Something the user list in
Microsoft Messenger. I was thinking on some sort of web page that can
check the registration of the sip clients on the asterisk but want to
know if already exist to avoid to reinvent the wheel.
That is actually quite easy and there are some projects that achive this
All,
I am very new to pbx hardware and equipment and any help will be greatly
appreciated. I am now the proud owner of a TDM422p and Iaxy/S100I. The server is
running debian testing so I first installed the asterisk deb package. To get the
zap modules, I compiled zaptel-1.0-RC2. After some
Greetings All,
I have a new post on the blog. It goes a little bit more in depth on
wcfxo.c and touches on zaptel.c. Two more screen shots. Loads of fun.
Take a look: http://zapteldoc.blogspot.com
Regards,
Victor
___
Asterisk-Users mailing list
[EMAIL
traceroute A - B:
traceroute to 192.168.2.44 (192.168.2.44), 30 hops max, 38 byte packets
1 192.168.11.1 (192.168.11.1) 1.964 ms 1.181 ms 0.852 ms
2 10.138.3.2 (10.138.3.2) 43.428 ms 49.634 ms 47.601 ms
3 192.168.2.44 (192.168.2.44) 53.440 ms 49.320 ms 48.968 ms
traceroute B - A:
[EMAIL PROTECTED] wrote:
Problem was with asterisk.. Mark had made a change in chan_sip.c
that affected noncodec capabilities, it's been fixed.
Do you have a bug number? Or something else to find it in the bug database?
--
Andreas SikkemaRits tele.com
Scheepmakersstraat 11
On Thu, 9 Sep 2004, Karl Brose wrote:
In order to dial out to a sip provider, you need to configure that
provider in your sip.conf file as a peer with your proper username
and secret, etc.
Cool! Just found that in the handbook too a second or two ago :-)
Thanks for taking
On Thu, 9 Sep 2004, hank smith wrote:
I have a SiS 900 PCI Fast Ethernet Adapter what do I put in there or is that
what I put in the xml file?
Go read: http://www.colinux.org/wiki/index.php/coLinuxNetworking
Specifically:
If in doubt, the name of the card can be found in colinux-daemon
On Thu, 9 Sep 2004, hank smith wrote:
is there going to be a gui for co linux and astwind?
No. AstWind is just a Debian GNU Linux distribution with a precompiled
Asterisk installation running under a CoLinux kernel.
I will have to see if either there is going to be a gui or if yasr a screen
--- hank smith [EMAIL PROTECTED] a écrit :
when you get this up up can you give the phone
number?
Ok, I just start the project it's for a local
televisoin in french polynésia TNTV.
I hope that this project will be concretized.
this sounds rather interesting, and fun!!!
- Original
--- William Suffill [EMAIL PROTECTED] a
écrit :
Sounds like it be best as a custom app or AGI
depending how many calls
you will be taking and how bad the performance hit
of using an AGI vs
Compiled app is for your needs
OK, I first try with AGI which sound like quicker to
implement. And if
yasr is text based but the interesting part is going to see if it works
running on a windows platform with this version of linux with out that I
can't do anything with this so I will have to see. take care.
hank
- Original Message -
From: Greg Boehnlein [EMAIL PROTECTED]
To: Asterisk
On 9 Sep 2004 at 23:24, Victor Rini wrote:
Greetings All,
I have a new post on the blog. It goes a little bit more in depth on
wcfxo.c and touches on zaptel.c. Two more screen shots. Loads of fun.
Take a look: http://zapteldoc.blogspot.com
Regards,
Victor
Keep up the good work!
And
On Tue, 2004-09-07 at 21:48, Glenn A. Thompson wrote:
Hi,
I must be blind, how does one check then act upon the return code from
the previous command?
For instance, Answer says it can return non zero. How do I check for
that. It doesn't set any other variables like Dial does.
