So, if zaptel will not read codes from my indications conf file, what
would be a suitable solution to feed it my country tones?
Is there a list of loadzone/defaultzone country codes included in
zonedata.c? Can this list be updated to include my country? (I have
the tones)
Thanks for the help
On Wed, 17 Nov 2004, Steve Underwood wrote:
It is working pretty well. I think it will be available about the end of the
year. I will not be free. It will be supplied with a commercially licenced
Asterisk.
Hi everybody,
I would like to know a rather basic thing: What do you use SS7 for ?
I
modify Makefile set BUSYDETECT_TONEONLY
But Asterisk can not detect busytone
I find that Asterisk can not detect busytone,
need I modify dsp.c?
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Peter Svensson wrote:
On Wed, 17 Nov 2004, Matt Riddell wrote:
Peter Svensson wrote:
I guess you just have to know that Brian is a bit trigger happy sometimes.
It has it's ups and downs. Things get fixed quickly, but sometimes his
instinct is wrong.
I was beginning to think he wasn't human.
Minimum P-300, PCI 2.2 is the recommendation, but how does the real world
works?
How fast should be the CPU if I have xx functions ???
How much RAM should I use for xx functions ???
How much hard disk should I reserver for xx functions ???
I did not write the functions, but can we make a
I think the wiki has most of this covered. Just requires a little
reading and investigation.
http://www.voip-info.org/tiki-index.php?page=Asterisk%20dimensioning
I really think it's going to be impossible to account for every
variable in asterisk. There's just too many. Okay so we document XX
In simple terms SS7 is the suite of protocols used by all major networks for
interconnecting their 'real' telephone exchanges. There are many telecomms
platforms which only support SS7 if you want to connect into them you will
need SS7 support. Note that SS7 is not limited to 'call' data, there
If you can't update with SQL commands from the CLI then you need to check
your permissions in database mysql.
Read Mysql docs.
info mysql
MySQL Database Administration - Privilage System
Br /Kev/
- Original Message -
From: Shaun Tierney [EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Hello There,
Have anyone had issues with caller ID on an outgooing ISDN line in Cyprus?
Best regards
Jorn
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I'm having problems with an FXS module on a TDM400P card with
asterisk telling me:
-- Zap/4-1 is ringing
Exception on 19, channel 4
Got event Ringer On(10) on channel 4 (index 0
Exception on 19, channel 4
Got event Ringer Off(11) on channel 4 (index 0)
Didn't finish Caller-ID spill.
Hi,
Thank you, it's working now!
Do you think that this patch will be included in the next cvs versions?
Sincerely,
Cyrille Demaret
- Original Message -
From: Karl Brose [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Tuesday,
Hi all,
I set up a IS64PHHiSax: IS64PH Hisax compatible card to terminate sip
calls to PSTN. Everything works as expected but I can get no ringing tone
when the call attempt is in progress. As soon as the called party
answers everything works again. I've tried all three options for mode
I keep getting console messages like the following:
Nov 17 11:22:33 NOTICE[16767]: channel.c:1320 ast_read: Dropping
incompatible voice frame on SIP/sipuser-5240 of format ALAW since our
native format has changed to ULAW
I think this is causing some incoming queue calls to ring on an agents
Hi Steve,
It will support ISUP v2 or it will be MAP or just MTP1, MTP2,MTP3 ?
I'm very interested in ISUP for the moment... but if it will support MAP as
well .. great !
Thanks,
Angel.
--
Message: 5
Date: Wed, 17 Nov 2004 09:05:21 +0800
From: Steve
Lex Lethol wrote:
So, if zaptel will not read codes from my indications conf file, what
would be a suitable solution to feed it my country tones?
Is there a list of loadzone/defaultzone country codes included in
zonedata.c? Can this list be updated to include my country? (I have
the tones)
Hi there,
hopefully I'm not too off-topic ...
I registered my first PSTN number in the e164.arpa service. I tested my *
setup in a testbed and I think it should work correctly.
