On Wednesday 15 September 2004 01:02 am, Thomas Gallaway wrote:
William C. Lohr Jr. wrote:
Marty,
My business would own the actual T1, but I may provide an outbound
call service for a client and would want their name sent as well as
their toll free number and not the local number for
Whom ever sells you your pstn connection would
have to insert the name from their database. Only the number is passed with the
call -pstn is as pstn does. The Clec, Ilec or whatever lec needs to add
name to CLID.
Brandon
- Original Message -
From:
William C. Lohr Jr.
To:
Hi friends,
I tried to dial 111 from CLI without any hard/soft phones.
I used the following config
when i called 111 from CLI by
CLI dial 111
I got these errors
-- Executing Dial(OSS/dsp, CONSOLE/dsp) in new stack
Sep 15 11:57:26 NOTICE[1217602880]: chan_oss.c:753
Hmm, I think the settings in sip.conf alone are not sufficient for what I look
for. Maybe I should be more precise to explain the problem.
When I make an outgoing call from my GrandStream (GS below) phone, I use a
prefix to differ between server A and server B (both PSTN gateways). A only
Thanks, that did the trick! :-)
Kinda weird though that the mp3's that actually come with Asterisk don't
work correctly 'out of the box'. Or is this a mpg123 bug?
Regards,
Evert Meulie
Andreas Roedl wrote:
Hello!
Am Dienstag, 14. September 2004 19:21 schrieb Evert Meulie:
Found new ID3
Has anyone found a solution for asterisk and r2
signaling ? Steve Underwood had given some information saying he had a working
asterisk working. I need it to work with Argentina R2 signaling
Sam
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Outbound proxy is an absolute must have. I will take a look at the
solution and provide feedback.
Thanks,
Chad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olle E.
Johansson
Sent: Monday, September 13, 2004 11:46 PM
To: Asterisk Users Mailing List -
I followed the Wiki instructions to get zaptel to work on
Fedora core 2. It looked like everything went perfect including the loading of
ztdummy. However, I am having meetme and MOH problems synonymous with ztdummy
not loading. Take a look at my lsmodAny ideas? (I am running stable
Hello List!
Since i can dial dial out now, i would like to dial in now :)
My problem right now is how asterisk knows, which msn belongs to wich
connected soft phone.
When i call 37, i get this error:
chan_capi.c:2051 capi_handle_msg: did not find device for msn = 37
This is my extentions.conf:
Steven Critchfield wrote:
On Tue, 2004-09-14 at 11:30, Maciek Kaminski wrote:
Apparently there are subtle diferences between meaning of MeetmeJoin
event depending on channel type.
Problem is: after originating a call from channel to MeetMe room i.e.:
[meetme]
exten = 1,1,Answer
exten =
Hello!
I am going to play around with transfering calls with capi in a bit.
Here is a little example from the readme:
---
example:
exten = s,1,Answer
exten = s,2,capiHOLD
exten = s,3,capiECT,55:50
will ECT the call to 50 using 55 as the callerid/outgoing msn
---
Now i am a little
Hi,
I wonder what FXO cards for 1, 2, or 4 analog channels are best (price,
reliability, features) for use in Europe ?
Where to purchase them - any recomendations for fast delivery ?
Thanks in advance,
Robert.
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Hi all,
I'm experincingIAX2 call drops for about 20% of
calls.
I tried 'notransfer=yes' and 'jitterbuffer=yes' but to
fail.
My systemconfiguration is like this.
PSTNAsterisk(TDM/Fxo
4port*3)=LAN(IAX2)=Iaxclient library
And iax.con is...
[EMAIL PROTECTED] wrote:
On 15 Sep 2004 at 1:52, San Singhania wrote:
Hello everyone,
I am having big problems trying to detect dtmf tones while a IVR
prompt is playing on zap channels. Sometimes the detection only starts
4-5 seconds into the prompts. Other times it works very well for the
1st
Hello,
I have investigated the issue a bit further, I was not able to find the
root cause, maybe it is the KT133 Chipset of my ASUS A7V. However I
found a bad hack to make it work under some circumstances. I commented
out the line:
printk(KERN_CRIT zaphfc: sync lost, pci performance too low.
Hi,
I got my * working fine with FWD at office with 2 extensions, i receive
calls and i can make calls thru FWD. I got also my * at home, and i
connected it using auth=rsa. From my home, i can make calls using my office
iax, but if i try to redirect incomming calls from FWD to my * at home, it
Hi all, use the brought up to date version of the Asterisk and I have the
following problem:
Mine asterisk stop to register the extensions and I do not obtain to execute
the command Stop Now.
