Bill Reid wrote:
> I have had a similar problem upgrading to .24 . Sipura support
> suggested
> using tftp which worked successfully.
>
> On the tftp server you use the URL
>
> http://aaa.bbb.ccc.ddd/upgrade?/path_name/spa.bin
>
> where aaa.bbb.ccc.ddd is the IP address of the Sipura.
>
> Do n
I'll look at it tomorrow, what url are you using? standard asterisk syntax?
Stig Andersson wrote:
Hi again,
Installed your new release today (after the sip bugfix).
Now SIP registers OK with asterisk, but calling fails...
Firefly says: Couldn't start call.
Asterisk in SIP debug mode shows th
Hi again,
Installed your new release today (after the sip bugfix).
Now SIP registers OK with asterisk, but calling fails...
Firefly says: Couldn't start call.
Asterisk in SIP debug mode shows the registration, but shows no response
when firefly tries to call.
Using NO stun, asterisk and Firef
Just a quick update, there's was a problem with SIP - if you were
getting SIP registration failed, grab the new version.
(http://www.virbiage.com/firefly/download/firefly-dev.exe)
thanks for the feedback about this bug,
Adam
Adam Hart wrote:
I've been sitting on this release for a week so
Hello!
On Tue, 16 Mar 2004, Michael Welter wrote:
> I have the current version of sethdlc, but none of the protocol options
> (hdlc, hdlc-eth, cisco, ppp) seem to work. Is there a preferred
> protocol for a direct connect?
>
> sethdlc hdlc0 hdlc
That's wrong!
Either you make use too old seth
I notice that CONF_SIZE in asterisk/apps/app_meetme.c is set to 160.
CONF_SIZE is used to set up a buffer of some sort. Can anyone explain to
me why the value of 160 was chosen? Seems kinda arbitrary.
--
Tracy Reed The attachment is a digital signature.
http://copilotconsultin
Mar 16 22:10:52 WARNING[9226]: chan_iax2.c:5309 socket_read: Received
mini frame before first full voice frame
Ok, I found the problem. The error message is not very descriptive but
I was able to fix it by playing around.
I had the register line in iax.conf below my peer definitions, and I
see
If the definition of Insane is doing the same thing over and over again,
but expecting different results.. I must be falling into this category
today... Ok let me lay out the configs first
extension.conf
[macro-stdexten]
exten => s,1,DBget(temp=CFIM/${ARG1})
exten => s,2,Dial(Zap/g2/${temp}
P.S. linux-2.4.25 on Debian
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I'm trying to run both voice and data over a PRI T1 circuit between two
* boxes (using a crossover cable). I'm using asterisk and libpri from
today's cvs. I can establish voice calls, but I can't pass data.
I have the current version of sethdlc, but none of the protocol options
(hdlc, hdlc-et
check context perhaps try include in the extensions.conf
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Try adding something similer to this in the default context of your
extensions.conf:
exten => ,1,Dial(SIP/user1,20, tr)
exten => ,1,Dial(SIP/user2,20, tr)
Always remember that device names - peers, users, friends - in any channel are
not the same as extensions. You call extensions (defined in ext
Adam Hart wrote:
Could be simple packet loss, the first frame (the full frame) gets loss
and retransmitted but in the mean time, the second frame (a mini frame)
arrives.
I guess that could be the case. My * box is behind NAT. I don't think
that would reliably cause this issue though.
I get t
On Tue, Mar 16, 2004, Barry Fawthrop thus spake:
>check NAT setting try taking it out of sip.conf, that worked for me
Nope. My bad-shoulda said up front that I've tried both with and
without nat=yes in sip.conf, no difference in symptoms.
Regards,
Ed Hintz
[EMAIL PROTECTED]
__
Chris Higgins wrote:
I'm trying to call my * box via iaxtel to diagnose my previous problem
(see earlier email today) and I get the following message spewing to
my console:
Mar 16 22:10:52 WARNING[9226]: chan_iax2.c:5309 socket_read: Received
mini frame before first full voice frame
It looks
check NAT setting try taking it out of sip.conf, that worked for me
Barry
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Looks like there was talk of Dialogic support two years ago, but I have
been unable to find furthor info. (I'm interested in the 'ol D/41D, and
infamous VOIP Blaster support).
