Nathan Goodwin wrote:
So it's a problem with my NAT, that's what I thought.. Ok, I got
another question, with SIP nantive brideing happening, do the CDRs
asterisk keeps still good enough for billing, or are they only good for
the short time asterisk is in the media stream?
That is the one major
So it's a problem with my NAT, that's what I thought.. Ok, I got
another question, with SIP nantive brideing happening, do the CDRs
asterisk keeps still good enough for billing, or are they only good for
the short time asterisk is in the media stream?
Kevin P. Fleming wrote:
Nathan Goodwin wro
Nathan Goodwin wrote:
Does anyone have any ideas how I could fix this, this is sort of
important, if it's just me because of my NAT causing it, would doing so
part forwarding and disable NAT support on asterisk and the Sipura fix
this problem?
It's almost impossible to fix this problem. Here's t
On January 29, 2005 11:29 pm, Brian Dingman wrote:
> This is driving me crazy. I have resorted to using the m option in the
> Dial command just so folks don't hang up. I can't believe nobody else
> is having this issue.
Simple test: try it with another VOIP provider. Throw $5 at a nufone account,
This is driving me crazy. I have resorted to using the m option in the
Dial command just so folks don't hang up. I can't believe nobody else
is having this issue.
Any ideas to work around this?
On Wed, 26 Jan 2005 12:11:42 -0500, Brian Dingman <[EMAIL PROTECTED]> wrote:
> Some more info. Using t
Don't know. Never used AMP.
Lyle
- Original Message -
From: "Chuck Keeter" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Saturday, January 29, 2005 9:16 PM
Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] and Zap Channels
> At 07:57 PM 1/29/200
At 07:57 PM 1/29/2005, you wrote:
;Extra FXS
exten => 112,1,Dial(Zap/11,30,r);
exten => 112,2,Hangup
These lines would go in the extension.conf if I'm not mistaken. and would
define the extension for the zap channel, and assign a extension number.
Is there a way to do it from within AMP? or just
Hi Folks,
I have found references in the wiki and elsewhere to using cisco voice
cards as SIP agenst for * Gateway,
BUT has anyone tried and has a config for using a BRI interface by
itself as a gateway ?
Gary
.
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Asteri
Sipura has implemented auto-answer in version 0.9.5 of the SPA-841
firmware. However, it is implemented via the Call-Info header, which
Asterisk stable doesn't currently support.
The attached patch implments a quick hack to support the Call-Info
header from the Dial() application by way of settin
This is probably going to sound really silly and I must be confused about
it. Maybe someone can set me straight.
I've been tinkering for a while with * and a number of different FXO/FXS
cards, SIP phones, and ATAs trying to get a feel for what works and what
doesn't. In the SIP phone group, I ha
Folks,
Eric is spot-on--both phones are happy with the modified
${CALLERIDNAME} value since I removed the quotes in the call to
SetCIDName(). I've replace my calls to SetCallerID() since I think using
SetCIDName() is cleaner.
Cheers,
On Sat, 29 Jan 2005 11:13:36 -0500, Jim Van Meggelen wrote:
> > does anyone out there made some experience with Varion
> > (www.govarion.com) based E1/T1 cards ?
Their cards work. The only problem about govarion is their delivery
time. The cards are just not shipped as promised. And it's not only
;Extra FXS
exten => 112,1,Dial(Zap/11,30,r);
exten => 112,2,Hangup
- Original Message -
From: "Chuck Keeter" <[EMAIL PROTECTED]>
To:
Sent: Saturday, January 29, 2005 7:01 PM
Subject: [Asterisk-Users] [EMAIL PROTECTED] and Zap Channels
> Hi all,
>
> I've been all over the [EMAIL PROTEC
Thanks
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of dean
collins
Sent: Saturday, January 29, 2005 8:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion; Ing.
Ignacio Ortega A.
Subject: RE: [Asterisk-Users] Eyebeam - asterisk - Messenger
I'm having a problem, I'm not sure if it has todo with the fact that my
phone is behind a NAT or not, but here it is..