I do not see no message of error in logs.
Somebody can help me?
Thank's,
JMoura
Hi
I have been playing with * for the last couple of weeks now.
I am also speaking to one of my customers about installing a * server in
addition to their Philips IS3090 switch. They are busy building a new office
block and I have convinced them to go VoIP. Currently the client is
You have installed asterisk = ok.
You try starting asterisk:
#safe_asterisk
errors ??
if none, then try
#asterisk -r
you enter the consol, and get the CLI command prompt.
/ Stig Henning
- Original Message -
From: Joao Carlos Moura [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent:
On Wed, 15 Sep 2004 03:45:59 -0600, Raul Elizondo (wizardteam)
[EMAIL PROTECTED] wrote:
I got my * working fine with FWD at office with 2 extensions, i receive
calls and i can make calls thru FWD. I got also my * at home, and i
connected it using auth=rsa. From my home, i can make calls using
Hi,
have you some idea, how to make roaming line with Asterisk?
i.e. is possible to have phone line assigned to user if migrating from one office to
another?
thanks
PJ
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I'm a newbie with a TDM11B. I've read the FAQs about linking
/usr/src/linux-2.6 to /usr/src/linux-2.6.8-1.521 and /lib/linux-2.6 to
/lib/linux-2.6.8-1.521 but still get a million errors and eventual abort
during compile.
Could someone point me in the right direction? I do a yum update every
day
Hi,
i am a lot further now. But i am stuck at the transfer feature from
chan-capi.
[default]
...
exten = 37,1,GoTo(demo,s,1)
[demo]
exten = s,1,Answer
exten = s,2,capiHOLD
exten = s,3,capiECT,35:37
This should transfer the call from 35 to 37. It gets into HOLD, but then
never tries to Ring
On Wed, 15 Sep 2004 13:39:49 +0200, Pavel Jezek [EMAIL PROTECTED] wrote:
have you some idea, how to make roaming line with Asterisk?
i.e. is possible to have phone line assigned to user if migrating from one office to
another?
Sure you can.
One way would be to simply ring the local and the
thanks for idea, but this is not exactly what I need, assume:
one employee working in office (open-space cubes), when this employee leave the
work, on the same place come another employee
so that, I can't ring both lines and can't use bluetooth device :(
I thing to do some login to phone
What I believe you want is this.
ln -s /lib/modules/2.6.5-1.358/build linux-2.6 Only pointing to your
kernel. Run this in the /usr/src directory
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeff
Borders
Sent: Wednesday, September 15, 2004 7:46 AM
To:
[EMAIL PROTECTED] wrote:
thanks for idea, but this is not exactly what I need, assume:
one employee working in office (open-space cubes), when this
employee leave the work, on the same place come another employee so
that, I can't ring both lines and can't use bluetooth device :(
I thing to
Being that its a lot of Bandwidth, I need to compress
the channels to use less bandwidth on the transfer.
TDMoE is ruled out. What can I do with Asterisk to
achieve this aim.
Best regards
Arinze Izukanne
--- Paul Mahler [EMAIL PROTECTED] wrote:
A full E1 is a LOT of bandwith to try and push
Senad Jordanovic wrote:
[EMAIL PROTECTED] wrote:
thanks for idea, but this is not exactly what I need, assume:
one employee working in office (open-space cubes), when this
employee leave the work, on the same place come another employee so
that, I can't ring both lines and can't use bluetooth
I use G.729 as the codec and use compressed RTP (CRTP) to reduce each call's
bandwidth. Over satellite and based on slot assignment overhead, that
equates to approximately 17Kbps bidirectional per call.
I haven't tried using IAX trunking, but I suppose you could transcode and
use one of the even
Very strange - I just compiled the most up to date version
(CVS-09/15/04) of asterisk using the asterisk-update.sh script and I
still don't have the command show file formats.
Does anybody of you have a clue why this could be?
Thanks,
Christian
Update your asterisk install them because you
maybe this will be possible to do this web application for
login even directly from phones!
because these phones have mini web browser and can display XML
formated pages,
so,thanks for good
idea!
PJ
make a simple web interface.. where user
logs in..interface tells a script on * server
Hi Benjamin,
Thanks for answering, now i got some other questions.