Could probably somebody who is already successfully using ENUM lookups
volunteerfor a short test? And contact me for
Hi,
Trying to update to spandsp-0.0.2-pre6 I got a compile error:
Making all in src
make[1]: Entering directory `/mnt/geracaodecd/rpm/BUILD/spandsp-0.0.2/src'
make all-am
make[2]: Entering directory `/mnt/geracaodecd/rpm/BUILD/spandsp-0.0.2/src'
source='t31.c' object='t31.lo' libtool=yes \
Marty Lee wrote:
I'm having problems with an FXS module on a TDM400P card with
asterisk telling me:
-- Zap/4-1 is ringing
Exception on 19, channel 4
Got event Ringer On(10) on channel 4 (index 0
Exception on 19, channel 4
Got event Ringer Off(11) on channel 4 (index 0)
Didn't finish
So, that's how my tax dollars are spent? Outrageous, and certainly
news-worthy. Good luck fighting off CNN and the like when this leaks
out.
Not at all, this is one of my favorite policies that has come from the
performance improvement department. Yes that is right, it is official
policy
Good time of day to all russan speaking world! :)
I would like to announce that we started a non-commercial project the
goal of which is promoting Asterisk on ex-USSR space and supporting
asterisk based solutions. We are also starting development projects.
If you speak russian and deal with
You might want to take a look at the ppt on www.astertest.com
Zoa.
Jon Radon wrote:
I think the wiki has most of this covered. Just requires a little
reading and investigation.
http://www.voip-info.org/tiki-index.php?page=Asterisk%20dimensioning
I really think it's going to be impossible to
How does IAX authenticated transfer work? Is there
any documentation available? Mark spoke about it in the paper comparing SIP and
IAX. However I cant seem to find additional info on it
Jason
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Hi Leonardo,
This is not a libtool issue. It looks like you must have an ancient C
compiler, that doesn't understand C99 constructs.
Steve
Leonardo Gomes Figueira wrote:
Hi,
Trying to update to spandsp-0.0.2-pre6 I got a compile error:
Making all in src
make[1]: Entering directory
I set an Asterisk server, what ports would I need to open for my
firewall? I'm using IAX and SIP if that helps. Thanks.
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I set an Asterisk server, what ports would I need to open for my firewall? I'm using IAX and
SIP if that helps. Thanks.
Read the Wiki below:
http://www.voip-info.org/wiki-Asterisk+firewall+rules
-Jeff
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Steve,
Steve Underwood wrote:
Hi Leonardo,
This is not a libtool issue. It looks like you must have an ancient C
compiler, that doesn't understand C99 constructs.
gcc 2.95.3
Any workaround or I really need to upgrade gcc ?
Leonardo
--
Leonardo Gomes Figueira
[EMAIL PROTECTED]
Gregory Junker wrote:
I'll stop doing it when Walsh stops posting about it:
http://www.faqs.org/rfcs/rfc1855.html
(from the RFC)
...Don't wander off-topic, don't ramble and don't send mail or post
messages solely to point out other people's errors in typing
or spelling. These,
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Mike Caley
Sent: Wednesday, November 17, 2004 9:14 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Port for Asterisk
I set an Asterisk server, what ports would I need to open for
I have already verified the permissions on the database. I had granted all
permissions on this database to the username I am using in the dialplan. I
used the statement GRANT ALL ON asteriskdb.* TO [EMAIL PROTECTED] IDENTIFIED
BY 'abc123';. I have logged into the MySQL console and was able to
[EMAIL PROTECTED] wrote:
Could it be that the MYSQL application is set up for read only?
Did I miss a compile option or something?
The MYSQL Application as it is is not suited for updates
and / or inserts.
See http://lists.digium.com/pipermail/asterisk-users/2004-August/060279.html
for my
Hi,
I'm using firefly to connect from a NATed network to a NATed asterisk
server, the lag between them is about 260-300 ms.
The problem is that the calls regularly hangup after a message like that :
chan_iax2.c:1139 attempt_transmit: Max retries exceeded to host
x.x.x.x on [EMAIL PROTECTED]/3
Hi!
I wanna know if somebody knows where I can buy this kind of VoIP phone here
USA?
TIA
*---*
*-Edwin Quijada
*-Developer DataBase
*-JQ Microsistemas
*-809-747-2787
* Si deseas lograr cosas excepcionales debes de hacer cosas fuera de lo
On Tue, 2004-11-16 at 23:34 -0700, Chris Modesitt wrote:
Steve,
Thanks for your feedback, after I restarted Asterisk the card came up as
expected. However I am still seeing these WARNINGS when I reload *, to be
clear I have not made any additional changes to zaptel.conf or zapata.conf
since
Thank you very much. That fixed my update problem.
Regards,
Shaun
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Andreas
Sikkema
Sent: Wednesday, November 17, 2004 9:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE:
Hi!
Somebody knows where can I buy this kind of VoIp Phone?