My problem is when I call out, my asterisk system routes the call to my
SIP provider, whoever, as soon as the other party answers, asterisk
tries to make a native bridge for th
I was able to find the answer. By using the Answer with a
priority of 1 before the Dial, the call signalling is setup
properly.
On Sat Jan 29 15:51:22 PST 2005, [EMAIL PROTECTED] wrote:
Asterisk experts,
I have been pulling my hair out troubleshooting what appears to
be a Zap channel disconnec
Yep, basically it is a SIP video phone like a grandstream is a SIP voice
phone.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ferguson,
Michael
Sent: Saturday, January 29, 2005 6:44 PM
To: Ing. Ignacio Ortega A.; Asterisk Users Mailing List
Hi all,
I've been all over the [EMAIL PROTECTED] site and all through the configs, and
I can't find how to create extensions that will use the zap channel. Sip
phones I can find, but no way to assign a extension number to the phone I
plug into the TDM400P card.
Am I just missing something?
Hmmm :-)
OK - grab a current RedHat Fedora Core 3 set of ISO's, install - about 2
hours... Do the up2date gig - time depends on your network connection.
Do a cvs based asterisk install - about 30 minutes (more than happy to
share the scripts to do this). Total elapsed time <24 hours max. This
d
[EMAIL PROTECTED] wrote:
> Hello,
> I am new here. I am also somewhat new to telephony and IT,
> however, I am technically adept.
That's probably the most important thing with Asterisk.
> I work for a small but growing non-profit in Colorado and I
> wear many hats. Because I am the only one who h
On Sat, 2005-01-29 at 19:06 -0500, Doug Lytle wrote:
> Jeff Konrade-Helm wrote:
>
> >I'm hoping I can get the initial installation and configuration done in
> >30-40 hours over two weekends and a few evenings. Does this sound
> >reasonable?
> >
> >
>
> Jeff,
>
> To tell the truth, this project
Jeff Konrade-Helm wrote:
I'm hoping I can get the initial installation and configuration done in
30-40 hours over two weekends and a few evenings. Does this sound
reasonable?
Jeff,
To tell the truth, this project is difficult enough for a Linux old
timer to grasp, let alone someone that has n
Asterisk experts,
I have been pulling my hair out troubleshooting what appears to be
a Zap channel disconnect on the TDM side. I am trying to have an
inbound call on a T1 zap channel on the TE405P from our Harris
switch automatically dial a SIP channel. Any help appreciated
very appreciated.
I am curious about Eyebeam so I went to xten's site and read up on it.
I still do not get a clear understanding as to what Eyebeam does. Help
me to understand it:
Am I correct?
It installs on a windows computer?
It connects/registers to my * box?
With a camera attached to my Windows computer I can
Hi!
> The transfer is always successful, but sometimes (lets say bout 50% of
> the times), the MusicOnHold does not "de-attach" itself from the call,
Maybe a good start for an investigation is to try a different softphone
for the operator and see if it makes a difference. Then put Asterisk into
Does anyone have a quad span 410P on a production Dell 1850? Digium
states that there are known incompatibilities with the 7x0 series,
probably related to interrupts.
Also I would like to find out if anyone attempted connecting more than
one of these cards to a dual Xeon system. I am strictly l
On Sat, 2005-01-29 at 20:20 +0100, Zdik Kudrle wrote:
> I called from my home thru Asterisk to my Cellphone. I've picked up the
> phone and set up input channel instead of microphone my TV card. I started
> the TV a listened to the latency cellphone<->TV. Then I said something to
> phone and liste
[EMAIL PROTECTED] wrote:
Folks,
Many thanks to Howard Lowndes who helped solve this problem; I ended
up using SetCallerID() instead of SetCIDName() as Howard suggested. Although
SetCIDName() changed the value correctly, desk set CID displays displayed
either "unavailable" or "out of area" o
Conflict with with codec.
bandwidth=low should be disabled as ulaw is not a low bandwidth codec.
#Joseph
On Sat, 2005-01-29 at 13:13 -0700, Joseph wrote:
> When I try to call out to FWD over IAX2 I get:
> Call rejected by 65.39.205.121: Unable to negotiate codec
>
> I'm using asterisk-1.0.5 (the
Eric Wieling aka ManxPower wrote:
That's really up to your carrier. Many carriers to not allow customers
to set the Caller*ID number. I don't know of ANY that let you set the
Caller*ID name.