Most
Hi,
I tried to make a call to extension 2001 with the setting [EMAIL PROTECTED] (Detailed:
exten = _7.,2,Dial(SIP/[EMAIL PROTECTED]/${EXTEN:1})
which does not work at all - i always get the failure message: No such host provider.com/2001 (the number i dialed) - why ??
when i try the same
Leo wrote:
Not necessarily so. Recently I discovered that Artisoft's Televantage
Soft PBX can support Toshiba Strata CS digital phones (DKT 2000 and
3000) through a PCI 16-port digital station card (Toshiba part
#CS-DKTU-TV). Apparently, the Strata CS is an OEM licensed version of
it works it works it works! sorry it took it so long for the info to
click thanks for the help guys!!!
take care
hank
- Original Message -
From: Greg Boehnlein [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Thursday, September
Hi,
I would like to realize a voip testbed that should simulate the scenario
in which two companies have an asterisk PBX connected through a PRI-ISDN
to the Telco operator.
I have no experience of T1/E1 connection but I think that the above
could be relized with 3 asterisk boxes equipped with
You need an E1 back-to-back cable.
Regards,
antonio
Francesco Delfino wrote:
Hi,
I would like to realize a voip testbed that should simulate the scenario
in which two companies have an asterisk PBX connected through a PRI-ISDN
to the Telco operator.
I have no experience of T1/E1 connection but I
Maciej Kietlinski wrote:
Are the FXOs on the 2x on ports 1-2 or 3-4? Maybe it has to
do with *any* FXO on port 1...
Please get back with the list with your findings.
My experience led to a replacement from Digium, but the card is a
TDM400P with 4 FXO...now that I think of it, during
I always just let the phone poll the Snom update server for updates but
while the server is back at version 2.03o the latest stable downloadable
version on the website is 2.04n..
Is Snom not distributing updates for the 200 from their server anymore??
On Fri, 10 Sep 2004, Johannes Hollerer wrote:
I tried to make a call to extension 2001 with the setting
[EMAIL PROTECTED] (Detailed: exten =
_7.,2,Dial(SIP/[EMAIL PROTECTED]/${EXTEN:1}) which does not work at
all - i always get the failure message: No such host
On Fri, 10 Sep 2004, Francesco Delfino wrote:
[...]One of the box will represent the Telco, the other two, the two
companies PBX. I would like to know if it is needed something
between the point-point connections or it is possible to just
cross-connect them.
As more
WipeOut schrieb:
I always just let the phone poll the Snom update server for updates but
while the server is back at version 2.03o the latest stable downloadable
version on the website is 2.04n..
Is Snom not distributing updates for the 200 from their server anymore??
Have a look here:
Do you have a bug number? Or something else to find
it in the bug database?
bug #2394
Seems, the minor issue with Non-codec capabilities
in sip debug still exists.
Arsen.
__
Do you Yahoo!?
Yahoo! Mail - You care about security. So do we.
Hi.
After a small war with underfined sybol error and conflicts between h323
and oh323 I successfully install h323 channel.
Now, I can connect from Netmeeting to SIP and ZAP channels, but I can't here
anything.
When I call at phone, and try to speak, on another end of line man said,
that my
Bastian Schern wrote:
WipeOut schrieb:
I always just let the phone poll the Snom update server for updates
but while the server is back at version 2.03o the latest stable
downloadable version on the website is 2.04n..
Is Snom not distributing updates for the 200 from their server anymore??
Hi, I'm new to telephony Software and Hardware, so please excuse my
questioning.
I plan to set up a little system, using Asterisk and VoDSL via Belcacom
or Scarlet here in belgium.
We are yust a little 2 man company and we are not always in our office.
My idea is, to get VoDSL and set up a
[EMAIL PROTECTED] wrote:
Hi, I'm new to telephony Software and Hardware, so please excuse my
questioning.
I plan to set up a little system, using Asterisk and VoDSL via
Belcacom or Scarlet here in belgium.
We are yust a little 2 man company and we are not always in our
office.
My idea is,
Hey Victor, that's really lot of fun!
I'm anxious for the next chapters!
