Hi,
-Original Message-
If your interested please let me know. Im gonna be putting up
a site for downloading if there is enough interest.
Aye! I'd love to have a look.
Best regards,
Florian
___
Asterisk-Users mailing list
[EMAIL
On Tue, May 25, 2004 at 05:53:55PM -0500, Roger spake thusly:
Thanks for the reply - I have version cell phone service. I did a work
around and called my cell phone via IAX2 as opposed to the zaptel
channels. This works and all 3 extensions ring w/ no problem.
I am having the exact same
--On Friday, May 28, 2004 2:57 pm -0400 Timothy R. McKee
[EMAIL PROTECTED] wrote:
My SIP users need to transmit the # key as part of data entry. Asterisk
intercepts and initates a transfer function. I'm almost positive I've
seen this discussed somewhere, but none of my searches are finding
Here is a macro I wrote to deal with sites that have a mixture of
extensions with DID numbers and extensions without DID numbers. This
macro is USA/Canada specific but could be adapted for the dial plans in
other countries. This macro uses features that are not in CVS -stable.
You must use CVS
Dear,
Kyle Hagan wrote:
We are writing a program using the manager for * for our receptionist to
use once the system go live. If anyone is interested in helping us with
testing please let me know.
We are designing it for a touch screen monitor for her to do transfers,
see whose on the phone
Fixed an error in the macro The line:
exten = external,1,GoToIf($[${LEN(${CALLERIDNUM})} = 4]?f,1)
should be changed to
exten = external,1,GoToIf($[${LEN(${CALLERIDNUM})} != 4]?f,1)
--Eric
On Sat, 2004-05-29 at 02:49, Eric Wieling wrote:
Here is a macro I wrote to deal with sites that
I was having trouble getting zaptel to startup. Thought I'd share my
experience since I saw others with the same issue and no solutions. I'm
running Fedora FC2. Symptoms were:
xtcfg -vv that says that it's unable to open master device /etc/dev/ctl
modprobe zaptel was also failing
Use r option in your Dial command.
- Original Message -
From: joachim [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, May 29, 2004 2:45 AM
Subject: Re: [Asterisk-Users] No ringing sound on GS phones
Make sure to use CVS-head and you'll get ringing.
At 23:51 28/05/2004, you
In article [EMAIL PROTECTED],
Mike Stupak [EMAIL PROTECTED] wrote:
I was having trouble getting zaptel to startup. Thought I'd share my
experience since I saw others with the same issue and no solutions. I'm
running Fedora FC2. Symptoms were:
xtcfg -vv that says that it's unable to
then run ldconfig or restart your machine...:)
W
- Original Message -
From: Sam Bingner [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, May 29, 2004 12:26 AM
Subject: RE: [Asterisk-Users] spandsp wont compile.
Add the path to it to /etc/ld.so.conf
-Original Message-
It seems the choppy (and almost unusable) audio in Head is only impacting
some cisco users, and since these problems are not impacting
the few that
can read code, use cisco phones, and are impacted, we're stuck with the
problem. The problem seems to be very evasive, however switching the
Hi.
cut
Obviously I chose the custom option when I setup the OS. Is there someway
to fix the zaptel code to not be so picky?
I think it expects the kernel source tree to match the running kernel.
If you had built a new kernel called 2.6.5-1.315custom and then booted from
it, you
On Sat, 2004-05-29 at 14:15 +0200, Brancaleoni Matteo wrote:
sure, that happens because kernel-source package from fedora
has the kernel version set to blahcustom, as long as many
other rh versions.
Mandrake does this as well. Great idea if you want to increase the noise
on a list.
--
Dave
CVS HEAD from about 1 week ago. TDM30P and call through Nufone. I was
talking and wanted to park the call and move to another phone to pick it up.
I hit #701 instead of #700 though -- after a pause, I got a fast busy and the
call was gone.
When I called the person back, she said that
Hi,
this is no bug. When you want to park a call just hit #700. Alison will then
tell you on which extension the call was parked.