Well, pretty much all of them will let you send it over a PRI... it's
just another facility message tha
to me there is a timing error:
The creation time for the file is:
Jan 29 15:30 18-20050129-152954-in.wav
The messages log is showing an error:
Jan 29 15:29:23 WARNING[3015]: Read error on sound device: Resource
temporarily unavailable
It would seems the file is still in use by soxmix when
Hello all, further to Rob's message, could I ask does zaptel.conf or
Zapata.conf need anything further (for caller ID) to allow the setting of
caller ID, as my below E1 pri debug shows that Asterisk seems to be
correctly sending the necessary Q931 instructions to the carrier (Colt) to
set the call
I think you mean
http://iaxclient.sourceforge.net/iaxcomm/iaxcomm-mac-1.0rc1.zip
:-)
Andrew
On 30/01/2005, at 4:02 AM, Michael Van Donselaar wrote:
On Sat, 29 Jan 2005 15:07:48 +0900, Kuniyoshi Murata
<[EMAIL PROTECTED]>
wrote:
Hi,
Is MacOSX version yet to come?
Thanks to Andreas Wrede for the bi
>On Sat, 29 Jan 2005 12:53:11 -0600
> -Original Message-
>From: <[EMAIL PROTECTED]>
>Subject: [Asterisk-Users] RE: Q: Can I over-ride the value of
> ${CALLERIDNAME} ?
>To:
>Message-ID: <[EMAIL PROTECTED]>
>Content-Type: text/plain; charset="us-ascii"
>
>Folks,
>
> Many tha
Hi,
After I boot of the CD ( version 0.4 ) I get following error message:
"CD not found
The centos release 3.3 DC was not found in any of your cdroms drives.
Please insert the CentOS release 3.3 cd and press OK to retry."
Well I just boot of that cd so what is going on?
I have done some searches b
Hi,
I was woredering if you could help me to put into practice this solution.
The idea: Create a IVR-Voicemail
The scene:
PSTN--/6--PBX/12- Internos
|
/4 ports
|
Hello,
I am new here. I am also somewhat new to telephony and IT, however, I am
technically adept.
I work for a small but growing non-profit in Colorado and I wear many hats.
Because I am the only one who has a clue about computers, I am the default
IT person.
Our communications needs are gro
Has anyone written any drivers for ISA bus Dialogic Proline 2v or Dialogic
4 cards? I have a couple of them here gathering dust, I was wondering if I
could use them for Asterisk
Pat
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Asterisk-Users@lists.digium.com
http:/
When I try to call out to FWD over IAX2 I get:
Call rejected by 65.39.205.121: Unable to negotiate codec
I'm using asterisk-1.0.5 (the same settings works fine with *0.9)
I've standard settings in iax.conf
[general]
bindport=4569
register => x:[EMAIL PROTECTED]
[iaxfwd]
type=user
context=fr
I met Sangoma guys this Tuesday, and got a AFT102 for evaluation.
Right now I am in progress to develop a * channel driver for AFT10*
devices.
In that case you will have much more flexibility and to use all their
API.
Steven Critchfield wrote:
On Sat, 2005-01-29 at 11:45 -0500, Jim Van Meg
On January 29, 2005 12:33 pm, Matt Schulte wrote:
> Yah tried that, they won't let me register. They want 50$ for the manual
What? I could download it for free along with all their other manuals less
than a week ago. Are you perhaps asking for a hardcopy rather than just
downloading the .pdf?
Hello Guys a pay in Mexico city a couple of days ago opened a E1 for testings.
wonder what was the concussions?
Did you get the stress test?
did the calls for the callers whent fine?
any comments about R2 in Mexico?
Does enyone have tested teh R2 support of asterisk with a R2 Card in
Option 1
Matt Riddell wrote:
Shoval Tomer wrote:
As far as I know it's not legal to join bellster in Israel.
It means that you're reselling the minutes you buy from the telco
company.
