the highest quality in the resulting
playback files.
What are you using to convert
I have had been recording my gsm files by getting through to the Asterisk
answering service using a GrandStream BudgeTone 102 phone. I then copy
the file which is stored in voicemail and use sox to increase the volume.
Results are okay but nothing to write home about particularly (or maybe
On Mon, 6 Oct 2003, Brian Capouch wrote:
I was just hoping there was something I could do to make the resulting
files a bit clearer.
put them through baudline and see what's happening
also you might wanna try bandpass filter using ecasound
- wasim
Up until yesterday I've had a lot of high pitched noise when connecting
a BT101 to the PSTN via the X100P. I was using an Asus A7V133 with raid
motherboard and an 850 AMD Duron. Over the weekend I thought I'd try
another machine. I had an HP Vectra 400 Mhz PII MMX with 128 Mb RAM
available, today
On 06/10/03 08:25, Shaun Ewing wrote:
The .wav files I'm producing are of stellar quality. However, once I
turn them into .gsm, they sound buzzy and muffled.
An example line to convert:
sox file.wav -r 8000 -c 1 file.gsm
It'll sound much better if you go:
sox file.wav -r 8000 -c 1 file.gsm
Dave Cotton wrote:
Up until yesterday I've had a lot of high pitched noise when connecting
a BT101 to the PSTN via the X100P. I was using an Asus A7V133 with raid
motherboard and an 850 AMD Duron. Over the weekend I thought I'd try
another machine. I had an HP Vectra 400 Mhz PII MMX with 128 Mb
I've had good luck with X-Lite (www.xten.com).
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of God Knows
Well
Sent: Monday, October 06, 2003 1:03 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Good W2K softphone
Hi
U can visit the
I failed to mention in my original post that I've looked at perl through
AGI, but haven't yet found a function that allows me to capture digits to
a variable that I can then manipulate. I probably should also mention
that I'm not a programmer-type, although I can usually muddle through
kapejod,
Thanks for the prompt reply, that didn't work, although I managed to work
it out, and the msn number I need to put in is the position number in the
S0 bus, for example for X616 it would 1, X617 2 and so on. Thanks for the
response it allowed me to work it out.
Thanks
Dave Sykes
Head
Hi everyone,
I know someone makes a product that's a POTS phone to SIP converter,
where you just plug your POTS phone in one side and the network cable in
the other. Has anyone successfully used any of these with Asterisk, and
if so how expensive were they?
I ask partly out of frustration
Matt Lawson wrote:
Hi everyone,
I know someone makes a product that's a POTS phone to SIP converter,
where you just plug your POTS phone in one side and the network cable
in the other. Has anyone successfully used any of these with
Asterisk, and if so how expensive were they?
I ask partly
Greetings
I
am a new user of Asterisk and am beginning to make tests.
My problem is the following one:
I
have a Cisco ATA 186, already obtains that it was validated in asterisk, but I
have a doubt of like being able to declare the extenciones for the equipment.
This
it I must
Hi All,
I´m thinking in install apache in my asterisk machine to host a litle site.
Anybody knows about problems doing that?
thanks
miklos
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On Mon, 6 Oct 2003, listas iPfone wrote:
I´m thinking in install apache in my asterisk machine to host a litle site.
Anybody knows about problems doing that?
I've got apache installed on my asterisk server - it's handy for setting
up calls (a cgi just drops a call file into the outgoing call
listas iPfone wrote:
Hi All,
I´m thinking in install apache in my asterisk machine to host a litle site.
Anybody knows about problems doing that?
thanks
miklos
I also run a mail server on mine... among varoius other things
Neil
___
We are trying Asterisk for the first time, and have been unable to get
the X-Lite softphone working (send/receive). Can anyone provide a
sip.conf that works?
Thanks
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joe Dennick
Sent: Monday, October 06,
Sip.conf
[7002]
type=friend
secret=blah
host=dynamic
You will also need an entry for 7002 in extensions.conf. You can then
configure the X-lite phone to use 7002 as both its user and
authenticated user with a password of 'blah'. Make sure you select the
'Send Internal IP' option to 'On'. It
On Mon, 2003-10-06 at 15:57, listas iPfone wrote:
Hi All,
I´m thinking in install apache in my asterisk machine to host a litle site.