To pick up this call just dial the announced extension (e.g. 701). When you
press #700 while in a call you connect this call to the call parked at this
extension and,
I got it to load BUT now i get when i try to load the module.
localhost*CLI load app_rxfax.so
localhost*CLI May 29 09:51:38 WARNING[1199209392]: loader.c:240
ast_load_resource: /usr/local/lib/libspandsp.so.0: undefined symbol:
TIFFDefaultStripSize
Unable to load module app_rxfax.so
May 29
/etc/ld.so.conf
/usr/X11R6/lib
/usr/lib/qt3/lib
/usr/local/libUnable to load module app_rxfax.so
May 29 09:51:38 WARNING[1199209392]: loader.c:240 ast_load_resource:
/usr/local/lib/libspandsp.so.0: undefined symbol: TIFFDefaultStripSize
/usr/local/lib/libtiff
/usr/lib/asterisk/modules
the mods
On Saturday 29 May 2004 09:28, FastJack wrote:
this is no bug. When you want to park a call just hit #700. Alison will
then tell you on which extension the call was parked.
To pick up this call just dial the announced extension (e.g. 701). When you
press #700 while in a call you connect this
Its a known bug I posted a patch eons back
but i can't find now with that fscking pathetic search on mantis
wish http://www.google.com/custom?sitesearch=bugs.digium.com
could work
ok Found it
http://bugs.digium.com/bug_view_page.php?bug_id=487
- Original Message -
From: Andrew
Or you can use ValetParking :P
bkw
- Original Message -
From: TC [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, May 29, 2004 8:33 AM
Subject: Re: [Asterisk-Users] transfer bug (#701 - remote party hears
alison, not me)
Its a known bug I posted a patch eons back
but i can't
Just setup *, got a developers kit FXO where the incoming/outgoing pstn is
plugged in. I've then got an IAXy that is plugged into a Philips DECT
phone. * is setup so that the [bell] section rings the phone - exten =
s,1,Dial(IAX2/myuser,30)
What's happening when someone calls my number is that
Has anyone successfully used Galaxy Voice with Asterisk?
I am getting the following SIP errors repeated whether it is or isn't
behind NAT.
May 29 12:17:11 WARNING[1142135600]: chan_sip.c:595 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED] for
seqno 104 (Critical Request)
May 29
First off they are not ERRORS they are NOTICE and WARNING.
bkw
- Original Message -
From: Kevin [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, May 29, 2004 10:26 AM
Subject: [Asterisk-Users] Galaxy Voice
Has anyone successfully used Galaxy Voice with Asterisk?
I am
Also I think someone posted a galaxy voice config example on the mailing
list a few weeks back.. have you searched google yet?
bkw
- Original Message -
From: Kevin [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, May 29, 2004 11:04 AM
Subject: RE: [Asterisk-Users] Galaxy Voice
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi!
I am having problems with the PlayTones application and VoIP softphones.
I have the following in my extensions.conf:
exten = 123,1,Answer
exten = 123,2,PlayTones(Busy)
exten = 123,3,Hangup
But when I connect with
In article [EMAIL PROTECTED],
Tony Hoyle [EMAIL PROTECTED] wrote:
Kevin Walsh wrote:
I downloaded the latest version of your patch, from your website, and
it works perfectly. I had waited until I had some time available
because I thought I'd have to play around with it for a while.
Hi!
Does anyone have a more clear beep tone for the voicemail?
Try Playtones():
http://www.voip-info.org/wiki-Asterisk+cmd+Playtones
Cheers, Philipp
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Hello,
I was planning to use the output of asterisk -rx show queues in a
script when I noticed that sometimes asterisk only outputs the first
line of the response. e.g:
debian:/# asterisk -rx zap show channels
Chan Extension Context Language MusicOnHold
debian:/# asterisk -rx zap
Tony Mountifield wrote:
Is that the X100P generically, including the X101P?
I think so (same driver)... they're just software modems really, so there's no
need for them to have that kind of detection on-chip.
I would include both algorithms - the line reversal one for the hardware
that can do
On Sat, 29 May 2004, Bartek Kania wrote:
I am having problems with the PlayTones application and VoIP softphones.
I have the following in my extensions.conf:
exten = 123,1,Answer
exten = 123,2,PlayTones(Busy)
exten = 123,3,Hangup
But when I connect with gnophone(IAX) or
Your most likely compiling against one tiff library version, but loading
up another...
Do a:
ldd app_rxfax.so
to see what tiff library it's compiled against,
and then also try to find all the places where libtiff is on your
machine and remove the incorrect one..