IANAL, but I doubt anyone will view this as selling.
The thing is, it still costs kmoney to terminate calls, even local on
On January 29, 2005 12:31 pm, Matt Schulte wrote:
> We are a voip terminating company, we're using Channelbank with FXS
> modules, Rhino, CAC, etc.. What we're wondering is, is how to would you
> echo cancel a channelbank. Of course we're realizing that cancel'ing on
> the T1 (on Ast) does no good
On Sat, 29 Jan 2005 11:56:56 -0500, dean collins <[EMAIL PROTECTED]> wrote:
> Hi, not sure if it is against the rules to sell second hand equipment in
> here but haven't seen anything against it so here it is.
>From the Digium.com website under mailing lists:
The [EMAIL PROTECTED] list is for bu
On Sat, 2005-01-29 at 00:38 -0700, Joseph wrote:
> When I try to reload configuration *-1.0.5 can not find my sip.conf.
> I don't see anything like:
> == Parsing '/etc/asterisk/sip.conf': Found
>
> WARNING[10301]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on
> call [EMAIL PROTECTED] for
Hello,
I'm looking for a
Cisco/Lucent/Asterisk VoIP SIP guru in the bay area. Full or part
time/consulting.
Please contact me if
you know someone or are interested yourself.
Thanks!
Todd
Houser
(916)
670.1873
___
Asterisk-Users mailing
> On Sat, 2005-01-29 at 15:48 +0100, Zdik Kudrle wrote:
>
> > I'm running Asterisk with HFC-S card connected to HW PBX in my office.
> > When I make a call from home using iaxComm connected to Office
> > Asterisk, the outgoing latency is about 0.25 sec, which is quite OK.
> > But to incoming late
A T1 is a two way transmission media. There is sound going both ways over a
'4 wire' interface. 4-wire means that Xmitt is seperate from Recv. In the
channel bank there is a 4 wire to 2 wire conversion. It's this junction
that introduces echo as some of the 4 wire recv gets feed back into the 4
On Sat, 29 Jan 2005, Matt Schulte wrote:
> Agh, what I meant was the echo is heard from the PSTN side. It seems
> echo canceling on the T1 (going to channelbank) does nothing, I'm
> assuming because the T1 is "digital" and the channelbank is the
> traversal from digital to analog.
Still, echo can
Agh, what I meant was the echo is heard from the PSTN side. It seems
echo canceling on the T1 (going to channelbank) does nothing, I'm
assuming because the T1 is "digital" and the channelbank is the
traversal from digital to analog.
-Original Message-
From: Peter Svensson [mailto:[EMAIL P
Folks,
Many thanks to Howard Lowndes who helped solve this problem; I ended
up using SetCallerID() instead of SetCIDName() as Howard suggested. Although
SetCIDName() changed the value correctly, desk set CID displays displayed
either "unavailable" or "out of area" on incoming calls from my
On Sat, 29 Jan 2005, Matt Schulte wrote:
> We are a voip terminating company, we're using Channelbank with FXS
> modules, Rhino, CAC, etc.. What we're wondering is, is how to would you
> echo cancel a channelbank. Of course we're realizing that cancel'ing on
> the T1 (on Ast) does no good (we thin
[EMAIL PROTECTED] wrote:
> Jim Van Meggelen wrote:
>> [EMAIL PROTECTED] wrote:
>>> [snip]
>>>
>>
>> Your diagram is a bit confusing to me. Still, the use of PRI in the
>> BCM is a good plan. You've avoided using the highly unstable VoIP
>> functions in the BCM.
>>
> Sorry, it got a little jumbl
No, not really, what information do you need?
I have no idea whether it is an asterisk problem,
and adit 600 problem, etc... I was hoping someone
would be able to give me an idea of which config
file to look into, etc... so I wouldn't have to attach
20 config files to my email.
Jon.
On Satur
Hi
I can't find out how to add more links to the Navigation box in
voip-info.org. Can someone help me out, please?
roy
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[EMAIL PROTECTED] wrote:
does anyone know when current HEAD is scheduled to stabilise? Is
there a plan, or is it still "some time in the future"?