Asterisk + Openvpn + Apache + MySql + Postfix + Amavisd + Sophie + local
mirror of Mandrake Cooker.
This is Linux at work not M$ at play.
--
Dave Cotton
Hi,
Trying to compile gnophone and am having a bit of a time finding the
source for phonecore. Anyone know of somewhere I can pull the source from?
Thanks,
James-
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[EMAIL PROTECTED]
Hello,
Could somebody tell me where I can find information
how should look
like the structure of database for dbget, dbput and
so on
Thank you,
-- Bart
Hi,
Has anyone managed to get X-Lite to work with Asterisk using the iLBC
codec.. I have just tried updating the the latest version 1079 (BTW this
new version supports up to 10 proxy configurations, Not that I can see a
reason to have 10 proxies setup, I would rather have the ability to
WipeOut wrote:
Hi,
Has anyone managed to get X-Lite to work with Asterisk using the iLBC
codec.. I have just tried updating the the latest version 1079 (BTW this
new version supports up to 10 proxy configurations, Not that I can see a
reason to have 10 proxies setup, I would rather have the
On Thu, 02 Oct 2003 11:26:56 -0700, Jan Rychter [EMAIL PROTECTED]
wrote:
Having worked with GPL software quite a bit, also in the commercial
world, and having gotten some legal advice, I believe that the
anti-patent clauses in the GPL and LGPL are quite possibly the biggest
problem preventing the
And a followup for some debug messages for ManxPower:
*CLI -- Starting simple switch on 'Zap/1-1'
NOTICE[262161]: File callerid.c, Line 238 (callerid_feed): Caller*ID
failed checksum
ERROR[262161]: File callerid.c, Line 192 (callerid_feed): fsk_serie
made mylen 0 (-30)
WARNING[262161]: File
Leif Madsen wrote:
WipeOut wrote:
Hi,
Has anyone managed to get X-Lite to work with Asterisk using the iLBC
codec.. I have just tried updating the the latest version 1079 (BTW
this new version supports up to 10 proxy configurations, Not that I
can see a reason to have 10 proxies setup, I
His Telco is ATT Broadband, Caller*ID works on an analog phone.
On Mon, 2003-10-06 at 10:11, Chris Hirsch wrote:
And a followup for some debug messages for ManxPower:
*CLI -- Starting simple switch on 'Zap/1-1'
NOTICE[262161]: File callerid.c, Line 238 (callerid_feed): Caller*ID
failed
I am relatively new at Asterisk but have a 2-machine system running with
the developer kit (FXS/FXO cards), some Cisco SIP 7960's and a Audiocodes
MP-104 SIP gateway. I would like to fix a couple of the voicemail boxes
so someone can press some numbers such as 911 and get sent to a priority
Hi!
I have an H.323 network. I am trying to do a SIP-H.323 gateway to be able to
accept all the calls that come from other networks (SIP).
I have some questions:
- Must I use IAX? If so, how? I have not SIP servers, I only have a H.323
gatekeeper.
- If not, where is that I say that one call
And a followup for some debug messages for ManxPower (2nd time since I
can't seem to get all the details :-) ):
FXO Card Installed:
[EMAIL PROTECTED] asterisk]# cat /proc/zaptel/1
Span 1: WCFXO/0 "Wildcard X101P Board 1"
1 WCFXO/0/0 FXSKS (In use)
Analog Line:
ATT Broadband coming
On Monday 06 October 2003 01:43 am, Chris Albertson wrote:
I think ODBC is the way to go. There is really nothing to write.
That's great. Why haven't you written and contributed it yet?
-Tilghman
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Why, oh why, do we have to be limited to 8kHz prompts in the first place?
Alastair Maw wrote:
On 06/10/03 08:25, Shaun Ewing wrote:
The .wav files I'm producing are of stellar quality. However, once I
turn them into .gsm, they sound buzzy and muffled.