-Mark
-Original
Hi
I was planning to use the output of asterisk -rx show queues in a
script when I noticed that sometimes asterisk only outputs the first
line of the response. e.g:
why don't you use the manager interface?
it's much better...
Matteo
--
Brancaleoni Matteo [EMAIL PROTECTED]
Espia -
yep use
exten = 123,1,Answer
exten = 123,2,Busy
bkw
- Original Message -
From: Bartek Kania [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, May 29, 2004 12:04 PM
Subject: [Asterisk-Users] PlayTones problem
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi!
I am having
Yes, I did a search and have what I think is the correct configuration.
I did a google search and I didn't see much. I was successful in
getting it to work both inbound and outbound with the exception of the
notices and warnings.
The config I am using is:
[galaxyvoice]
nat=yes
port=5060
Known issue
bkw
- Original Message -
From: Julien Levi [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, May 29, 2004 12:12 PM
Subject: [Asterisk-Users] Odd behaviour with asterisk -rx
Hello,
I was planning to use the output of asterisk -rx show queues in a
script when I
Yes, it's a known issue on the bug tracker (#1110), but no solution has
been found to date, afaik.
Julien Levi wrote:
Hello,
I was planning to use the output of asterisk -rx show queues in a
script when I noticed that sometimes asterisk only outputs the first
line of the response. e.g:
Tony Hoyle wrote:
I'll probably do some tidying up (change ukcallerid to callerid=uk as
it's neater).. the zaptel side though is stable.
OK... now uses usecallerid=uk (or usecallerid=us for symmetry).
Tony
--
Te audire no possum. Musa sapientum fixa est in aure.
Tony Hoyle [EMAIL PROTECTED] Key
its not really a critical issue... wonder when someone will take the time
and fix it. :P
bkw
- Original Message -
From: Karl Brose [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, May 29, 2004 12:29 PM
Subject: Re: [Asterisk-Users] Odd behaviour with asterisk -rx
Yes, it's a
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Sat, 29 May 2004, Hermann Wecke wrote:
On Sat, 29 May 2004, Bartek Kania wrote:
I am having problems with the PlayTones application and VoIP softphones.
But when I connect with gnophone(IAX) or kphone(SIP) and dial 123 the call
just hangs up
Hi,
my VoIP provider routes my main phone number and all extensions to the
same sip account.
In the sip header of the invite message is the To: field that shows me
which extension the caller dialed ...
for extension 0:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
[...]
To: sip:[EMAIL PROTECTED]
for
At 7:33 AM -0700 on 5/26/04, Maveric wrote:
I've noticed that when i pass a wait in an exten = that it doesn't
allow for dtmf tone input. Also on another note i've noticed that
when using gotoif it will also cut the dtmf tones and drop the first
part if the gotoif is hit in the middle of
John Todd wrote:
At 7:33 AM -0700 on 5/26/04, Maveric wrote:
I've noticed that when i pass a wait in an exten = that it doesn't
allow for dtmf tone input. Also on another note i've noticed that
when using gotoif it will also cut the dtmf tones and drop the first
part if the gotoif is hit in
AJ Grinnell wrote:
Well, it worked for 1 call, but now I am back to getting half a ring from
the ATA and then nothing. I am only seeing one rtp packet recieved per call.
Any other ideas?
Does it work ok if canreinvite=no ?? To be honest I have not
experimented much with canreinvite. All our
Hi!
Excellent!!
Philipp
See near the bottom for the interesting bit :-)
OK, while composing this post I decided to write a perl program to read
a uLaw stream on standard input and create a suitable header, writing
the result to an output file.
It can be found at
On 29/05/2004 at 13:52 brian k. west wrote:
its not really a critical issue... wonder when someone will take the time
and fix it. :P
bkw
to you bkw_ .. it's actually quite important to some of us... a bit like DTMF callerid
:D
Andy
___
On 29/05/2004 at 19:16 Tony Hoyle wrote:
Me too - the current patch could also be used to do DTMF caller ID without
too
much work (there isn't a line reversal in the specs for that, you just
have to
look for valid digits).
I'll probably do some tidying up (change ukcallerid to callerid=uk as
Andy Powell wrote:
there's a bounty (although it's not much but it's better than a poke in the eye with a
sharp stick) for DTMF callerid (some of us have been bitching about it for ages)...
http://bugs.digium.com/bug_view_page.php?bug_id=0001265
I can't do it unfortunately, as I'm not in a
Accually you can issue the cli commands via manager and get full outputs!
