On Mon, Sep 15, 2003 at 10:27:15PM +0200, Dave Cotton wrote:
On Mon, 2003-09-15 at 22:11, Tom (UnitedLayer) wrote:
Anyone have a good source for BT-101 phones?
Yes.
But it may not work for you because I've no idea on which of the 5
continents you are.
I am looking for Grandstream
I have implemented a work around for now.
Compiled and installed Asterisk on the gateway (which will only be used for
talking externally).
Setup an IAX link between the main Asterisk box and gateway Asterisk.
All works perfectly. This is probably the most secure solution anyway,
rather than
-= On 15 Sep 2003 11:09:38 -0600, tom [EMAIL PROTECTED] said:
And interestingly, the Digium card looks a lot like a product sold
by Tigerjet, called the Personal Phone Gateway. I'm purely
speculating on this, but Digium could have used Tigerjet's reference
design for their own board.
That's
Hello all,
Does anyone has experience of deploying Asterisk based VoIP
solution in a universitywide campus. We are at present
investigating various Soft PBX for this purpose from different
vendors Digium,Snom, Pingtel...
We are looking at serving more than 5000 clients and we want to be
very
Using a channelized T1 here for our * box, we get ANI and DNIS
information inband over
DTMF, in the format *1234567890*222333*, where 1234567890 is the ANI
and 222333 is the
DNIS. Any hope for processing this effectively without resorting to AGI
scripting? Right now, * gets confused and
guys,
what firmware version of audiocodes mp104 fxs are
you using with asterisk?
i'm having sip stack problems.
~kelvin
Nikotel is shipping from US and Germany. Germany to France will take
3-4 days in shipping.
Louis-David Mitterrand wrote:
On Mon, Sep 15, 2003 at 10:27:15PM +0200, Dave Cotton wrote:
On Mon, 2003-09-15 at 22:11, Tom (UnitedLayer) wrote:
Anyone have a good source
Hello
That's true and I agree with this but I haven't found a multiport (say
4 port) FXO card from Digium.
Regards,
Daniel
Peter Brown a crit:
Tarun,
The Digium site shows other hardware that is compatible with Asterisk.
I would strongly urge you to support Digium by buying their
Sorry, did not catch up that you want to have a bt-102, we have just
bt-101 in stock currently which have only one network connector.
Anyway, bt-101 is for free, you just have to open an account at nikotel
and charge your account with 119 euro (inkl. taxes) to receive the
bt-101.
Shipping is
On Mon, 2003-09-15 at 11:52, Klaus-Peter Junghanns wrote:
i dont think that the Eicon Diva Server 4BRI's NT mode feature will
work with linux/capi. I think the feature in the driver is for their
PRI cards (where everything is always P2P). i may be wrong, though.
I just had a chat with
I'm a newbie, not afraid to admit it. I tried to install my TDM400P
without the auxilliary (drive) power connection and got this failure
when loading the wcfxs module:
Freshmaker version: 63
Freshmaker passed register test
ProSLIC on module 0 failed to powerup within 510 ms
Unable to
Hi all,
i have tried to connect to a clarent gatekeeper.
I have used both of h323 drivers chan_h323.so and chan_oh323.so.
But no one can register to this gatekeeper.
Our ip is activated on this gatekeeper.
Maybe, i do wrong anything
I have only set the gatekeeper option in the h323.conf or
If the gatekeeper requires a password and you don't provide one
during the registration, then it will fail.
In oh323.conf use the gatekeeperPassword to provide the passwd.
If this is not the case enable tracing info in oh323.conf, rerun
and send me
the trace file to take a look.
Michael.
Thomas
Yes you can use the Linejack on asterisk and Yes you can use it as an FXO, **BUT**
only for incoming calls :( Nobody explained me properly why the code was not developed
but as you know Asterisk is Digium and Digium makes voice boards, so... In other
words, what they are saying is: Buy Digium.
Hi Michael,
this gatekeeper works without a password but with a H323-ID, but this will
be send with the dial command, i think.
Here is the trace with trace level 10 (?)
Regards,
Thomas.
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Michael
I'm still stuck on this. The * is on a private network IP 192.168.0.7
and the 7960 is on 192.168.0.6. When I show 'sip peers' the 7960
appears to be registered although I can't make any calls and still get
the packet retries. I have also checked and re-checked the settings on
the phone. Any
Hi everybody,
I'm trying to configure * for
make SIP calls. Now I'm doing several test but I have some errors.