An example line to convert:
sox
On Mon, 2003-10-06 at 11:08, Brad Waite wrote:
Why, oh why, do we have to be limited to 8kHz prompts in the first
place?
Because that is what telephony is based on. 8khz by 8 bit if on a
digital link and 7 bit if in RBS signaling.
Why are you so worried about that amount of degradation when
I have had to work on some files recently with a similar problem.
It seems that when a file is recorded in 16 bit and converted to 8 bit,
the clarity is lost.
I have found the following ways the most productive:
1)Record through the voicemail system then import and edit them
afterwards. As long
And then use standard Unix commands to move that recording to where you
want it like /var/lib/asterisk/sounds/new-recording.gsm so you can then
call it from your menus or prompts.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian West
Sent: Monday,
Where does one find a hard-phone for $65?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Leif Madsen
Sent: Monday, October 06, 2003 11:33 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk, X-Lite and iLBC..still..
WipeOut wrote:
Leif
Two phones for $130 at
http://www.sipphone.com/tiki-index.php?page=Order%20Now
On Mon, 2003-10-06 at 11:42, Joe Dennick wrote:
Where does one find a hard-phone for $65?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Leif Madsen
Sent: Monday,
On Monday 06 October 2003 09:40 am, Bartosz Jozwiak wrote:
Hello,
Could somebody tell me where I can find information how should look
like the structure of database for dbget, dbput and so on
Yep, see the manpage for dbopen(3).
-Tilghman
___
Joe Dennick wrote:
Where does one find a hard-phone for $65?
Sorry, should have been $75..
But if you look at sipphone.com you can get two and it will cost you $65
each..
Later..
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Hi
I work at a small company that has some IVR solutions that use Dialogic
hardware for everything.
Everything is written in C++ using MS VC++ using the Dialogic API and runs
only on Windows.
Being the rebel that I am, I would like free myself from Dialogic.
To do this without porting all our
On Monday 06 October 2003 12:13 pm, Ívar Ragnarsson wrote:
I work at a small company that has some IVR solutions that use
Dialogic hardware for everything.
Everything is written in C++ using MS VC++ using the Dialogic API
and runs only on Windows.
Being the rebel that I am, I would like free
Which one would one should I use to solve my problem? Does an loadable
application give you more control than an AGI script?
If you want something that runs continuously (such as a listener process
or control process), it'll have to be a loadable module. AGI scripts only
get run when the
Hi Ivar-
I was in a similar position, doing mostly IVR apps, and having written lots
of Dialogic-based code.
The approach I took was to re-write some of my more popular applications in
Perl and use Asterisk's AGI interface. However, if I were in your position
with lots of applications to
On Mon, 6 Oct 2003, James Coberly wrote:
Hi,
Trying to compile gnophone and am having a bit of a time finding the
source for phonecore. Anyone know of somewhere I can pull the source from?
I have a copy of the most recent gnophone source located at
WipeOut wrote:
Joe Dennick wrote:
Where does one find a hard-phone for $65?
Sorry, should have been $75..
But if you look at sipphone.com you can get two and it will cost you $65
each..
You can get them for $65 at
https://secure.pulver.com/pulverinnovations/order_grandstream.html
--- Tilghman Lesher [EMAIL PROTECTED] wrote:
On Monday 06 October 2003 01:43 am, Chris Albertson wrote:
I think ODBC is the way to go. There is really nothing to write.
That's great. Why haven't you written and contributed it yet?
Any code I write for free is GPL'd. If they don't accept
Hi all!
One easy question... I hope someone will answer me.
I've installed asterisk with the samples. Somewhere in my network I have an
H.323 Gatekeeper. What must I do to make that the gatekeeper talk with
Asterisk?
And I another little question... with the samples installed asterisk works
ok?
It seems consistant after dialing dozens of times that the call that
doesn't go through is the one the gets the log message No
echocancellation requested (chan_zap.c) and the Scheduleing timer
(channel.c) in the middle of receiving the DTMF tones.
I'm now using the T400P card last week very
On Mon, 2003-10-06 at 12:13, Ívar Ragnarsson wrote:
Hi
I work at a small company that has some IVR solutions that use
Dialogic
hardware for everything.