(Most people dont know that)
bkw
- Original Message -
From: Andy Powell [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, May 29, 2004 2:59 PM
Subject: Re: [Asterisk-Users] Odd behaviour with asterisk -rx
On 29/05/2004 at 16:49 brian k. west wrote:
Accually you can issue the cli commands via manager and get full outputs!
(Most people dont know that)
bkw
yes you can, but you have to have blocking=yes ... and I'm still waiting for info
on what the implications of doing this are.. eg if the
Hi,
On 28-May-04, at 6:16 AM, Rich Adamson wrote:
I there a problem with CVS ? My card TDM04B does not want to answer
calls
on 2 ports. Strange.
Yes there is a problem. Pull an older copy of wcfxs.c in zaptel (from
about
5/24) and it will work again. Mark is aware of the problem.
Do you have a
Hi
Does anyone know of an algorythm to extract the country code from a number.
I understand that the country codes are of different length and there is no
fixed length of local area code or phone numbers.
I am sure there is a way, if not how to telephone switches handle them
Umar.
Hello Everybody...
probably this is an FAQ item but I can't find it anywhere so here it goes.
I am trying to compile the latest CVS release of Asterisk under Suse 9.0
zaptel and libpri compile without any problems but when I do a 'make
install' in asterisk, it compiles ok until it reaches
the
On Sat, 2004-05-29 at 18:01, Vassilis Konstantinou wrote:
I am trying to compile the latest CVS release of Asterisk under Suse 9.0
zaptel and libpri compile without any problems but when I do a 'make
install' in asterisk, it compiles ok until it reaches
the cdr directory.
There I get a
Yes there is a problem. Pull an older copy of wcfxs.c in zaptel (from
about
5/24) and it will work again. Mark is aware of the problem.
Do you have a bug# we can track?
I'm about to deploy this card in a production env - I was going to pull
the latest drivers, but if bug is still
Title: iConnectHere broken?
I moved from 0.7 to Stable recently and my dialing out/in from iConnectHere stopped working. I made no changes to my conf files or anything else (other than the upgrade). Now I get a Unsupported Media error even though I'm still using ulaw and alaw.
It stopped
Hi all,
Anybody try to configure Asterisk and Digium card on linux-2.6.5-1.358
FEDORA CORE2 ?
Making the zaptel getting error: storage size.
Thanks,
Oliver
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Is the webmin module in the download directory of asterisk on site still
maintained. I can't get it to install with the latest webmin - 1.140, it
says it can't find the module.info file in the webmin module.
Thanks,
Nicholas Ruddick
___
Asterisk-Users
I moved from 0.7 to Stable recently and my dialing out/in from iConnectHere stopped
working. I
made no changes to my conf files or anything
else (other than the upgrade). Now I get a Unsupported Media error even though I'm
still
using ulaw and alaw.
It stopped working with my softphone,
is there a webmin add in and where I would like to test it...
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
search google... rgagon posted something to -dev that does just this a few
months back.
bkw
- Original Message -
From: usedcanon [EMAIL PROTECTED]
To: Asterisk users [EMAIL PROTECTED]
Sent: Saturday, May 29, 2004 5:01 PM
Subject: [Asterisk-Users] extracting country code from a number
Firstly we've setup a SIP proxy that uses e164.org to do enum lookups,
also rather then issuing people with yet more numbers they have to
remember we've coded up a watered down version of e164.org for people
that would just like to have a single SIP phone rather then run their
own PABX.
I would love to test it, but I need some help with the zapata config
settings. In the US you can have up to 4 numbers on a line each with a
ring pattern. The ring patterns are:
Long ring - 1st number (this is a normal ring - like if you didn't have
distinctive ringing at all)
Double ring - 2nd
I run 2 small call centers, would love to help
Zac
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kyle Hagan
Sent: Friday, May 28, 2004 11:33 AM
To: Asterisk
Subject: [Asterisk-Users] Asterisk Receptionist manager program.
We are writing a program
How do I get the lights to work correctly on a SNOM 200 when I configure it
for more than one line? The lights stay on solid, although the buttons work
correctly for making calls. Thanks in advance.
Dennis Engdahl
SnowCrest, Inc.
www.snowcrest.net
64 matches
Mail list logo