Acording to http://www.voip-info.org/wiki-Asterisk+iax+rsa+auth, and my
understanding of peer and user, [FWD-service] in the sample you provide me
should be type=peer as it will be the master/server, and [FWD-gw] should be
make a simple web interface.. where user logs in..
interface tells a script on * server about users
location/extension/device. then your script will re-create TFTP
files, sends
reboot to 7940/7792 and you are done...
Seems like overkill when you can just use the agent support in
Well Julio, in countries where there are no reliable
regional communication backbones with lower return
times, satellites are used and in most cases the
quality is outstanding for a good implementation even
for a double hop.
I could give you a call over a satellite link, G729
and latency of up
We have a satellite line and using IAX.
Everything work fine. Latency about 620ms - 680ms
Greetings.
- Original Message -
From: Arinze Izukanne [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion [EMAIL PROTECTED]
Sent: Wednesday, September
Hi Folks,
I've been playing with call recording for our support department which was
kinda going ok until I spotted something odd in the CDR. None of the
support calls are being entered into the CDR properly.
I'm using mysql as the back end and Areski's web based front end and all
was going
On Wed, 15 Sep 2004 14:15:24 +0200, Pavel Jezek [EMAIL PROTECTED] wrote:
thanks for idea, but this is not exactly what I need,
It wasn't supposed to be *exactly* what you need ;-) That's not what a
mailing list is about. We can give each other clues and ideas for how
things can be done *in
On Wed, 2004-09-15 at 22:37, Senad Jordanovic wrote:
make a simple web interface.. where user logs in..
interface tells a script on * server about users
location/extension/device. then your script will re-create TFTP
files, sends
reboot to 7940/7792 and you are done...
Seems
Benjamin on Asterisk Mailing Lists wrote:
On Wed, 15 Sep 2004 14:15:24 +0200, Pavel Jezek [EMAIL PROTECTED] wrote:
thanks for idea, but this is not exactly what I need,
It wasn't supposed to be *exactly* what you need ;-) That's not what a
mailing list is about. We can give each other
Actually that should br your round-trip-time. One way latency would be half
that.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bartosz
Jozwiak
Sent: Wednesday, September 15, 2004 8:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
I've seen a lot of times, people that try to get R2 MFC to
*, most of them trying to use Dialogic Boards (BTW They 're Very expensive),
none of them where succesfully,
If you want to use PCI Cards on your server,why don´t
u ask to your carrier to provide you E1/PRI? or better put a Gateway
Hello!
I have been googling a lot and asked wiki a few times now, but i cant find
a howto for setting up a voicebox.
Any link/hint would be great!
Thanks, Mario
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Thanks, that looks great. I was not aware of the -L option but it will
work just fine. Once I get it added, unless somebody else comes up with
a patch first, I will try to get it submitted to astcc.
Thanks,
Darren Wiebe
[EMAIL PROTECTED]
Nicolás Gudiño wrote:
Hello,
On Tue, 14 Sep 2004
Hey Guys,
I am getting ready to implement Asterisk for my company. We plan on housing
an asterisk server at a local termination provider and have another here in
the office. The two communicate via IAX. Looking something like...
PSTN
|
PROVIDER
| (PRI)
Asterisk
| (IAX over the Internet)
I have one * with an avm c2 Isdn card installed and connected to 2 isdn
lines with 2 different numbers (444 and 555). I configured capi.conf as
follows:
[interfaces]
msn=444
incominggmsn=*
controller=1
devices=2
..
msn=555
incominggmsn=*
controller=2
devices=2
..
I would like to
Hi All,
I have what IMHO is an interesting issue. I'm using Cisco 7940's with
the 7.2 SIP load and Asterisk CVS-HEAD-09/10/04-10:11:46. Everything is
working great so far, except one small issue.
When a user presses the 'Trnsfer' soft-key, dials the other extension,
and presses 'Trnsfer'
Hello!
Am Mittwoch, 15. September 2004 08:34 schrieb Evert Meulie:
Thanks, that did the trick! :-)
Kinda weird though that the mp3's that actually come with Asterisk don't
work correctly 'out of the box'. Or is this a mpg123 bug?
This is a mpg123 bug. I guess the latest version won't dump
On Wed, 2004-09-15 at 16:10 +0200, wrote:
Hello!
I have been googling a lot and asked wiki a few times now, but i cant find
a howto for setting up a voicebox.
Any link/hint would be great!
I'd hate to refer you to the Wiki but the answer is in there :) (you did mean a
voicemail box, right?)
I have cisco phones running
SCCP, and a cisco 2600 with FXO Im using for PSTN access.