Everything is written in C++ using MS VC++ using the Dialogic API and
runs
only on Windows.
Being the rebel that I am, I would like free
However, if I were in your
position
with lots of applications to convert, I might consider writing an API to
control the Digium hardware directly. (I'm not sure how hard this would
be). There are likely licensing issues to overcome - open source vs.
proprietary code etc
Just to clarify the GPL,
On Mon, 2003-10-06 at 10:12, Eduardo Goncalves wrote:
Hello,
I have the fowling scenario:
fxs[asterisk1]-iax-[asterisk2]e1em---PSTN
I want to know the steps to transmit fax from a machine connected to the fxs
to a fax machine on the PSTN. The same for
On Monday 06 October 2003 12:57 pm, Chris Albertson wrote:
--- Tilghman Lesher [EMAIL PROTECTED] wrote:
On Monday 06 October 2003 01:43 am, Chris Albertson wrote:
I think ODBC is the way to go. There is really nothing to
write.
That's great. Why haven't you written and contributed
Hi,
On Fri, 2003-10-03 at 15:52, mattf wrote:
I've seen various suggestions thrown around for hardware when people ask,
but can we all agree on some basic hardware recommendations for a few basic
setups(and post them on a website) to make it easier for new people to avoid
some of the
On Mon, 06 Oct 2003 13:43:21 -0500
Steven Critchfield [EMAIL PROTECTED] wrote:
On Mon, 2003-10-06 at 10:12, Eduardo Goncalves wrote:
Hello,
I have the fowling scenario:
fxs[asterisk1]-iax-[asterisk2]e1em---PSTN
If asterisk2 is your only access to the PSTN, then it
On Mon, 2003-10-06 at 13:55, Eduardo Goncalves wrote:
On Mon, 06 Oct 2003 13:43:21 -0500
Steven Critchfield [EMAIL PROTECTED] wrote:
On Mon, 2003-10-06 at 10:12, Eduardo Goncalves wrote:
Hello,
I have the fowling scenario:
On Sat, 2003-10-04 at 18:53, Rich Adamson wrote:
Why not add an Article to the www.voip-info.org site, and those that are
interested with helping can list their FWD, IAXTEL, or other access number,
probable hours of availability, any special focus skills, size of their
current * environment,
OK, I've been playing with it and I must be missing something. Here's a
script that I've written:
#!/usr/bin/perl
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
my $repid =
$AGI-get_data('sai-enter-rep_id', 5, 5);
$AGI-say_digits($repid);
exit;
The script is called as part of Extension
Fax with G711 works fine. Modem will be slow, but if you really need to
use it slown them down to 28.8 or 33.6
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Eduardo Goncalves
Sent: Monday, October 06, 2003 2:56 PM
To: [EMAIL
I've got * up and running everything seems to work ok except for when you
dial out using the X100P card.
Everything sounds great this end but the person you call complains that they
can't hear you very well (Very Whispered).
Is their any way to turn up the volume. I've fiddled with the gain
Eduardo asks:
I want to know the steps to transmit fax from a machine
connected to the fxs to a fax machine on the PSTN.
It was suggested:
Put a couple of modems on asterisk2 with
matching FXS ports and learn to use Hylafax.
Not to take anything away from Hylafax, but our commercial fax server
On Mon, 2003-10-06 at 15:12, Andrew Joakimsen wrote:
Fax with G711 works fine. Modem will be slow, but if you really need
to
use it slown them down to 28.8 or 33.6
This depends on if you have consistent latency and otherwise no jitter.
On my 12 hop link with the office over a cable modem, it
DAve,
JUst wondering whether you can disclose the number of users you have on
your system and what CPU memory and disks you have. I'm looking to do
multiple functions on a single boxen too.
Peter
At 16:25 6/10/2003 +0200, you wrote:
On Mon, 2003-10-06 at 15:57, listas iPfone wrote:
Hi All,
Is there any way to take an incoming callerid string and remove the
given name part of it and replace it w/ something arbitrary, or add to
a blank name string (possibly by looking up the number in a database)?