*---*
*-Edwin Quijada
*-Developer DataBase
*-JQ Microsistemas
*-809-747-2787
* Si deseas lograr cosas excepcionales debes de hacer cosas fuera de lo
comun
Leonardo Gomes Figueira wrote:
Steve,
Steve Underwood wrote:
Hi Leonardo,
This is not a libtool issue. It looks like you must have an ancient C
compiler, that doesn't understand C99 constructs.
gcc 2.95.3
Any workaround or I really need to upgrade gcc ?
Leonardo
That's really old. Right now
On Tue, 2004-11-16 at 19:47, Joseph wrote:
Is it possible to send a fax with asterisk in between? When I try to
send a fax I've got some COMREC error I think it is some kind of
communication error, it fail to negotiate protocol I think.
I'm trying to send a fax over standard line (not over
On Wed, 17 Nov 2004, Steven Critchfield wrote:
On Tue, 2004-11-16 at 23:34 -0700, Chris Modesitt wrote:
Thanks for your feedback, after I restarted Asterisk the card came up as
expected. However I am still seeing these WARNINGS when I reload *, to be
clear I have not made any additional
On Wed, 17 Nov 2004, Steven Critchfield wrote:
On Tue, 2004-11-16 at 23:34 -0700, Chris Modesitt wrote:
Thanks for your feedback, after I restarted Asterisk the card came up
as
expected. However I am still seeing these WARNINGS when I reload *, to
be
clear I have not made any
You're welcome.
It's been submitted.
Cyrille Demaret wrote:
Hi,
Thank you, it's working now!
Do you think that this patch will be included in the next cvs versions?
Sincerely,
Cyrille Demaret
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I've noticed this quite a bit recently as well. Also having major
problems with consistency registering with FWD. Is this a problem on
the IAXTel/FWD side, or is it possibly related to my configuration?
PF
On Wed, 17 Nov 2004 13:09:10 -0200, Fernando Pieri [EMAIL PROTECTED] wrote:
Hi,
I'm
Hi
I posted this on the ser list also, not sure what the best place for the
question
--
I have ser front ending asterisk, and all calls except sip--sip go
via asterisk, which then connects to a cisco box. Now I want to do
inbound from pstn, is is better to go again via ser, and then route
Hi,
I have a TDM400P with one FXO and an FXS. My Australian telco has enabled
call waiting and three way calling.
When I'm using my telephone connected to the FXO I can see/hear on my
handset that call waiting is enabled, however when I attempt to send a hook
flash or use the combination *0 I
Using Firefly 1.9.5 (thirdparty) on Win2k
Using Asterisk CVS HEAD 20041117 (also tried with 20040806 and
200410-something)
IAX2, no NAT. Firefly-Asterisk audio works, but I can't hear anything from
the other side.
Using GSM codec, also tried ulaw.
Any ideas?
-A.
relevant bits of iax.conf
Hi All,
I'm curious to know what software based SIP phones people are using under
Linux that work with Asterisk. I have tried several including kphone,
linphone, and SJPhone, I have the same problem with all of them, my voice
comes out quiet on the other end, and there is quite a bit of
Does anybody know if the CS version of the Polycom handset will take the SIP
image. If I have read correctly, the CS version is for Cisco Call Manager,
and is Cisco certified. Thanks in advance.
B. J.
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I have done that successfully with a SoundPoint IP500CS. It functions just
like the non-CS version once you put the SIP image on it.
MATT---
-Original Message-
From: B. J. Bomar [mailto:[EMAIL PROTECTED]
Sent: Wednesday, November 17, 2004 11:57 AM
To: [EMAIL PROTECTED]
Subject:
Greetings all,
Kinda new to the whole asterisk scene
- just got a test system up and running and stuff is really slick (but
you already knew that). We're heavily invested in LDAP on our campus
and I would very much like to store my SIP and H.323 device info in LDAP
along with the associations
Hi,
I am using X-Lite with Wine!
Diego Aguirre
- Original Message -
From: Peter Osborne [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Wednesday, November 17, 2004 2:47 PM
Subject: [Asterisk-Users] Software SIP Phones
Hi All,
On Wednesday 17 November 2004 12:15, Diego Aguirre wrote:
Hi,
I am using X-Lite with Wine!
wow! I triied to get it working under wine but it was a no go. I'm very
familiar with Wine, we run a few apps under wine here. Coudl you share your
config or somet tips to help me get it running?