Firstly I willdescribe my
scenario.
Client Software(Private IP
192.168.0.181, SJ Phone over Windows 2000) Router Adsl (Public ip A.B.C.D,
and NAPT on port 5060 to
Hi,
Can incoming fax from X100P FXO received on TDM400P
FXS port (which is connected to a fax machine)?
If possible, at what speed, 14400 or 9600 baud
rate?
Thanks.
Hi
I've just signedup for Distinctive ringing on my PSTN line in the UK, could
anyone explain what I need to add in the conf files to detect and route based
on in comming Distinctive ringing
Thanks in advance for your help
Robb
___
Asterisk-Users
I am having problem with the TDM400P. The ABIT BH6 cannot detect the card
from the BIOS, and loading of module returns No such address.
Though the card TDM400P can work OK on another Pentium 4 MB.
- Original Message -
From: Steve Haehnichen [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent:
Possible to call the user's phone and allow the
user to access or read the voice mail whenever there is a voice
mail?
I know the email notification works.
Thanks.
Does it means that a PCI TAM modem with the linux driver and module will
work with *?
- Original Message -
From: Steve Haehnichen [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, September 16, 2003 3:17 PM
Subject: RE: [Asterisk-Users] Analog FXO Card
-= On 15 Sep 2003 11:09:38
At 21:11 16-9-2003 +0800, you wrote:
Possible to call the user's phone and allow the user to access or read the
voice mail whenever there is a voice mail?
I know the email notification works.
Sure. Whenever I program a voicemail entrance (i.e. after a busy signal or
timeout on the Dial) I allow
Thomas Haeger wrote:
No. I have installed the versions wich your special friend has recommended.
Shall i try to update to the newest versions ? (But then wouldn't work the
chan_h323.so further...)
I don't know what are the problems with that driver, but, yes,
you should install the latest
I'm using cico's with SIP... And it works great :-)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Edward Gomez
Sent: dinsdag 16 september 2003 15:52
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Cisco Gateways
Hi all,
Just wondering if * can work
Hi,
On 15 Sep 2003 at 11:52, Klaus-Peter Junghanns wrote:
We are working on an alternative, a passive multiport ISDN card that
supports TE and NT mode with zaptel drivers for asterisk.
this sounds very interesting. Are there any details available yet?
cu
Reinhard
Download latest drivers from www.openh323.org
untar
./configure
make
make install
modprobe ixj
see your devices here:
cat /proc/ixj
phone.conf
[interfaces]
mode=fxo
context=incoming
device = /dev/phone0
where /dev/phone0 is your Linjack (could be /dev/phone1) if you had a phonejack PCI as
And how do you manage to have it actually authenticating? My Sjphone when I try to
register on the outgoing proxy, keeps saying the register failed...
-P
- Original Message -
From: Shaun Ewing [EMAIL PROTECTED]
Date: Sun, 14 Sep 2003 04:37:25 +1000
To: [EMAIL PROTECTED]
Subject: Re:
The X100P together with asterisk does not support the distinctive ringing
detection on the line. Asterisk however can generate the distinctive ring
over FXS ports.
regards
Martin
On Tue, 16 Sep 2003, Robert Boardman wrote:
Hi
I've just signedup for Distinctive ringing on my PSTN line in the
On Mon, 2003-09-15 at 22:35, Shaun Ewing wrote:
As per my problem yesterday with the Cisco 7960 and getting it talking to
Asterisk on a different subnet, I gave up trying and just put the Asterisk
box back on the internal subnet.
However, I made two changes:
- the external IP address is
Can the C7960 sip phone do distinctive ringing based on the origin of
a call? If so, anyone have a starter config or sample for a newbie?
(Example: two incoming X100P pstn lines. If call arrives via line 1,
ring an extension with a certain distinctive ring. If call arrives via
pstn line 2, use
Would like to deploy * in a small help desk environment (five to ten
people) using call queues and some sort of CTI interface to pop Remedy
screen data in front of the help desk person receiving the call. Data
to be popped would be based on CallerID.
Anyone doing something similar?
Anyone
Hello,
Thanks very much for help.
To install driver for LineJack I need kernel source.
I have debian, and I installed from apt-get install kernel-source.2.4.20 but
while
it make ./configure it still asks me for the kernel source. What can be
wrong ?