I've gotten the fobbit app to run under FreeBSD, but its a standalone app,
Asterisk doesnt seem to be able to use it in a
I'm trying to call my * box via iaxtel to diagnose my previous problem
(see earlier email today) and I get the following message spewing to my
console:
Mar 16 22:10:52 WARNING[9226]: chan_iax2.c:5309 socket_read: Received
mini frame before first full voice frame
It looks like this problem has
Matthew Marlowe wrote:
(reposted to be in text format, sorry. :))
The FT201 is currently being manufactured and will be available shortly!
The retail price will be $129.95 USD...
http://www.virbiage.com/products/lanphones.php
Let me clarify the FT 201 situation, the current ETA is 8 weeks. Th
Running * with default config files except for sip.conf. Any call made is
dropped 5 seconds after connection, with the following messages:
Mar 17 16:37:41 WARNING[1009461760]: chan_sip.c:495 retrans_pkt: Maximum
retries exceeded on call 6C94C1B1-77C4-11D8-91FB-
[EMAIL PROTECTED] for seqno 48221 (R
Eric Wieling wrote:
The FT201 is currently being manufactured and will be available shortly!
The retail price will be $129.95 USD...
http://www.virbiage.com/products/lanphones.php
The web page does not say:
1) how many call appearances does the phone has
It can present 5 calls and you can
On Tue, 16 Mar 2004, Scott Stingel wrote:
> Hi Andrew-
>
> The "unknown error 500" and the frame rejects are somewhat normal - I get
> thousands of these in a busy IVR system. The underlying cause for these, I
> think, is that your processor occasionally does not keep up with the frame
> transmi
Back in January I started having a problem with my Sipura (and there was
at least one other on the list with the same problem) that if I answer
an incoming call (via X100P) on line 1 of my Sipura, the caller cannot
hear any voice from the internal extension. If the internal user puts
the exter
On my system, "show application dial" doesn't mention a "d" option.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
Sent: Tuesday, March 16, 2004 8:09 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Does anyone have faxes working well w
I'm TRYING to be a reseller if that helps... Hasn't happened yet. :-/
As soon as I get more information, I'll let you know.
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Eric Wieling
> Sent: Tuesday, March 16, 2004 9:27 PM
> To: Asterisk Use
> The FT201 is currently being manufactured and will be available shortly!
> The retail price will be $129.95 USD...
>
> http://www.virbiage.com/products/lanphones.php
The web page does not say:
1) how many call appearances does the phone has
2) does firmware costs extra
3) does it come with a p
This really does seem too good to be true.
On Tue, 2004-03-16 at 19:36, Matthew Marlowe wrote:
> The FT201 is currently being manufactured and will be available shortly!
> The retail price will be $129.95 USD...
>
> http://www.virbiage.com/products/lanphones.php
___
"show application dial"
On Tue, 2004-03-16 at 19:49, Jim Sneeringer wrote:
> Thanks so much. This sounds encouraging.
>
> I can't find a description of the "d" option. What does it do? Do I just put
> a "d" in the extension? Right now I have this:
>
> exten => fax,1,Dial(Zap/8)
>
> and I'm e
Title: spandsp in process_baud()
Hi,
Steve, if you want me to get more info from this coredump, please, let me know.
(gdb) bt
#0 0x408184e5 in process_baud () from /usr/local/lib/libspandsp.so.0
#1 0x40818853 in v29_rx () from /usr/local/lib/libspandsp.so.0
#2 0x4081215f in fax_rx_proce
Thanks so much. This sounds encouraging.
I can't find a description of the "d" option. What does it do? Do I just put
a "d" in the extension? Right now I have this:
exten => fax,1,Dial(Zap/8)
and I'm envisioning changing it to
exten => fax,1,Dial(Zap/8/d${EXTEN})
Is that right?
-Ori
> > > I want to build in my small company little PBX with asterisk.
> > > I have one ISDN BRI link with DDI preselection and a couple of analog
> > > phones.
> > >
> > > So, my problem is, I'm lost in lots of abbreviations, and I have no
idea
> > > which pci card to select.
> > >
> > > I know, I ne
(reposted to be in text format, sorry. :))
The FT201 is currently being manufactured and will be available shortly!
The retail price will be $129.95 USD...
http://www.virbiage.com/products/lanphones.php
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[EMAIL PROTECTED
The FT201 is currently being manufactured and will be available shortly!
The retail price will be $129.95 USD...
http://www.virbiage.com/products/lanphones.php
Derek Bruce wrote:
depends on which Cisco softphone you are refering to... they have a few
different versions... including an NBX version which will not work with
Asterisk...