Renato
On Fri, 10 Sep 2004 19:27:24 +1200, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
On 9 Sep 2004 at 23:24, Victor Rini wrote:
Greetings All,
I have a new post on the blog. It goes a little bit more in depth on
Hi all,
I'm sorry, but I'm stupid because I haven't load res_parking.so.
Regards,
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Sergio
Serrano
Enviado el: viernes, 10 de septiembre de 2004 9:35
Para: 'Asterisk Users Mailing List -
Are the FXOs on the 2x on ports 1-2 or 3-4? Maybe it has to
do with *any* FXO on port 1...
Please get back with the list with your findings.
My experience led to a replacement from Digium, but the card is a
TDM400P with 4 FXO...now that I think of it, during troubleshooting
Helloall!
I am trying to load sip.conf from mysql database. I have followed the
instructions at http://www.voip-info.org/wiki-Asterisk+sip+mysql+peers. Seems that the authentication (user psw) works fine but I would
like to get more information from mysql and I don't know how to retrieve
I'm sorry, but I'm stupid because I haven't load res_parking.so.
And you reply to a different discussion thread. Don't use reply if you
don't want to reply, create a new message instead.
Hint: res_parking was renamed into res_features
Hint2: get rid of /usr/lib/asterisk/modules and do
On 09/09/2004 at 18:48 Josh Roberson wrote:
I wrote cepstral regarding this at the beginning of the week, thought it
might be relevant to post the reply:
Thanks for contacting us. Our Linux package is off the site right now
because we are releasing a new version, 3.02, next week. This is an
On 07/09/2004 at 23:57 Benjamin on Asterisk Mailing Lists wrote:
On Tue, 07 Sep 2004 08:14:57 -0500, Brian Capouch [EMAIL PROTECTED]
wrote:
If you have a Linux laptop with you, then in fact the SIP devices can be
configured to hide behind it. The laptop can then run an instance of
asterisk
Hi,
I have asterisk,openh323-v1_13_5 and pwlib-v1_6_6 installed on my PC. each time
i run asterisk -c, i get the following error:
[chan_oh323.so] = (OpenH323 Channel Driver)
== Parsing '/etc/asterisk/rtp.conf': Found
== Parsing '/etc/asterisk/oh323.conf': Found
I have built latest
Asterisk w/ OpenH323 channel driver. We have a SIP softphone registered to the
Asterisk. We can place outbound calls from the SIP phone to the PSTN via
OpenH323 connection to our gatekeeper. Everything works okay - DTMF and
Audio...
But in the reverse
- if we call from
Have you configured;
_ sip.conf_
..add this line:
dtmfmode=inband
..also you have uncomment the right line that
matches your dhcp setup:
localnet=192.168.0.0/255.255.0.0; All RFC 1918
addresses are local networks
;localnet=10.0.0.0/255.0.0.0 ; Also
RFC1918;localnet=172.16.0.0/12 ;
AFAIK, one can just send SMS via smsbug.com, but I want to be able to
receive sms, without using an external sms-gateway wich should work with
the sms-applikation if sms-ing is supported by VoDSL.
Greetings, Sascha
Am Fr, den 10.09.2004 schrieb Thorsten Neumann um 14:35:
I have come across an
We use a nice Polycom conference phone and plugged it into the Sipura
and it works crystal clear. Was cheaper than Polycom's conference phone
w/ built in VOIP capabilities.
Joe Dennick wrote:
If it were me; I'd opt for one of the Polycom Conference phones (they
are just regular analog phones),
Hi everyone.
I'm a bit of a Linux newbie, but I've been doing tech stuff for ages.
I'm also brand new to *.
I've been reading the Voip.org wiki, and perusing the list archives for a
while since I've been asked to investigate using IP telephone / soft phones
for a call-center type scenario. People
Have you had chance to look at Jeff Pulver's Communicator? This is a
soft-phone, currently in beta, that allows you to bring together your
contacts from MSN, ICQ, AOL and, importantly from your point of view, add
contacts that are SIP users.