-- Bart
- Original Message -
From:
On Tue, Sep 16, 2003 at 11:10:33AM +0200, Jean-Marc V. Liotier wrote:
On Mon, 2003-09-15 at 11:52, Klaus-Peter Junghanns wrote:
i dont think that the Eicon Diva Server 4BRI's NT mode feature will
work with linux/capi. I think the feature in the driver is for their
PRI cards (where
Hello Barry.
I was searching the archives and google for information on connecting to a
proprietry PBX and came across an response from you dated Oct 31st, 2002.
Quite good information.
Since then, has more information become available ?
As in, what PBX's have been successfully connected, and
-Original Message-
From: Bartosz Jozwiak [mailto:[EMAIL PROTECTED]
Sent: Tuesday, September 16, 2003 11:53 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] LineJack + Asterisk HELP!
To install driver for LineJack I need kernel source.
I have debian, and I installed from
I upgraded my 7960 firmware to ver 4.4. I now can't make any calls and
I get errors (retrans_packet) on call on the console maximum retries
exceeded. And ideas?
Thanks
Kevin
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[EMAIL PROTECTED]
Yes I fixed it thanks.
But I have another problem. I am not so good with linux... so sorry If I am
irritating...
this is what i got:
bmtst:/usr/src/ixj-1.2.1# modprobe ixj
/lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: unresolved symbol
try_inc_mod_count_Rsmp_e6105b23
I upgraded my 7960 firmware to ver 4.4. I now can't make any calls and
I get errors (retrans_packet) on call on the console maximum retries
exceeded. And ideas?
Kevin,
I upgraded to identical 7960's to v4.4, one upgraded fine and the second
had several issues. There has been a few
On Tue, 2003-09-16 at 18:05, Louis-David Mitterrand wrote:
I am using the Diva 4BRI daily with our * and indeed it does support NT
mode on a port by port basis
Good news : I was not 100% sure about that.
is with the open-source Melware drivers from http://mmm.melware.de.
Very nice. I did not
At 07:52 PM 9/14/2003, you wrote:
Any chance of getting this feature added (preferrable as another option
on each mailbox setting in voicemail.conf (after the pager # maybe))? I
know it could be hacked, but I am trying to avoid those type of
improvements. :)
Asterisk already has an outgoing call
Have your tried setting busydetect=no and callprogress=no in
/etc/asterisk /zapata.conf?
On Tue, 2003-09-16 at 12:19, John Sellens wrote:
| From: Dan Fernandez [EMAIL PROTECTED]
| Date: Tue, 26 Aug 2003 16:30:40 -0400
|
| All of a sudden I am getting the following warning Ring/off-hook in
- Original Message -
From: Shaun Ewing [EMAIL PROTECTED]
Date: Sun, 14 Sep 2003 04:37:25 +1000
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] SJphone DTMF?
- Original Message -
From: Kevin Bockman [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday,
Could someone elaborate on this a little? I wasn't aware that this was
necessary.
Quoting Steve Haehnichen [EMAIL PROTECTED]:
I'm a newbie, not afraid to admit it. I tried to install my TDM400P
without the auxilliary (drive) power connection and got this failure
when loading the wcfxs
Zara Trousk wrote:
( Nobody explained me properly why the code was not developed but as you know Asterisk
is Digium and
Digium makes voice boards, so... In other words, what they are saying is: Buy Digium.
I think that's unfair. Asterisk is Open Source - everyone's free to add or change
stuff
Hi Rich,
We have done this before. We basically developed a small client that
sits on every machine and communicates with * to get information about
an incoming call. Contact me off-list and I will be glad to tell you
more about the entire solution.
-Original Message-
From: [EMAIL
Hi Rich,
We have done this before. We basically developed a small client that
sits on every machine and communicates with * to get information about
an incoming call. Contact me off-list and I will be glad to tell you
more about the entire solution.
Actually you might be surpised that there are
The way it was explained to me is that the voltages required to generate the
ringer singals is quite high, so an external power feed is required.
Tom Walsh
::Could someone elaborate on this a little? I wasn't aware that this was
::necessary.
::
::Quoting Steve Haehnichen [EMAIL PROTECTED]:
::
::
On Tue, Sep 16, 2003 at 03:27:44PM -0300, Paulo Mannheimer wrote:
Hi Rich,
We have done this before. We basically developed a small client that
sits on every machine and communicates with * to get information about
an incoming call. Contact me off-list and I will be glad to tell you
more
Yes, Please share.
On Tue, Sep 16, 2003 at 03:05:33PM -0400, Yifang Dai wrote:
On Tue, Sep 16, 2003 at 03:27:44PM -0300, Paulo Mannheimer wrote:
Hi Rich,
We have done this before. We basically developed a small client that
sits on every machine and communicates with * to get information
Sure, here it it goes.