Are
Peter Osborne [EMAIL PROTECTED] writes:
I would blame my onboard sound (I'm using a Toshiba M30 laptop)
except that I have had no problems using Skype on this machine.
I hear the Skype folks have done a very good job of getting high
quality sound out of laptops, so I guess the difference is
Hi,
I am also interested in what softphones other asterisk people are
using. I am using xlite but that doesnt seem to have any voicemail
capabilities (correct me if im wrong). You have to purchase xpro for
that. Does anyone have any suggestions?.
Apologies to the person who first sent this
Hello.
I'm using Asterisk with chan_capi to connect to the Netherlands PSTN.
Dialing in works allright, but dialing out does not work.
When using early B3 connects, I get the dialtone from the PSTN (but it
is a little choppy), but they do not seem to receive the digits I dial,
dialtone goes on
On Tue, 2004-11-16 at 16:12, Joseph wrote:
Is it possible to set a variable for a channel that follows the life of
the call?
I am doing a SetVar(TRANSFER_CONTEXT=from-sip) when a call comes out of
the queue, but if the agent tries to transfer the call, that variable is
empty.
Do I have
Well...I now have X-Lite running under Wine and it works pretty well (thanks
Diego) but there is still some background hum. I'm using a headset but it
seems to make no difference. The sound quality is now tolerable as I would
only be using this for placing remote calls as well as testing
Hi all,
We're considering using Asterisk in our small (8 user)
office. There is one feature that we have on our
current phone system that I haven't seen in the
documentation that I've read that I'd like to be able
to replicate with Asterisk.
On our current phones (Iwatsu) we have a button on
We had the same with our old phone system, I replaced it be adding a little
panel in our web based support system that shows extension status, call
duration, etc. I used an iframe that refreshes every 5 seconds, I wrote a
script in python that generates the data by watching the Asterisk
Hi Richard,
Richard Hansberry wrote:
Hi all,
We're considering using Asterisk in our small (8 user)
office. There is one feature that we have on our
current phone system that I haven't seen in the
documentation that I've read that I'd like to be able
to replicate with Asterisk.
On our
Probably a stupid question, but does astcc require cdr_mysql? thanks!
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Does anyone successfully run * on Solaris??? Can
anyone provide me any links discussing the topic?? Im looking for
more information I found this http://www.voip-info.org/wiki-Asterisk+Build+Notes+for+Solaris
but is there anyone that can provide me with more information.
Specifically I just
hi,
i made a patch that allows the compilation of chan_capi-0.3.5 against
current CVS-HEAD of asterisk.
it also incorporates the capiAnswerFax patch
the patch can be downloaded at
http://www.levigo.de/VoIP/chan_capi-0.3.5-cvs-HEAD-patch.tar.bz2
regards
Frank Sautter
We're considering using Asterisk in our small (8 user)
office. There is one feature that we have on our
current phone system that I haven't seen in the
documentation that I've read that I'd like to be able
to replicate with Asterisk.
On our current phones (Iwatsu) we have a button on the
phones
On our current phones (Iwatsu) we have a button on the
phones for each extension that lights up when that
extension is ringing or is in a call, so I can see at
a glance if one of my coworkers is on the phone before
I go barging into his office. Also, if I am in a
coworker's office and
Hello, I'm an Asterisk new user,
I need to know if it is possible to use the IVR and Voicemail using G729, I
have two SIP phones that uses G729 and I can not heard the IVR and the voice
mail.
thanks in advance.
Alvaro G.
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Peter Osborne a écrit :
Hi All,
I'm curious to know what software based SIP phones people are using under
Linux that work with Asterisk. I have tried several including kphone,
linphone, and SJPhone,
Try X-lite under wine. Works well.
___
No, It uses its own table cdrs and does not require cdr_mysql.
Look astcc.cgi and see sql statement :)
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I only want to dial to the Internet and stay nailed up for data.
I want my voice to be VOIP.
Anyone doing this?
James Taylor
Metrotel
903-793-1956
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Matt Riddell wrote:
Leandro Morgado wrote:
Although, I still think that there is some kind of incompatibility or
battery drop timing problem between Asterisk and the Nokia 32. I wish
I knew more about telecomms and wcfxs.c to fix it!
:-) You and I both!