I've not tried it yet with asterisk, but now you have
On Fri, 10 Sep 2004 14:05:09 +0200, Andy Powell
[EMAIL PROTECTED] wrote:
At the risk of stating the obvious if you have a laptop not running MacOSX (ie
perhaps running windows) download my asterisk live! cd (
http://www.automated.it/asterisk/ ), burn it and test it on your laptop and bung
Try this :)
?xml version=1.0 encoding=UTF-8?
colinux
!-- This line needs to point to your root file system.
For example change root_fs to the name of the Debian image.
Inside coLinux it will be /dev/cobd0 --
block_device index=0 path=\DosDevices\c:\program
I run Asterisk on Redhat 8.0 with a VM hosted by Microsoft's Virtual PC
which, in turn, runs on Windows 2000 Server. Works like a charm. Can't use
Zaptel cards but that's OK for me. I can put it into standby any time and
it takes only a few seconds to start up the VM from its saved state and at
[EMAIL PROTECTED] wrote:
Hi,
I have asterisk,openh323-v1_13_5 and pwlib-v1_6_6 installed on my PC. each time
i run asterisk -c, i get the following error:
[chan_oh323.so] = (OpenH323 Channel Driver)
== Parsing '/etc/asterisk/rtp.conf': Found
== Parsing '/etc/asterisk/oh323.conf': Found
What alarm is it. Is it red or is it yellow.
If it's red then it's the /etc/zaptel config
But if it's yellow then it's a problem with sync the channels
Which could be a master - slave problem.
Very easy to fix.
Sean
-Original Message-
From: Jan Goericke [mailto:[EMAIL PROTECTED]
Hi John,
I'm also new to *, but if you want to set up a callcenter, with 40
people calling the same number at the same time, you probalbly will need
a T-1 or E1 line wich AFAIK handles at least 30-calls.
You then need at least one Digium E1/T1 card to get the calls into * and
other cards to
For whom which may be interested:
Here in Italy we have GSM #numbers without leading zero
PSTN instead has prefix starting with '0'
to have '0' recognized by * i need to insert
nationalprefix=0
as Jason Williams suggested me in irc;
now, you cannot have:
pridialplan=natonal
otherwise * will
what phone did you purchase and how much
- Original Message -
From: Deon Rodden [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion [EMAIL PROTECTED]
Sent: Friday, September 10, 2004 5:59 AM
Subject: Re: [Asterisk-Users] Conference Phone
We
Hi,
I'm unable to pick up parked calls after they are transfered.
I get the transfer message when I press # and then I'm told 701 The
extension I'm dialing goes to the on hold music. I'm disconnected, I hang
up, dial 701 and I see this message on the console Everyone is
busy/congested at this
Can anyone who is using this, give me an idea of performance impact of
using this?
Thank You,
Matt Pusateri
On Thu, 9 Sep 2004 20:00:02 -0500 (CDT), Lenny Tropiano / asterisk.org
Mailing list [EMAIL PROTECTED] wrote:
Did I see something on here about using an AGI script to do reverse
Thanks. It seems like I do not have much of a choice left.
Anyway, I just found out that if the usecallerid=no in zapata.conf. Asterisk
does not wait for 2 rings before processing the call.
On Thu, 2004-09-09 at 06:38, HengWee Chin wrote:
I am wondering if there is any way or settings I can
The 'parkedcalls' code dynamically creates and deletes entries in the
dialplan to handle the calls that have been parked, so the parking lot must
not overlap your regular extensions. The initial parking extension is
statically created on startup, thus the 'exten =' entry is matching the
parking
What more information? Are you talking about mailbox, nat, etc..all those
other options for SIP phones? I want to do SIP from database as well but
most of our phones are NAT and need that option stored in the database.