I can dial out, but inbound
calls are not ringing a phone. Please see my config
In the 2600 Im
PLARing the line and I have a SIP Dial-Peer for 4001
voice-port 1/1/0
output attenuation 0
On Wed, 15 Sep 2004 06:36:48 -0600, Raul Elizondo (wizardteam)
[EMAIL PROTECTED] wrote:
Acording to http://www.voip-info.org/wiki-Asterisk+iax+rsa+auth, and my
understanding of peer and user, [FWD-service] in the sample you provide me
should be type=peer as it will be the master/server, and
Hello!
Am Mittwoch, 15. September 2004 16:35 schrieb Gonzalo Servat:
I have been googling a lot and asked wiki a few times now, but i cant
find a howto for setting up a voicebox.
It was a simple search for voicemail in www.voip-info.org so I'm not sure
what freaky search terms you were
Hi.
I see that message in console.
-- Zap/1-1 is ringing
!! Unknown IE 40 (cs6, Unknown Information Element)
-- Zap/1-1 answered SIP/1016-e34b
As see in older messages it some Information send by phone station to my via
PRI.
But what does IE 40 mean? I cann't find information element 40
Hello all,
I have a question concerning the calling number with an incoming PSTN call
through a E100P :
Here is what I see with a pri debug :
Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony
Numbering Plan (E.164/E.163) (1)
Hello!
Am Mittwoch, 15. September 2004 16:35 schrieb Gonzalo Servat:
I have been googling a lot and asked wiki a few times now, but i
cant find a howto for setting up a voicebox.
It was a simple search for voicemail in www.voip-info.org so I'm not
sure what freaky search terms you were
I've used it and it works great!
I think it's vital that chan_sip include outbound proxy support. * is not
only acting as a PBX and a telephony gateway but also to the termination
provider it acts as a SIP UA and needs to have all of the features that a
SIP UA would have including outbound proxy
I was looking for this as well for use with my own prepaid callingcard app.
I have updated the Wiki to include some extra options that I found inside
the app_dial.c source.
http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+Dial
My additions are right underneath the L description.
Hello--
I've been playing with the privacy options on my home/home-office system
since August last year, and have some results, gleaned from my CDR
records, which over the last 13 months, number a total of 8672, which
includes incoming, as well as outgoing calls.
Before I start spitting out
Hello!
Am Mittwoch, 15. September 2004 17:03 schrieb [EMAIL PROTECTED]:
I guess, he used voicebox as search term...
Yup, i did :)
I found a lot about setting up my voicemail now.
But i couldn`t find much how you collect/call/retreive the recorded mail.
Any ideas/links about that?!
Title: Asterisk and Cisco MC3810 Help needed
Does anyone have experience with using an MC3810 with asterisk? I have an MC3810 with an AVM6 and 3 FXO ports plus a VCM6 voice compression module. I am trying to use this for fxo ports in an OS X Asterisk installation which does not support
For some reason, voicemail doesnt seem to be firing
is there something that anyone can see wrong with my extensions.conf ?
exten = 4001,1,SetCalledParty(Justin
4001)
exten = 4001,2,Dial(SCCP/justin)|5
exten = 4001,3,Voicemail,u4001
Hi, another problem.
I configure TAPI driver for outlook.
(https://sourceforge.net/projects/asttapi/)
Yesterday all work fine.
I configure that Local Phone is Zap/g1/772323 and external call is going
to context default.
When I call to sip - all work ok.
When I call to city (via Zap) local phone
Hi,
Thank you for your reply.
I guess what I am trying to do is only different in the sense that I
want to have multpile 'main menus' for each of our different clients.
I was hoping to send them to the correct sub context from the main
inbound context with a Goto statement using the extension
Yup, i did :)
I found a lot about setting up my voicemail now.
But i couldn`t find much how you collect/call/retreive the recorded mail.
Any ideas/links about that?!
Thanks, Mario
ok, that would be a dialplan issue. You need to do something like this in
your dialplan.
[mycontext] ; The
Daniel Poulsen wrote:
Hi,
Thank you for your reply.
I guess what I am trying to do is only different in the sense that I
want to have multpile 'main menus' for each of our different clients.
I was hoping to send them to the correct sub context from the main
inbound context with a Goto statement
Is anyone using Galaxy Voice with the current CVS head/Stable RC1/RC2
and is able to receive incoming calls?
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To UNSUBSCRIBE or update options
Hello,
Has anyone else ever experienced Static Problems with a T100P connected
to an Adtran Total Access 750? We have two FXO modules in the Chasis to
interface to Verizon. At first I thought it was just one line, and was
Verizon's fault, but now we are seeing it across all lines and it comes
I've a few days ago turned off our (old) IP-Telephony-System Swyx and
activated Asterisk.