Well.. i'm just leaving this but the
according to Sun, all Linux apps run under Solaris 10 ... would be interested in anyone who has actually done it
- hcir
On Nov 17, 2004, at 9:06 AM, Gerardo Bassett wrote:
x-tad-biggerDoes anyone successfully run * on Solaris??? Can anyone provide me any links discussing the topic?? Im looking
On Wed, 2004-11-17 at 18:43, Leandro Morgado wrote:
[SNIP]
Well.. i'm just leaving this but the mailing list archives. It really
must be some kind of Disconnect Supervision incompatibility between the
Nokia 32 GSM and Asterisk. Maybe asterisk doesnt like the duration that
the Nokia drops
I am waiting for solaris 10 for x86.
On Wed, 17 Nov 2004 09:53:31 -0900, Rich Allen [EMAIL PROTECTED] wrote:
according to Sun, all Linux apps run under Solaris 10 ... would be
interested in anyone who has actually done it
- hcir
On Nov 17, 2004, at 9:06 AM, Gerardo Bassett wrote:
Hey,
I'm trying to patch the latest CVS snapshot with the BroadVoice patch
but I get this when I try:
[EMAIL PROTECTED]:/usr/src/asterisk/channels# patch chan_sip.c sip_patch.diff
patching file chan_sip.c
Hunk #1 FAILED at 213.
Hunk #2 succeeded at 315 (offset 9 lines).
Hunk #3 FAILED at 485.
Jongsuk Lee wrote:
I am waiting for solaris 10 for x86.
You can download 32 bit versions now.
I just downloaded the sparc version.
On Wed, 17 Nov 2004 09:53:31 -0900, Rich Allen [EMAIL PROTECTED] wrote:
according to Sun, all Linux apps run under Solaris 10 ... would be
interested in anyone who
Hi
I know this is more or less an AMP specific question. But..
I'm currently testing the features provided by AMP - VoiceMail etc.. I have
modified the dial plan to suite my needs. I can provide it if anyone like to
see it.
I have a problem. Instead of sending the call to: exten =
Hello everyone,
Since releasing my very beta, test version of AstLinux almost two weeks
ago, there have been over 300 downloads from all over the world, with
over 100 in the first 24 hours. I was very surprised by the response,
and I have come up with a new version with many, many more
Hi list!
I'm trying to configure the Kirk IP 600 wireless (DECT) system. The
wireless phones are regular DECT phones, the Kirk IP 600 is doing the voip
part by registering/gatewaying the phones to a callmanager server.
The phones do not work and I think the problem is that they do not
register
Bob Knight wrote:
I am waiting for solaris 10 for x86.
You can download 32 bit versions now.
I just downloaded the sparc version.
Under Solaris Express, Solaris 10 is available now for
testing/evalutation (a public beta really) - and free.
It contains the 32 and 64 bit SPARC binaries, but only
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Arsen Chaloyan
Sent: Friday, November 12, 2004 3:24 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Re: Snom 190/220 dialplan strings?
Anyone have an example dialplan string as
Steve Underwood wrote:
Hi Angel,
It is working pretty well. I think it will be available about the end
of the year. I will not be free. It will be supplied with a
commercially licenced Asterisk.
Here's a question: if the author has purchased a commercial license to
use Asterisk, and I get
I would like to have CallerID reading from my outlook contacts. I could
export the Outlook thing to a csv file - does anyone have a script that can
read CallerID of it then?
thanks
Jens
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Hello All,
Please let me know if you have any comments or suggestions how to
resolve the following situation. My configuration is:
- Dell PowerEdge 1750 (standard), one CPU Xeon 3.0GHz
- TDM400P with 4 FXO, connected to 4 regular phone lines (PSTN)
- Asterisk 1.0.0 from cvs
- Zaptel and Zapata
Hi Angel,
It is working pretty well. I think it will be available about the end
of the year. I will not be free. It will be supplied with a
commercially licenced Asterisk.
Here's a question: if the author has purchased a commercial license to
use Asterisk, and I get binary modules from
Here's a question: if the author has purchased a commercial license to
use Asterisk, and I get binary modules from him, I can still use them
with my CVS-based Asterisk, right?
You may be able to do that. You could always run a couple Asterisk
boxes, run IAX2 between them and leave the
Question: Does anyone know of a lightweight popup method to put an
incoming call ID string on a client machine? Something as simple as
winpopup would work great- for example: I have a call coming in on Zap/4
but the phone on Zap/4 doesn't have a call ID display. Could I somehow
configure
On Wed, Nov 17, 2004 at 01:13:57PM -0500, Noah Miller spake thusly:
On our current phones (Iwatsu) we have a button on the
phones for each extension that lights up when that
This seems to be a popular request these days. Most places I've seen
call this shared lines I thought this was
On Wed, 17 Nov 2004, Jason Becker wrote:
On our current phones (Iwatsu) we have a button on the
phones for each extension that lights up when that
extension is ringing or is in a call, so I can see at
a glance if one of my coworkers is on the phone before
I go barging into his office.