I believe I saw an announcement recently that it will start
stabilizing in February, with the goal of releasing 1.1 on the
six-month anniversary of the 1
According to the CAC manual for a Access Bank 1, there is no such option.
The FXS channels adapt according to the signaling bits sent to it over the
T1. ABC should be off for each channel.
Lyle
- Original Message -
From: "Matt Schulte" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing Lis
The current astcc Makefile puts the sound files into the wrong directory.
It uses /usr/share/asterisk/sounds but it should be
/var/lib/asterisk/sounds.
Karl Putz
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Daniel Eboa
> Sent: Saturday, January 29,
Bill Seddon a écrit :
<>
Yes it's working fine for me. Try by deactivate/reactivate IAX on your
account. I use the same config that Gonzalo send.
Mondial Software Limited
020 7043 2795
www.mondialsoftware.com
Click here to view our presentation of Cash Controller showing its
forecasting and au
We are a voip terminating company, we're using Channelbank with FXS
modules, Rhino, CAC, etc.. What we're wondering is, is how to would you
echo cancel a channelbank. Of course we're realizing that cancel'ing on
the T1 (on Ast) does no good (we think?) because the analog conversion
is at the chann
Yah tried that, they won't let me register. They want 50$ for the manual
>:(
-Original Message-
From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED]
Sent: Saturday, January 29, 2005 10:35 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] CAC Access Bank
On January 29, 200
- Original Message -
From: "Robert Rozman" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Friday, January 28, 2005 10:22 PM
Subject: [Asterisk-Users] Integrating with existing 1BRI,6 POTS Panasonic
PBX ?
> Hi,
>
> at the university department we
Which phones do you have? We are using Cisco 7940G phones and I have
been able to do this by modifying the dialplan.xml for the phone to
rewrite numbers as they are dialed to include the "9" in front of
whatever is dialed from the phone. Now you can use the received calls
menus without having
Hello,
I'm looking at building a couple new PRI Gateway boxes using
TE405P cards, and was wondering if anyone has had any experiences (good or
bad) with the Intel SE7210TP1_E motherboards from Intel. General Technics
builds some really nice (and cost effective) 1U servers based on the
b
I got this error when i try to dial:
-- Executing Answer("SIP/8000104-71a3", "") in new stack
-- Executing DeadAGI("SIP/8000104-71a3", "astcc.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/astcc.agi
Jan 29 18:11:37 WARNING[3412]: file.c:475 ast_openstream: File
astcc-
i did evrything you mentioned, i thing is for my eyebeam version, mine is 3002s
what`s yours?
On Fri, 28 Jan 2005 23:10:40 -0500 (EST), Francois Meehan
<[EMAIL PROTECTED]> wrote:
> Thanks Wessel,
>
> You really have to know about that little switch on button, I had 2
> eyebeam connected with the
Hi,
probably I won't be much of help, but I'm also looking for speech
recognition solution. But we're actually looking at two problems:
- one would be so called voice dialing (similar to celular phones) - one
records its own spoken names and speaks them after to call certain person -
this problem
I would recommend using the local trunk and then you just need a context
that will dial out in your extensions.conf. Just put the context name
into the "Peer/Trunk" field on the trunks page. Currently there is not
support in astcc for oh-323. It would be trivial to add but
Darren Wiebe
[EMA
On Sat, 2005-01-29 at 10:36 -0600, Jon Gabrielson wrote:
> I have asterisk connected to an adit 600 with fxo ports.
> When I place a call with asterisk, asterisk will try to dial
> out on the first line even if the first line is already being
> used by someone else. Any ideas on what I'm doing
> w
On Sat, 2005-01-29 at 11:45 -0500, Jim Van Meggelen wrote:
> David Norton wrote:
> > Hi,
> >
> > Currently I only have 1 PRI which I am using for dial-in customers.
> > The line is connected to a Portmaster3. I have never used more than
> > 10 concurrent channels. The calls can be both analog or I
On Sat, 29 Jan 2005 15:07:48 +0900, Kuniyoshi Murata <[EMAIL PROTECTED]>
wrote:
>Hi,
>
>Is MacOSX version yet to come?