...the starting off all my problems ;-)
When calling from the outside (via Pri E1-connection to a HST Saphir V card
with capi-drivers on original shipped SuSE LINUX 9.0) and the pbx is playing
some
Steve-
That's an interesting/amusing story! The only thing I would worry about is
using the Zapateller SIT tone as the first thing whenever there's no caller
ID. In many places (like here in California), a good percentage of people
have caller ID blocking on outbound calls from their home
Hi,
I am new to Asterisk and have some general questions _before_
I start buying equipent to install and get everything up-and-running.
(this means I have no running Asterisk (yet)).
I have read already a lot of doc, but some things are not
clear to me, since I'am inexperienced in Asterisk and
On Wed, 15 Sep 2004 13:25:35 +0100 (BST), Arinze Izukanne
[EMAIL PROTECTED] wrote:
Being that its a lot of Bandwidth, I need to compress
the channels to use less bandwidth on the transfer.
TDMoE is ruled out. What can I do with Asterisk to
achieve this aim.
Use IAX and the ILBC codec.
You
Greetings folks;
I guess I must be missing something, because for the life of me I can't
seem to make this work. I have remote clients connecting to Asterisk using
IAX2, these clients have changing IPs so we're using the useful register
tool.
The client can call out successfully, that's not an
I am and as far as I can tell its OK.
Kevin said:
Is anyone using Galaxy Voice with the current CVS head/Stable RC1/RC2
and is able to receive incoming calls?
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[EMAIL PROTECTED]
Christopher L. Wade wrote:
Hi All,
I have what IMHO is an interesting issue. I'm using Cisco 7940's with
the 7.2 SIP load and Asterisk CVS-HEAD-09/10/04-10:11:46. Everything is
working great so far, except one small issue.
When a user presses the 'Trnsfer' soft-key, dials the other extension,
On Wed, 15 Sep 2004, JP Hindin wrote:
Greetings folks;
I guess I must be missing something, because for the life of me I can't
seem to make this work. I have remote clients connecting to Asterisk using
IAX2, these clients have changing IPs so we're using the useful register
tool.
On Wed, 15 Sep 2004 [EMAIL PROTECTED] wrote:
iax2 debug will give you more info on what is happening.
Tried this - there is *no* debug returned when this call is attempted.
But perhaps its that you need to include a username in your dial
IAX2/[EMAIL PROTECTED]/${EXTEN} and put it in the
On 15/09/2004 22:41 Benjamin on Asterisk Mailing Lists said the following:
On Wed, 15 Sep 2004 06:36:48 -0600, Raul Elizondo (wizardteam)
[EMAIL PROTECTED] wrote:
Acording to http://www.voip-info.org/wiki-Asterisk+iax+rsa+auth, and my
understanding of peer and user, [FWD-service] in the sample you
hallo,
my asterisk server is working fine, the only problem i have is, that every
morning when i look at my sipgate page i am logged out, when i do a cli
reload, everything is working until the next morning.
so my question, how can i force my asterisk server to keep logged in at my
sip
Hi!
This brings up a good point that has had me scratching my head for a long
time. Is there a good searchable archive of the asterisk mailing lists?
http://www.voip-info.org/wiki-Asterisk+FAQ
Philipp
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[EMAIL
Atuc a écrit :
hallo,
my asterisk server is working fine, the only problem i have is, that
every morning when i look at my sipgate page i am logged out, when i
do a cli reload, everything is working until the next morning.
so my question, how can i force my asterisk server to keep logged in
at
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
I try to get callers forwarded to by mobile phone when they dial a
certain digit.
In my extensions.conf I have defined the following:
[279]
exten = s,1,SetLanguage(de)
exten = s,2,Wait,5
exten = s,3,BackGround(demo-congrats)
exten =
At 19:43 15.09.2004, you wrote:
if you're using ppp, you can add an asterisk reload command in your
ip-up.local
how does it look like? do you meen over the manager server? any infos on
the net?
thanks,
alex
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[EMAIL
Atuc a écrit :
At 19:43 15.09.2004, you wrote:
if you're using ppp, you can add an asterisk reload command in your
ip-up.local
how does it look like? do you meen over the manager server? any infos
on the net?
man ppp
--
Daniel
___
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Huddleston, Robert wrote:
The only disadvantage we found to using the OH323 channel driver is that we
cannot now register netmeeting or other h323 directly to the * With the
What do you mean cannot now register? asterisk-oh323 doesn't implement
gatekeeper functionality. It never did. Just use
On 15 Sep 2004 06:26:29 -, Murali [EMAIL PROTECTED] wrote:
Hi friends,
I tried to dial 111 from CLI without any hard/soft phones.