Matthew
- Original Message -
From: Victor Alvarez [EMAIL PROTECTED]
That fixed it. Thanks
The 'parkedcalls' code dynamically creates and deletes entries in the
dialplan to handle the calls that have been parked, so the parking lot
must
not overlap your regular extensions. The initial parking extension is
statically created on startup, thus the 'exten =' entry
Don't remember our costs exactly, was almost a year ago. But this would
work for you:
Polycom Soundstation - $110
http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemcategory=41374item=6322819848rd=1
Spira SPA-1000 - $85
Hello,
On Fri, 10 Sep 2004 09:49:43 -0500, Matthew Boehm [EMAIL PROTECTED] wrote:
What more information? Are you talking about mailbox, nat, etc..all those
other options for SIP phones? I want to do SIP from database as well but
most of our phones are NAT and need that option stored in the
Hi Guys,Im having some problems with a Wildcard
TE410P card.. During a call I getsome strange messages and the voice drops
out:Aug 24 16:40:17 DEBUG[1101416512]: chan_zap.c:4036 my_zt_write:
Writereturned -1 (Resource temporarily unavailable) on channel 1Aug 24
16:40:17 DEBUG[1101416512]:
Victor Alvarez wrote:
I am trying to load sip.conf from mysql database. I have followed the
instructions at
_http://www.voip-info.org/wiki-Asterisk+sip+mysql+peers_. Seems that
the authentication (user psw) works fine but I would like to get more
information from mysql and I don't know how to
should this work with the x101p? or just the tdm400?
Thanks for your help
Robb
Edward Eastman wrote:
Brilliant - thanks, took me half an hour but it's working now.
Just for the record, settings as follows:
The patch on http://bugs.digium.com/bug_view_page.php?bug_id=009
What versions of Asterisk, asterissk-oh323 do you use?
What is the current configuration of oh323?
Can you send the backtrace of the core file dumped?
Michael.
Asterisk CVS-HEAD-08/26/04-11:46:11
asterisk-oh323-0.6.3b
I think it should be the openh323 because i got the same
how much ram you got on the pc running the vm? also will microsoft Virtual
PC run on xp home?
thanks
hank
- Original Message -
From: Bill Seddon [EMAIL PROTECTED]
To: 'Benjamin on Asterisk Mailing Lists' [EMAIL PROTECTED];
'Asterisk Users Mailing List - Non-Commercial Discussion'
hmm really need to test this thing.
On Fri, 2004-09-10 at 10:02, Greg Boehnlein wrote:
On 9 Sep 2004, khurram bhatti wrote:
Well I wanted to test astwind and consulted * person
he gave me this comment
lord help us all ... why would you want to simulate a linux system on
top of a
Date: Fri, 10 Sep 2004 16:37:33 +0300
From: Michael Manousos [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Problems with 0penh323 Channel Driver
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain;
Hi!
Net2Phone is getting a common SIP status
code, 404 Not Found, when trying to place a call to our Asterisk
server. We're hoping someone on the list can shed some light on why
this is happening. We can process a call from Asterisk to Net2Phone
without any problems.
Net2Phone sends the
Hi,
First of all thank you Matthew, Nicolas and
Ryan for your response.
I would like to get information like context,
mailbox, callgroup, pickupgroup, codecs... also nat! If I make the substitution
of the text file i wouldn't like to miss information in the
process.
I posted a while ago, about the FXO card entering a non-operational
state. While in a call, all of a sudden, there's this loud noise, and
the card remains like that until I reload the wcfxo module. There's no
way to dial in or out the FXO unless the module is reloaded.
I made some progress... I
Maybe this is silly but I had a similar problem when I installed my
kit. The problem was my motherboard didn't provide 3.3V...
Rgds,
Renato
On Fri, 10 Sep 2004 13:16:26 +1200, Colin Haxton [EMAIL PROTECTED] wrote:
Hi Lyle,
I don't have lspci on my system. It's a dump of what is in the
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Marconi Rivello
Sent: Friday, September 10, 2004 1:47 PM
To: Asterisk
Subject: [Asterisk-Users] Red Alarm
I made some progress... I was looking for an indication in the system
that
Apparently, Plan A is hard coded to only select out certain info from the
database. If you know C you could probably take a crack at adding some more
code. This is what I am going to do here in a bit or over the weekend.