We first developed a small client that sits on a Windows machine taskbar
(sorry guys, but customer had only windows machines ... Hehehe). Upon
boot, the client is loaded and communicates with the * server telling
its IP address and extension number.
When a call is about to
| From: Eric Wieling [EMAIL PROTECTED]
|
| On Tue, 2003-09-16 at 12:19, John Sellens wrote:
| I'm having the same problem with current asterisk versions - asterisk
| sees the FXO line ringing, and claims to answer it, but doesn't
| actually pick up the phone:
|
| Have your tried setting
The way it was explained to me is that the voltages required to generate the
ringer singals is quite high, so an external power feed is required.
Technically, old style ringers require about 90 volts AC, with multiple
ringing frequencies ranging from 20 hz to about 66hz (mostly related to
the
Title: Installation Configuration Questions
I am new to Asterisk (and phone systems for that matter) and was looking for guidance.
My company is looking at installing a new phone system/PBX in our new location and I am trying to convince them to go with the Asterisk PBX. I see on
On Tue, 16 Sep 2003, Bartosz Jozwiak wrote:
Thank you very much for reply.
Yes I want to make LineJack card only for incomming calls.
It will be just for test while I am waiting for X400P from Digium.
X400P? a secret 4-port fxo?
-Dan
___
Same here... Works great once you get the little bugs worked out.
Brian.
- Original Message -
From: Michiel Betel [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, September 16, 2003 10:09 AM
Subject: RE: [Asterisk-Users] Cisco Gateways
I'm using cico's with SIP... And it works
Hi
As the manual states that Dialogic channel is provided as an add on per
price.
What does it cost and how can one buy it. Hasn't anyone been able to make
a 3rd party dialogic channel using GlobalCall.
I do have a couple of dialogic boards that I would like to use, I dont
want my old
First off. NO HTML EMAIL!
Now to your questions.
On Tue, 2003-09-16 at 14:33, Dana Rawson wrote:
I am new to Asterisk (and phone systems for that matter) and was
looking for guidance.
My company is looking at installing a new phone system/PBX in our new
location and I am trying to
Steven Critchfield wrote:
Pattern matching.
exten = _*NN*NN*,1,SetVar(ANI=${EXTEN:2:10})
exten = _*NN*NN*,2,SetVar(DNIS=${EXTEN:13:10)
exten = _*NN*NN*,3,Goto(DifferentContext|${DNIS}|1)
Thanks, this looks like it should work!
You disappoint me. You appear to be a somewhat knowledgeable person
since you use pine, but then you don't figure out how to start a new
thread.
On Tue, 2003-09-16 at 16:54, [EMAIL PROTECTED] wrote:
Hi
As the manual states that Dialogic channel is provided as an add on per
price.
What
Am Die, 2003-09-16 um 18.05 schrieb Louis-David Mitterrand:
I am using the Diva 4BRI daily with our * and indeed it does support NT
mode on a port by port basis: when you configure the card initially you
are specifically asked whether you want ports in TE or NT mode. And this
is with the
Steven,
Thanks for the response.
First off. NO HTML EMAIL!
Sorry. Thought Rich text was acceptable.
Well we can start by dealing with your users. You can go with analog
phones. On the analog side anything over a SMALL
handful need to slide up to T1 + channel banks. T1 is 24 channels,
Hi
Looks like you did not do a make install after compiling the drivers, and
it is still loading the stock kernel ixj.
Please try doing a make install in the ixj-x.x.x source directory.
Hope that helps
On Tue, 16 Sep 2003, Bartosz Jozwiak wrote:
Yes I fixed it thanks.
But I have another
On Tue, 2003-09-16 at 15:34, Dana Rawson wrote:
Well we can start by dealing with your users. You can go with analog
phones. On the analog side anything over a SMALL
handful need to slide up to T1 + channel banks. T1 is 24 channels, and
therefore you need to get 5 of these to cover
your
Hi
I would be interested in finding out about your solution, i can send you
and email offline if you want to, but if you dont have much to hide, it
may be better to post it here.
On Tue, 16 Sep 2003, Paulo Mannheimer
wrote:
Hi Rich,
We have done this before. We basically developed a
Sorry for disappointing you, (or actually breathing, probably that might
bother you too). Im lazy, actually im so lazy I will ignore most of your
email. Please do not lecture me, since I am stupid I will not understand
your comments anyway!!