>
Thanks to Andreas Wrede for the binary!
http://iaxclient.sourceforge.net/iaxclient/iaxcomm-mac-1.0rc1.zip
___
Asterisk-Users mailin
David Norton wrote:
> Hi,
>
> Currently I only have 1 PRI which I am using for dial-in customers.
> The line is connected to a Portmaster3. I have never used more than
> 10 concurrent channels. The calls can be both analog or ISDN. It
> would be a waste to order another PRI for my Asterisk box. Is
Hi, not sure if it is against the rules
to sell second hand equipment in here but haven’t seen anything against
it so here it is.
I’m upgrading to 2 lines so I have
some spare equipment for sale here. This is an ideal starter pack and will get
you going with 1 line and 1 extension.
On Sat, 2005-01-29 at 15:48 +0100, Zdik Kudrle wrote:
> I'm running Asterisk with HFC-S card connected to HW PBX in my office.
> When I make a call from home using iaxComm connected to Office Asterisk,
> the outgoing latency is about 0.25 sec, which is quite OK. But to incoming
> latency begins on
Does anyone know where I can find the Asterisk Python interpreter?
The one referenced here:
http://www.sineapps.com/news.php?rssid=173
It seems like the site
http://vox.groovy.net/moin/PyAsterisk
is down
___
Asteris
Ok no probs, I thought the whole reason for CVS was so that it saved
downloading an entire copy of the book each time I wanted to update to
the latest version.
Thereby making for easier burden for you guys hosting the docs.
I'm happy to download a pdf each time (unlimited bandwidth available
here
Jim Van Meggelen wrote:
[EMAIL PROTECTED] wrote:
[snip]
Your diagram is a bit confusing to me. Still, the use of PRI in the BCM
is a good plan. You've avoided using the highly unstable VoIP functions
in the BCM.
Sorry, it got a little jumbled
PRI support on the BCM is based on the Norst
On January 29, 2005 11:11 am, Matt Schulte wrote:
> We have an old CAC and we're trying to get groundstart working on it, we
> think it may be a dip switch setting. Does anyone have the config
> settings and/or the manual to config this thing? Any help is greatly
> appreciated :-)
Carrier Access i
Hi All,
After several months using a TDM400p to access my 4 POTS lines I'm
thinking that there has to be a better way. I could order up a BRI from
SBC and ditch the POTS lines altogether.Does anyone have ISDN working
in the US? What are the issues involved with that interface?
Thanks,
Michael
-
I have asterisk connected to an adit 600 with fxo ports.
When I place a call with asterisk, asterisk will try to dial
out on the first line even if the first line is already being
used by someone else. Any ideas on what I'm doing
wrong?
Thanks,
Jon.
__
Eric Wieling aka ManxPower wrote:
The issue happens when you want to use the same username/password for
more than one line. Can't do that (unless the phone has special support
for only registering once for all lines, i.e. the Polycom). Also you
can't have more than one device register with the
On Sat, 29 Jan 2005 11:10:25 -0500, dean collins <[EMAIL PROTECTED]> wrote:
> What should I be using to read this so it doesn't show the XML?
This would have been more appropriate to post to the asterisk-docs
mailing list, but either way.
The documentation is written in DocBook. Whatever is check
On 16:11, Sat 29 Jan 05, [EMAIL PROTECTED] wrote:
> I am looking at setting up redundant servers for my client and I know the
> first step in creating an auto fall back would be to have the FQDN set up in
> such a way that it points to both of the server IP addresses, but, how can I
> make it so
You can get a support login at CAC for free, You can download all the
manuals in PDF from their support site. www.carrieraccess.com
-Matt
On Jan 29, 2005, at 11:11 AM, Matt Schulte wrote:
We have an old CAC and we're trying to get groundstart working on it,
we
think it may be a dip switch sett
Dave Morrow wrote:
> Hi all, I have asterisk installed and working just fine with a couple
> of Cisco IP Phones. I am now ready to pilot connectivity to PSTN and
> am wondering what hardware would be recommended to make minimum
> connectivity to the public telephone network.