Thanks,
John Lawler
___
Asterisk-Users
I know that Asterisk supports DID, but does anyone have documentation on
how to write the configuration for it?
I'll be trying to setup a hybrid system where some incoming numbers will
be DID enabled and others won't, so I'll need to be able to sort between
the two, i.e. directly connect the
On Mon, 2003-10-06 at 15:45, john lawler wrote:
Is there any way to take an incoming callerid string and remove the
given name part of it and replace it w/ something arbitrary, or add to
a blank name string (possibly by looking up the number in a database)?
Be glad I'm ridding my angst in
On Mon, 2003-10-06 at 15:45, john lawler wrote:
I know that Asterisk supports DID, but does anyone have documentation on
how to write the configuration for it?
I'll be trying to setup a hybrid system where some incoming numbers will
be DID enabled and others won't, so I'll need to be able
I have been struggling with echo cancellation for the last few days. It
seems to me that it would be useful to start up a technical discussion
of the issue so that we don't have to solve the problem empirically. My
system is SIP (Grandstream) = Asterisk = Adtran TSU600 =FXO
=POTS. From what
Try putting an Answer() in your extensions.conf before you call the AGI
code - a common gotcha I think?
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quote who=Steven Critchfield
On Mon, 2003-10-06 at 15:45, john lawler wrote:
Is there any way to take an incoming callerid string and remove the
given name part of it and replace it w/ something arbitrary, or add
to a blank name string (possibly by looking up the number in a
database)?
Be
It's pretty easy, in your extensions.conf.
If your DIDs are in a range, you could set up some pattern matching to take
a block of incoming DIDs and map to extension numbers then dial or hand off
to the dial'n'voicemail macro thing. If your DIDs are non-contiguous, you'll
have to set up a separate
I am looking for examples or instructions on how to route calls to
voicemailmain based on remote-party-id.
I have the following entry in my extensions.conf file:
exten = 200,1,Voicemailmain(${CALLERIDNUM})
I am routing calls to * via SER and sending Remote-Party-ID in the SIP
headers. I am
Hello,
I am trying to conference two or more calls on a Cisco 7940 phone. When I have one
inbound call and one outbound (I initiate the second call by pressing conference) I
get the join button at the bottom of the screen and I can conference.
When I initiate both calls or I receive both
I'm trying to get a Snom100 configured with H.323. Right now, the
phone is not even connecting to the Asterisk server, so there's
obviously a problem with the snom config. Does anybody have a
sample working configuration with the snom phone, using H.323?
I've checked the archives, but everybody
That makes a lot of sense, but...it still doesn't work.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Crick
Sent: Monday, October 06, 2003 4:18 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] IVR Questions?
Try putting an Answer() in your
Has anyone any experience using the TDP400P to connect to analog DID
(Direct Inward Dialling) trunks?
Analog DID trunks are the opposite to non-DID analog lines and require
an FXS port (rather then the FXO used for non-DID analog lines).
Any hints or pointers much appreciated. A google search
I haven't checked in a few months but while the info below is correct the
102 limits the PC Lan port to 10mb even if using a 100mb NIC card.
Can anyone else confirm or deny this?
John
-
NetRom Internet Services
On Monday 06 October 2003 05:13 pm, Carlton J. O'Riley wrote:
Are there any plans to incorporate the running of Asterisk as a
non-root user into the current CVS? There is nothing in Asterisk
that requires root access as far as I know and this would solve the
vmail.cgi script permissions
That makes a lot of sense, but...it still doesn't work.
DOH! :-(
Hmm.. how are you connecting to the box? Zaptel device? SIP connection? I
wonder what audio format's being used?
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simply add...
..
my $AGI = new Asterisk::AGI;
my %input = $AGI-ReadParse(); ## this line
..
Joe Dennick wrote:
That makes a lot of sense, but...it still doesn't work.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Crick
Sent:
Hi All,
I have just compiled the newest version of mpg123 on a RedHat 9.0 system
(mpg321 has not been installed) and I am using the newest CVS version of
asterisk. Whenever I place any mp3 files in the
/var/lib/asterisk/mohmp3/ directory, Asterisk crashes a horrible fiery
death.