Leandro Morgado wrote:
Correction! I do need busydetect (i had forgotten to comment it out) to
detect hangups. I'm not as familiar with this Smartcell GSM terminal but
I dont think it drops the battery when call are hung up (in
/var/log/messages voltage stayed at 9V during the call. After I hung
On Wed, 17 Nov 2004, Thomas Hutton wrote:
Question: Does anyone know of a lightweight popup method to put an
incoming call ID string on a client machine? Something as simple as
winpopup would work great- for example: I have a call coming in on Zap/4
but the phone on Zap/4 doesn't have a call
Actually it seems like I have solved the problem.
I turned off the Unconditional Call Forward option by dialing *73 from the
phone. I don't get the Got SIP response 302 Moved Temporarily back from
192.168.0.20 message anymore.
Worth mentioning; this is a Grandstream Bridgetone-100 phone.
On November 17, 2004 03:05 pm, Thomas Hutton wrote:
Question: Does anyone know of a lightweight popup method to put an
incoming call ID string on a client machine? Something as simple as
winpopup would work great- for example: I have a call coming in on Zap/4
but the phone on Zap/4 doesn't
On Wed, 2004-11-17 at 13:18 -0600, Kristian Kielhofner wrote:
Hello everyone,
Since releasing my very beta, test version of AstLinux almost two weeks
ago, there have been over 300 downloads from all over the world, with
over 100 in the first 24 hours. I was very surprised by the
Anyone have success with fax detection and zap cards and 1.0?
exten = 5124512424,1,Answer()
exten = 5124512424,2,Wait(1)
exten = 5124512424,3,Dial(SIP/3044,20,t)
exten = 5124512424,4,Hangup()
exten = fax,1,SetVar(FAXFILE=/var/spool/asterisk-fax/${UNIQUEID}.tif)
exten =
On Wed, 2004-11-17 at 17:05 -0300, Thomas Hutton wrote:
Question: Does anyone know of a lightweight popup method to put an
incoming call ID string on a client machine? Something as simple as
winpopup would work great- for example: I have a call coming in on Zap/4
but the phone on Zap/4
On Wed, 2004-11-17 at 11:49 -0800, Tracy R Reed wrote:
On Wed, Nov 17, 2004 at 01:13:57PM -0500, Noah Miller spake thusly:
On our current phones (Iwatsu) we have a button on the
phones for each extension that lights up when that
This seems to be a popular request these days. Most places
When I'm using my telephone connected to the FXO I can see/hear on my
handset that call waiting is enabled, however when I attempt to send a hook
flash or use the combination *0 I don't get a dialtone and the feature
doesn't work. The same goes for three way calling and cancallforward.
Are you
Jim Dossey wrote:
On Wed, 2004-11-17 at 13:18 -0600, Kristian Kielhofner wrote:
Hello everyone,
Since releasing my very beta, test version of AstLinux almost two weeks
ago, there have been over 300 downloads from all over the world, with
over 100 in the first 24 hours. I was very surprised by
/SNIP/
Can you legally redistribute the Digium
G.729 code?
/SNIP/
Is that a
question or a Statement? You know that no one can redistributeDigium's
G729 Codec for free.
What is included
here is the Executable that registers your codec, if you buy the license from
Digium and have the
In my quest to determine why asterisk wont talk to our Merlin Legend I
tried moving the D-Channel in zaptel to other position. But If I try
putting it anything different than dchan=24 asterisk fails to load
with start_pri:
Unable to open D-channel 24 (No such device or address)
I wonder if there
Any luck with these phones and their SIP firmware? I just changed one of
my IP10s to use SIP and I can't get it to work with Asterisk (it doesn't
register at all).
Regards,
Alejandro.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Florian
Overkamp
On Wed, 2004-11-17 at 14:44 -0600, Matthew Boehm wrote:
Anyone have success with fax detection and zap cards and 1.0?
exten = 5124512424,1,Answer()
exten = 5124512424,2,Wait(1)
I doubt 1 second is long enough to detect the CNG tones.
exten = 5124512424,3,Dial(SIP/3044,20,t)
exten =
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