- Original Message -
From: "Tim Sailer" <[EMAIL PROTECTED]>
To: "Asterisk Users" <[EMAIL PROTECTED]>
Sent: Tuesday,
Hi All,
I have posted before asking for a Connect message sent from Zap
(ISDN/PRI - by *) when receiving a call (incoming) and dialing to
another extension. To clarify the situation, I will describe the problem:
1) My * box is connect to a Cisco (E1-ISDN/PRI) using a crosscable.
2) Sending a c
Have a HP Officejet combo working OK with inbound fax
detection, receiving faxes, and sending them
using X100p and TDM400P, I just make sure I use the
"d" option on the dial string, for outbound and also
inbound on the FAX detection.
have the "d" option on the internal fax redirection,
and have set
Ok.. After upgrading the PIX to version PIX6.3(3). I can register the
phone, but I am having related issue of sorts... Here's the low down..
The outside interface of the PIX is doing PAT. And I have one to one
NAT translation for the * Server... But if I configure everything this
way... I get an
I have a machine with no zaptel hardware in it running * (I also have
two other machines running * both with X100P cards).
In the machine with no zaptel hardware, I had everything running
perfectly, until I wanted to add conference rooms with MeetMe.
I recompiled the zaptel package with ztdummy
Does anyone know of any FAX software that will work with asterisk?
signature.asc
Description: This is a digitally signed message part
is this fixed on cvs -stable branch??
Miguel
On Tue, 2004-03-16 at 23:02, Andres wrote:
> Miguel Cavazos wrote:
>
> >hello guys heres my setup i have 2 asterisk servers in 2 different
> >houses sipura ulaw --- asterisk ---> iax2 (ilbc) ---> asterisk -- sipura
> >ulaw, this is my setup but when i
Carey -
OK, here it is (and please don't make fun of my poor perl programming
skills. My philosophy is if something works and it's stupid, then it
works)
First, make sure your Polycom phones are setup as in
http://www.voip-info.org/wiki-Polycom+auto-answer+config
Then, copy this perl script i
Andres wrote:
Senad Jordanovic wrote:
Hi,
Essentially these are general issues I have with Sipura SPA 2000:
* If SPA 2000 is behind NAT, calls are not hanged up when receiver
is replaced. I think asterisk does not get hung-up signal from
SPA so called party user agent is ring
On Tue, Mar 16, 2004 at 02:16:01PM -0600, Jim Sneeringer wrote:
>
> Why does Asterisk not work as well with fax modems as any other phone
> system?
>
Because Asterisk does TDM in software. It's handled by 64 bits chunks,
instead of bit by bit on TDM hardware.
It has two consequences :
- if for
Jim Flagg wrote:
Firefly's Protocol Support now is:
Voip Protocols: SIP, IAX
Codecs: ulaw, alaw, iLBC, GSM, G.729 (via DLL)
Sounds good.
Any plans for Speex codec support?
___
Adding it this week, along with some bug fixes
__
Miguel Cavazos wrote:
hello guys heres my setup i have 2 asterisk servers in 2 different
houses sipura ulaw --- asterisk ---> iax2 (ilbc) ---> asterisk -- sipura
ulaw, this is my setup but when i call the other sipura i can listen
like click click click click it doesnt seem a bandwidth issue becau
One was posted to this mailing list TODAY.
On Tue, 2004-03-16 at 12:32, DanJr wrote:
> Anyone know of any software Fax program that will work with asterisk
--
Eric Wieling <[EMAIL PROTECTED]>
BTEL Consulting
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Anyone know of any software Fax program that will work with asterisk
signature.asc
Description: This is a digitally signed message part
Whats the script doing.. Is the script failing...?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Vikram
Rangnekar
Posted At: Tuesday, March 16, 2004 2:35 PM
Posted To: Asterisk User Group
Conversation: [Asterisk-Users] AGI test script
Subject: [Asterisk-
depends on which Cisco softphone you are refering to... they have a few
different versions... including an NBX version which will not work with
Asterisk...
- Original Message -
From: "Tim Sailer" <[EMAIL PROTECTED]>
To: "Asterisk Users" <[EMAIL PROTECTED]>
Sent: Tuesday, March 16, 2004 2:
Darrin Johnson wrote:
Hello all,
I do not have a Zaptel card, but still wanted to utilize the conferencing
capabilities in Asterisk. I am trying to find a site to download the
required usb-uhci module, but have not had much luck. Can anyone point me
in the right direction?