Minimum would be an
Jay Wilton wrote:
I just purchased a Cisco 7960 IP phone and I couldn't be
more happier.
It supports 6 lines. You can usually find a good one off
eBay.
I thought asterisk could only register once per ip? guess
i'll go read the wiki for a while longer. :)
No. Asterisk only supports one "line/devi
Currently I have a toll free line set up on my PBX at home, and it is working
great, BUT, when an inbound call comes in, the CID is reporting toll free call.
Does anyone know how I can get around this problem?
Thanks,
Dan
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Asterisk-Users mailing
Stefan Gofferje wrote:
Hi folks,
I have a rather nasty problem. I have set up an asterisk test system
with a Cisco phone, a X-Lite client and so on and did some testing.
To the developers: great work! Hell of great!
However, it seems to me like asterisk puts the Caller ID Number into the
SIP Disp
[EMAIL PROTECTED] wrote:
>>> does anyone know when current HEAD is scheduled to stabilise? Is
>>> there a plan, or is it still "some time in the future"?
>>
>> I believe I saw an announcement recently that it will start
>> stabilizing in February, with the goal of releasing 1.1 on the
>> six-month
We have an old CAC and we're trying to get groundstart working on it, we
think it may be a dip switch setting. Does anyone have the config
settings and/or the manual to config this thing? Any help is greatly
appreciated :-)
Matt
___
Asterisk-User
I am looking at setting up redundant servers for my client and I know the first
step in creating an auto fall back would be to have the FQDN set up in such a
way that it points to both of the server IP addresses, but, how can I make it
so that the phones will (with minimal delay) authenticate wi
[EMAIL PROTECTED] wrote:
> Hi community,
>
> does anyone out there made some experience with Varion
> (www.govarion.com) based E1/T1 cards ?
I have no hands on experience with them, but they are based on the
open-source Tormenta card:
www.zapatatelephony.org
With Asterisk, I think any hardware
Matthew Simpson wrote:
Hello, I have a dial plan that tries to place a call over several
different outbound gateways, like this:
exten => _1X.,1,Dial(SIP/[EMAIL PROTECTED])
exten => _1X.,2,Dial(SIP/[EMAIL PROTECTED])
exten => _1X.,3,Dial(SIP/[EMAIL PROTECTED])
exten => _1X.,4,Dial(SIP/[EMAIL PROT
On Jan 29, 2005, at 16:15, Daniel Eboa wrote:
Hi Daniel,
Would it be possible for you to turn off attaching two image files as
signature replacements to each of your email and maybe use a text
signature instead?
Thanks!
jens
___
Asterisk-Users maili
Pedro wrote:
understood - I use the # sign as well, but some users are not used to
using the # sign so decreasing the timer helps those that may forget
to use the # key.
Properly setting up the dialplan on the SIPura eliminates most of these
issues. Here's my dialplan on my SPA-2k:
([2-7]xxx|9,1
Thank you so much for your help.
I was obviously missing the second ':' after the username
This will help me now in that I only need to do an update before reading
rather than downloading the entire document again.
Having said that I have another question for you. I've downloaded the
file to win
On January 28, 2005 09:16 pm, Ryan Cavanaugh wrote:
> I would like to link the Nortel BCM to * using the a digital trunk card.
> The BCM will continue to service the Tampa location, the * box would
> simply be used to pass extensions over the PRI to another * server in
> Sarasota and for a few SIP
[EMAIL PROTECTED] wrote:
> I am looking at setting up the following configuration and any
> help/input/comments before signing the PRI contracts will be greatly
> appreciated.
>
> PRI Tampa PRI
> Sarasota PRI
> <--> Nortel BCM-->Asterisk<-->Aste
On Sat, 29 Jan 2005 10:46:21 -0200, Denis Galvão - iSolve <[EMAIL PROTECTED]>
wrote:
>Hi Michael.
>
>Any work to support some USB Phones!? The ability to dial using the phones
>keypad!?
Not yet, but I'll probably add suport for the TigerJet phone eventually.
>Thanks.
>
>Denis.
>
>
>Em Sáb 29 Ja
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