If mp3s
Like most others on this list I also have some really annoying echo whenever
a call goes out to the PSTN from a SIP phone...
SNOM/Budgettone - Asterisk - X100P - PSTN
I have tried every echo canceler in the makefile and turned on and off
aggressive suppressor etc. etc. etc. tried 32,16,128, and
I have a new echo can I'm working on, let me see if I can get it fixed
tonight and in CVS.
Mark
On Mon, 6 Oct 2003, Brian Schrock wrote:
Like most others on this list I also have some really annoying echo whenever
a call goes out to the PSTN from a SIP phone...
SNOM/Budgettone - Asterisk -
On Fri, 2003-10-03 at 14:53, Babak Pasdar wrote:
This issue was resolved by adding the @context in the voicemail.conf
file for the extension to the mailbox=XXX command.
[EMAIL PROTECTED]
Thanks so much for your help.
Is there anything special I need to configure on the Cisco phone
Leif Madsen wrote:
Hi All,
I have just compiled the newest version of mpg123 on a RedHat 9.0 system
(mpg321 has not been installed) and I am using the newest CVS version of
asterisk. Whenever I place any mp3 files in the
/var/lib/asterisk/mohmp3/ directory, Asterisk crashes a horrible fiery
Works fine on my 7960 with 5.3 firmware.
bkw
On Mon, 6 Oct 2003, Babak Pasdar wrote:
Hello,
I am trying to conference two or more calls on a Cisco 7940 phone. When I have one
inbound call and one outbound (I initiate the second call by pressing conference) I
get the join button at the
Hi,
I am having some trouble with ISDN Dialout. Using a Netjet-s PCI Card.
When in Minicom, the only way I can dialout is if i issue ATS18=1 First.
Otherwise I get a BUSY message. So thats fine.
But when I dialout from asterisk, I get an immediate hangup, so my guess is
that asterisk is not
--- sip.conf
[8991]
type=friend
username=8991
secret=
nat=no ; This phone may be natted
host=dynamic
canreinvite=no ; Cisco poops on reinvite sometimes
qualify=500 ; Qualify peer is no more than 200ms
I added the line suggested below, and now I hear the prompt for input,
but then nothing. The CLI says its playing the prompt, and nothing
more. When I finally end the call (hang up the phone)(BTW, I'm using
X-Ten's soft-phone for testing) Asterisk crashes and has to be
restarted.
-Original
The biggest feature we hope to offer, which we're going to call
Wireless InterCon, allows customers who opt for the service to expose
a local PSTN line for sharing with other members of the club. Because
we operate across many ILEC exchanges and two LATAs, we have the ability
to route
use
mailbox=500
instead of [EMAIL PROTECTED]
[EMAIL PROTECTED]
since he doesn't have his stuff in the default context
bkw
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http://lists.digium.com/mailman/listinfo/asterisk-users
Not familiar with it... You have a URL?
- Original Message -
From: Matteo Brancaleoni [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, October 05, 2003 4:52 PM
Subject: [Asterisk-Users] Re: DB virtualization for multiple database
support - Was Re: [Asterisk-Users] How to use
Use in your sip.conf:
[EMAIL PROTECTED]
You need to use a the @context with voicemail2
Kevin,
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Juan J.
Sierralta P.
Sent: Monday, October 06, 2003 6:58 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users]
Hey all,
I am in the middle of creating a new user wizard which will generate all
the .conf's the new Asterisk user will require to get themselves up and
running in Asterisk without having to touch a single configuration file.
This is what I have come up with as a rough draft. It is far from
On Tue, 2003-10-07 at 00:44, John Vozza wrote:
I haven't checked in a few months but while the info below is correct the
102 limits the PC Lan port to 10mb even if using a 100mb NIC card.
Can anyone else confirm or deny this?
Yes my system indicates a 10mb connection.
--
Dave Cotton [EMAIL
If u are using the latest CVS then i suppose that u don;t need to do
anything except configuring h323.conf for the ip address of the h323 gK.Or
u can also use oh323 channe driver available from www.inaccessnetwork.com .
The default sample works fine for test purposes.
Rgds
Manoj K Gupta
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