Well, the ellegant solution is to disable OSS/ALSA and use a softphone :)
I suggest SJphone if you want a SIP client, or iaxcomm if you rather use IAX2.
On Thu, 16 Sep 2004 00:49:11 +0800, Dinesh Nair [EMAIL PROTECTED] wrote:
if the office asterisk had an iax entry of type friend, and the home
asterisk did a register with the office asterisk, then the office asterisk
would not need another entry for the home asterisk ? is this assumption
On Wed, 15 Sep 2004, Marconi Rivello wrote:
On 15 Sep 2004 06:26:29 -, Murali [EMAIL PROTECTED] wrote:
Hi friends,
I tried to dial 111 from CLI without any hard/soft phones.
Well, the ellegant solution is to disable OSS/ALSA and use a softphone :)
I suggest SJphone if you want a
Hi Guys,
Does anyone know of E3 PCI cards that work with
Asterisk?
Arinze
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Alex Ongena [EMAIL PROTECTED] writes:
1) Status info: can I see on my 7960 equipment (eventualy
with the 7914 extension) who is free/busy and alike ?
While it looks like recent SIP images support this multiple call
appearance feature, to my knowledge asterisk does not.
--
Matt Ranney -
On Wed, 2004-09-15 at 14:55, Arinze Izukanne wrote:
Hi Guys,
Does anyone know of E3 PCI cards that work with
Asterisk?
None currently, and it may be a while before it is wise to trust that
many voice calls in and out of a single PC. You would do well to split
it apart into single E1s and
Arinze Izukanne wrote:
Hi Guys,
Does anyone know of E3 PCI cards that work with
Asterisk?
Arinze
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Hello
Asterisk is compiled and running perfectly...
But when i try to compile Channel_h323... that's another story :-|.
I'm compiling using RH9, OpenH323_1.12.2, pwlib_1.5.2, i compiled and
installed them myself... GnuGk is running smoothly using them.
Also my intention is to terminate calls
Here is the information on doing SIP to H323 :
http://lists.digium.com/pipermail/asterisk-users/2004-July/056425.html
On Wednesday 15 September 2004 08:43 pm, Carlos Maynard wrote:
Hello
Asterisk is compiled and running perfectly...
But when i try to compile Channel_h323... that's
On 2004-06-25 at 22:12, Steve Hanselman ([EMAIL PROTECTED]) wrote:
If you cat /proc/interrupts is anything else sharing with the TEs?
It doesn't seem to:
CPU0 CPU1
0: 5413 5623IO-APIC-edge timer
1: 0 5IO-APIC-edge keyboard
2:
Hi... I think I've done that by typing dial 12345, the digits...
For example:
dial sip/[EMAIL PROTECTED]
and when the answer is attended, you type again:
dial 123456778... the digits you want to send... you can repeat that as
many times as you like... at least I think it has worked with me. I'm
How are you determining what company they are going to hit? You can always
use a menu for that, of course. press 1 for Acme Press 2 for Company z
press 3 for a directory.
My comment about the 9 was against your question #2. Once you jump to
Voicemail main, your keypress are captured and acted
exten = 4001,3,Voicemail(u4001)
- Original Message -
From:
[EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, September 15, 2004 10:26
AM
Subject: [Asterisk-Users] voicemail
For some reason, voicemail doesnt
seem to be firing
is there something
Hi.
I have a x100p card installed on my asterisk box... my zapata.conf file
includes the following lines:
[channels]
context=default
switchtype=national
signalling=fxo_ls
rxwink=300
echocancel=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
Basically, the zapata.conf file generated
Hi all,
I have problems with rxfax application. It seems to be ok but I don't receive the fax in my directory.
My extension.conf is as follow:
[macro-fax]
exten = s,1,SetVar(FAXFILE=/var/spool/asterisk/incoming/${UNIQUEID}.tif)exten = s,2,rxfax(${FAXFILE})
[fax]
exten = 100,1,macro(fax)
Rodolfo,
Shouldn't it be siganlling=fxs_ls for the x100p ?
Where is your channel = 1
What is in your zaptel.conf ?
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Rodolfo
Grave
Sent: 15 September 2004 22:52
To: Asterisk Users Mailing List -
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