Basically, there are a lot of disconnect between
Speaking as former Dialogic/Bayonne user who was frustrated for months with
Dialogic's complexity and months of initial testing with GlobalCall only to
use their many-years-old and very complex base Dialogic drivers(and
eventually scrapping it all for Digium/Asterisk and being up one week
later),
Title: Adpcm, 6KHz codec
Is there a way to play adpcm, 6KHz in asterisk? If yes, where can we get this codec?
Thank you.
Alex Zarubin
Please rest assure that I have been following the * development for a
while and understand the value the Digium hardware gives me vs any other
vendor. Most of the people on this list probably know whats good for
everyone else, but I like to find out for myself (I am not a CNN junky).
Now the *
6KHz != 6kbps
bkw
On Tue, 16 Sep 2003, Alex Zarubin wrote:
Is there a way to play adpcm, 6KHz in asterisk? If yes, where can we get
this codec?
Thank you.
Alex Zarubin
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Asterisk-Users mailing list
[EMAIL PROTECTED]
*CLI NOTICE[81926]: File chan_sip.c, Line 5144 (handle_request): Unknown SIP
command 'NOTIFY' from '192.168.1.1'
I think my settings on the budgetone (192.168.1.70) might be wrong as I'm
getting this message in the console.
Any hints/tips ?
Regards...martin
Title: RE: [Asterisk-Users] Adpcm, 6KHz codec
What I need is adpcm algorithm (4 bits per sample) with 6 KHz sampling rate
(6000 samples per second). This is 24kbps.
The standard (cvs) asterisk adpcm plays at 8 KHz, i.e. 32kbps.
Alex Zarubin
-Original Message-
From: Brian West
Sure, here it it goes.
When a call is about to be transferred to that extension, an * AGI sends
the client all information that was programmed to be transferred. We had
to patch app_queue.c to do this (giving it the ability to call an AGI
just before a call is being answered by a queue member
I'm not sure I understood your question.
As far as I know, listening to the manager interface wouldn't give me
enough information. At the moment where the call is transferred, the
client has already browsed through a couple of menus, setting some
variables. The AGI sends the content of these
Ok, Skinny gurus. (btw, I'm super pleased to see development happen on
this).
Thoughts on this??
I added this context to my skinny.conf:
[ppc]
device=SEP00022D494F2A
context=employees
line = 50 ; Dial(Skinny/[EMAIL PROTECTED])
I've downloaded the 30 day Window eval of VTGO!
What kind of PCs are people using? For a system with
100 users I'd be thinking of some rack mount boxes
with dual power suplies and mirrord disks. Ideally
the P/Ses and disk could be swapped out without
opening the box. each P/S plugs into it'sown UPS.
It's good to have two or three boxes
Is there anyone out there with a custom client softphone and is
interested in integrating both solutions?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of TC
Sent: September 16, 2003 3:53 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] call center
Yes, setting callprogress=no fixed the problem.
Thanks to everyone.
- Original Message -
From: Martin Pycko via RT [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, September 16, 2003 6:43 PM
Subject: [digium.com #860] Fw: x100P: Ring/off-hook in strange state 6 on
channel1
it
Hi,
Try as I might, I can't get hangups detected on a Zap channel with loop start
lines. So, after someone leaves a voicemail and then hangs up, Asterisk
doesn't know it, exits VoicemailMain2, and loops back to the corporate
greeting, tying up the line even though the outside caller has hung
Does any have a copy of the 30202 7960 firmware?
Thanks
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[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Hello,
I made install.
Why I am getting this.
My linux is Debian.
--
Hi
Looks like you did not do a make install after compiling the drivers, and
it is still loading the stock kernel ixj.
Please try doing a make install in the ixj-x.x.x source directory.
Hope that helps
On Tue,
Hi Tarun,
Does anyone has experience of deploying Asterisk based
VoIP solution in a universitywide campus. We are at
present investigating various Soft PBX for this
purpose from different vendors Digium,Snom, Pingtel...
you might want to have a look at
http://graphics.cs.uni-sb.de/VoIP/
[EMAIL PROTECTED] wrote:
All I want to know is how, where. And is there any other third party
channel for Dialogic is available.
Now I dont see anything wrong with my question!!.
Congratulations on learning how to start a new thread properly. :)
As Stephen helpfully stated (and you seem to
I have 2 xp100's and one TDM400P.