Matthew
- Original Message -
From: Victor Alvarez [EMAIL PROTECTED]
It sounds like my lockups may be related since my TDM422b card has the FXS FXS FXO
FXO configuration and doesn't have an FXO in position 1 either.
My card is identified in software as Rev E/F and has the wire jumper on the back.
David
Richard Scobie said:
Maciej Kietlinski wrote:
Are the
The ast-UK-and-DTMF-pol-CID.diff patch will only work for the tdm400 FXO
modules due to the fact that the x101p is not capable of detecting
polarity reversal events.
Dan
On Fri, 2004-09-10 at 17:38, Robert Boardman wrote:
should this work with the x101p? or just the tdm400?
Thanks for your
I need to debug a call quality issue with remote users on the other
end of a satellite link. The symptoms are: we here on the Internet
side can hear them just fine. On their end, things work sorta OK most
times, but they often suffer from severe dropouts and digital
warbling, both of which I
did you try to add
canreinvite=yes
to
[net2phone3]
??
Marc
[EMAIL PROTECTED] wrote:
Hi!
Net2Phone is getting a common SIP status code, 404 Not Found, when
trying to place a call to our Asterisk server. We're hoping someone on
the list can shed some light on why this is happening. We can
I've recently started playing with the RxFax application on my
Asterisk box. I've had success, mostly, but I've had some failures,
too...
The most recent failure is specific to receiving from a particular fax
machine -- a Canon Laser Class 9000S. The TIF images received are
readable, but the
thanks for the reply Dan
Does anyone know if the history buffer CID patch still works with the
latest cvs?
Robb
Dan Tucny wrote:
The ast-UK-and-DTMF-pol-CID.diff patch will only work for the tdm400 FXO
modules due to the fact that the x101p is not capable of detecting
polarity reversal events.
On 9 Sep 2004 at 15:35, [EMAIL PROTECTED] wrote:
I am using CVS-HEAD-08/29/04-22:41:39
I have notransfer=yes in my iax.conf
I have been on the phone most of the day...dropped twice so far.
Paul Seniuk
-Original Message-
From: Kris.Boutilier [mailto:[EMAIL PROTECTED]
Marcello Lupo wrote:
Hi to all,
we have a community of people on an * box that use SIP softphones to talk each
other. Can you suggest me the quickest and simple way to let someone know who
is online without have to call one by one the persons to look if they are
present or not?? Something the
Has any one put 3 or more TDM405P or TDM410P cards in a single server?
I would like to fit as many as 6 into one box.
I am concerned about several things such as power requirements and the
amount of cooling as well as CPU and memory utilization.
Is there a difference in the power consumption and
From what I have heard, read and seen, the most you will ever want to
do is two, and that is only in certain situations, i.e. you are not
doing much/any transcoding, IVR's, a bunch of conferences, etc. A
better solution would be multiple 1U servers, potentially, even though
I realize, space is
When we need that many T1s, we use routers. Much less complex and roughly
the same cost.
William
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Friday, September 10, 2004 12:40 PM
To: 'Asterisk Users Mailing List -
I would think the first issue regarding the number of cards is that each
card has to have a seperate and unique IRQ and cann't share IRQ's with
anything else. So from that requirement, six would seem out of the
question.
As far as the rest, there are limits on number of calls, but they are more
Matt,
That interesting. We have even had the problem occur with SIP clients
Using SNOM handsets. The gateway has a PRI, so I dont think busycount
Even applies too me?
Cheers,
Paul Seniuk
-Original Message-
From: matt.riddell [mailto:[EMAIL PROTECTED]
Sent: September 10, 2004
Hey All,
I have a question that I'm curious about. I want to
set up a 4 phone system in my home with 2 actual lines
coming into the house. Both or just regular lines
(not sure of this matters?), one being VoIP and the
other just a regular analog line. For now though I
just want the VoIP line
Benedict P. Barszcz wrote:
Can I use this card with asterisk in any way but without subscription to
a Frame Relay account? Perhaps in similiar manner as T1/E1 between a
channel bank and an asterisk server. Or perhaps there is way to make it
behave like a kind of an FXS interface (to anything).