I've plugged a phone into the tdm40b, and when i take it off hook sometimes
i get a dialtone, other times i get the message
Power alarm of module2, resetting spit out to the console from the wcfxs
driver
does anyone know what this could be? I've tried two
iaxclient.sourceforge.net is the home of Steve Kann's crossplatform port of the
iax library.
iaxComm is a client written in c++ using wxWindows. There is a Win32 binary on
the site. I think that it should be compilable on Linux and MacOSX, but can't
test it.
Feedback is welcome.
I'm not sure I understood your question.
typos b4 :)
As far as I know, listening to the manager interface wouldn't give me
enough information. At the moment where the call is transferred, the
client has already browsed through a couple of menus, setting some
variables. The AGI sends the content
[ Sorry, I incorrectly copied some Reference headers into this post
and tacked it onto the wrong thread. -Steve ]
So far, I have been able to receive incoming iaxtel calls via my
assigned 1-700-xxx- number, but only when using md5
authentication in iax.conf:
[iaxtel]
type=user
-= On Sun, 14 Sep 2003 07:25:42 -0600, Rich Adamson [EMAIL PROTECTED] said:
1. music on hold does not require any exten = 302,1,... commands
at all. Once moh is defined in the musiconhold.conf file, it seems
to work on all extentions without additional per-extension
definitions.
Thanks for
I just updated to the new CVS and now I am getting the following error from
chan_zap (modprobe wcfxo works fine):
WARNING[16384]: File chan_zap.c, Line 577 (zt_open): Unable to
specify channel 1: Device or resource busyERROR[16384]: File
chan_zap.c, Line 4781 (mkintf): Unable to open
- Original Message -
From: Steve Haehnichen [EMAIL PROTECTED]
MoH works fine with my (local) Grandstream phones. It's just the
direct-dialed music-only extension that does not. I can live with
that. I'll see if I can poke around a bit more in the configuration
relating to
Rich Adamson a écrit :
Would like to deploy * in a small help desk environment (five to ten
people) using call queues and some sort of CTI interface to pop Remedy
screen data in front of the help desk person receiving the call. Data
to be popped would be based on CallerID.
Anyone doing something
Ernest,
I hadn't thought of doing that, though having that added protection would
be nice. However, what I'm trying to do it have an incoming call at my home
number follow me to my cell phone for selected numbers -- Since I already
have three way calling, I'd like get Asterisk to essentially
Please try to tell me exactly what steps you did, and I will try to help
you. It seems to be a non-asterisk issue so you can just email me
directly. Please use a subject line or the spambouncer may not like it.
Regards
F
On Tue, 16 Sep 2003, Bartosz Jozwiak wrote:
Hello,
I made install.
[EMAIL PROTECTED] wrote:
Does any have a copy of the 30202 7960 firmware?
Thanks
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[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
It is on Cisco's FTP server if you have a CCO account.
[EMAIL PROTECTED] wrote:
I'm still stuck on this. The * is on a private network IP 192.168.0.7
and the 7960 is on 192.168.0.6. When I show 'sip peers' the 7960
appears to be registered although I can't make any calls and still get
the packet retries. I have also checked and re-checked the
Thanks for the kind reply, and sorry if Ive been meeing up the threaded
mail readers.
But this is just half of the story, bacause besides $15 charge, that
channel (just like quicknet) only supports incoming calls, but a man must
know his limitations!!
Regards
On Wed, 17 Sep 2003, Alastair
On Tue, 2003-09-16 at 20:27, Dan Fernandez wrote:
I just updated to the new CVS and now I am getting the following error
from chan_zap (modprobe wcfxo works fine):
WARNING[16384]: File chan_zap.c, Line 577 (zt_open): Unable to
specify channel 1: Device or resource busy
snip
our university is going to roll-out 1000 lines in the next few months. we
are going to deploy either quintum, audiocodes, vg248 or ata-186 around
campus (and soon maybe grandstreams and cisco). we have a cisco callmanager
to do the call routing and asterisk for voicemail and protocol conversion.
hi!
I've got a asterisk system
running with around 50 per calls per minute. I've connected * to internal
pabx and outside telecom using E1 (ISDN pris). Sometimes calls disconect
abnormally. Is this something we have to live with or is it a bug in CVS
code ?
denzel.
What I need is adpcm algorithm (4 bits per sample) with 6 KHz sampling rate
(6000 samples per second). This is 24kbps.
Are you sure you're not thinking of 3 bits per sample 8000 Hz ADPCM (also
2400kbps)?
Mark
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