When sending calls to my Long Distance Provider I've
come across this problem.
After about 3 or 4 seconds into a call, it gets cut off.
This is what I have concluded after doing a trace.
1. An invite is sent to the Asterix PBX
2. Asterix sends back a 100 trying.
3. Asterix then sends a 200
Hello,
MOH always is choppy when someone calls from a cell
phone to my pots or nufone 866. It sounds fine when
it originates from a land line. I use zaptel
hardware, and plenty of resources.
I have tried to use different songs. None have the
id3 tags, I tried the custom settings with -q -r
First, START NEW MESSAGES. don't respond to something totally different
and then remove the contents. You message has NOTHING to do with the
message your mail client said you responded to.
In-Reply-To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] 7 Dialing gives a busy signal
On Fri,
On Sat, 11 Sep 2004 [EMAIL PROTECTED] wrote:
I posted to the -dev list the other night (although I was a little
drunk) about whether the busydetect code recognizes the cadences as
well as the tone. Reason being that there are definitely not 6 x
busy length tones being played that would
Does anyone know what Q.931 Information Element that * pulls the RDNIS
variable from?
Jody N. Rudolph
Heartland Communications Internet Services, Inc
1301 Boadway
Paducah, KY 42001
[EMAIL PROTECTED]
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Steve,
Are you for real about the voice pitch?
I am both laughing and fascinated at the same time!?!?! :P
Paul Seniuk
-Original Message-
From: steve [mailto:[EMAIL PROTECTED]
Sent: September 10, 2004 2:36 PM
To: asterisk-users
Subject: RE: [Asterisk-Users] IAX2 dropping call?
I love the functionality of the Valet Park Application. I have a
question regarding its operation. The problem I am having when there is
a call already parked on specific park extension. If a caller uses
'blind' transfer on a Cisco Phone the caller gets disconnected. Can any
offer any
[snip]
From Digiums site, you get this
The Wildcard TDM400P is a half-length PCI 2.2 compliant card that
supports from one to four telephone interfaces for connecting analog
telephones or analog lines to a PC.
...
The naming convention for the TDM bundles is
On Fri, 10 Sep 2004 [EMAIL PROTECTED] wrote:
Steve,
Are you for real about the voice pitch?
I am both laughing and fascinated at the same time!?!?! :P
Paul Seniuk
Yeah - I'm quite serious. I was trying to get busydetection working for
the UK, so I had loads of debugging in the
I can't seem to get *80 to do its thing on a Zap channel. Looks like
*8 is being seen by asterisk first, and *80 is basically inaccessible.
What *80 is intended to do, by the documentation on the wiki and by
inspection of the source code, is add the last callerid to the
blacklist.
Looking at
I will not be using all of the T1s for voice. I will be using a
combination of voice and data and I don't expect that all of the lines
will ever be full.
Since the people how answered only recommend 1 TE4**P card (thanks
Steven) in a box I imagine that the solution is to setup peering between
No it doesn't/shouldn't.. If a call is already parked in that location you
shouldn't be able to complete the transfer and you'll have to press resume
and try again.
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Kevin
Sent:
Hi there,
So I've finally got our Definity and * box talking back and forth, but
can't figure out how get callerid sent from the Definity to *.
Has anyone had any success with this? I've tried every combination of
zapata.conf variables pertaining to callerid with the same results:
Accepting call
On 10 Sep 2004 at 22:51, [EMAIL PROTECTED] wrote:
On Fri, 10 Sep 2004 [EMAIL PROTECTED] wrote:
Steve,
Are you for real about the voice pitch?
I am both laughing and fascinated at the same time!?!?! :P
Paul Seniuk
Yeah - I'm quite serious. I was trying to get
1 - 100 of 134 matches
Mail list logo