Thanks,
Darrin Johns
That is how I upgrade my sipura. I believe I learned that from the
manual. I know they update their manual occasionally. Perhaps you have
an old version.
On Tue, 2004-03-16 at 13:44, Bill Reid wrote:
> I have had a similar problem upgrading to .24 . Sipura support suggested
> using tftp which wor
exten => 666,1,Answer
exten => 666,2,AGI(agi-text.agi)
exten => 666,103,Hangup
iwhy is that not working any idea. Does answer need to be there or does the
AGI script answer the call.
--
regards
Vikram (http://www.vicramresearch.com)
___
Asterisk-User
Title: Message
Sure... I will call SIpura..
Thanks
for the info!
Ta
SJ
Senad Jordanovic wrote:
Hi,
Essentially these are general issues I have with Sipura SPA 2000:
* If SPA 2000 is behind NAT, calls are not hanged up when receiver
is replaced. I think asterisk does not get hung-up signal from
SPA so called party user agent is ringing until timeo
When I send or receive faxes with a modem, the faxes are stretched and have
missing information.
Here are the things I have tried so far:
- verified that the 2.4 GHz P4 is not overloaded
- tried different Asterisk channels for both FXO and FXS
- turned off echocancel
- bumped up rxgain and tx
On Tuesday 16 March 2004 12:54, Darrin Johnson wrote:
> I do not have a Zaptel card, but still wanted to utilize the
> conferencing capabilities in Asterisk. I am trying to find a site
> to download the required usb-uhci module, but have not had much
> luck. Can anyone point me in the right direc
Title: rxfax as of 03-16, crash in libtiff-3.5.7-11
Hi,
RH9 (2.4.22 #4 SMP)
The 10-21-2003 release of softfax had it's problems but didn't crash in libtiff-3.5.7-11.
The 03-16-2004 release does.
Questions:
1. Does it make sense to upgrade to libtiff-3.6.0?
2. We don't use the latest sour
I was given an eval of the Cisco softphone to try out. Has anyone
gotten this to work with * yet?
Tim
--
><
>> Tim Sailer >< Coastal Internet, Inc. <<
>> Network and Systems Operations >< P
I have had a similar problem upgrading to .24 . Sipura support suggested
using tftp which worked successfully.
On the tftp server you use the URL
http://aaa.bbb.ccc.ddd/upgrade?/path_name/spa.bin
where aaa.bbb.ccc.ddd is the IP address of the Sipura.
Do not know why these instructions are not
Are you natted (behind a NAT screen)?
Do you have any other SIP devices connected to the same network segment as
your WiFi Access Point? Are they working correctly?
I would try turning off the canreinvite (set = 0) and, if there's a NAT
between you and the Asterisk, turn on nat support (nat = ye
How do I do this
1) ZAP-> * -> IAX(1) --> IAX(2) -> DG104S --> Handset
2) No Answer on Handset
3) Back to IAX(1)
4) IAX(1) tries a cell phone
5) Still no Answer
6) Local * Voicemail.
I have 1 working, and I had 4 working when there was only one box, i.e.
when the handset did not answe
Dear Steve
I have try your very nice app for * but i have found the problem (see
down) when i receive a fax to extension, can you give some more info about
it?
Another question how can i send a fax to a specified number?
Thanks in advance
Dimitri
Hello all,
I do not have a Zaptel card, but still wanted to utilize the conferencing
capabilities in Asterisk. I am trying to find a site to download the
required usb-uhci module, but have not had much luck. Can anyone point me
in the right direction?
Thanks,
Darrin Johnson
Systems Engineer
IS
-邮件原件-
发件人: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] 代表 Steve Underwood
发送时间: 2004年3月17日 0:37
收件人: [EMAIL PROTECTED]
主题: Re: [Asterisk-Users] about voice conference system. I need suggests
>Have you actually tried playing with that software? Its is very limited,
>and doesn't seem have m
At 10:16 AM 3/16/2004, you wrote:
and what would i need to connect asterisk to 2 normal phone lines
You would need two FXO cards from Digium to connect to two Telco lines.
You would need two FXS cards from Digium to connect to two telephones.
Telephone -> FXS + Asterisk + FXO -> Wall (Telco)
If yo
Hi all,
Need to register a SIP Server into Asterisk. But I must, before, to send a
Call-ID to the service provider like: [EMAIL PROTECTED]
Anyone here knows how to implement this?
Thank You
Joao Carlos Moura
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Hi all,
Been trying to call FWD#'s from my astbox in the office without much luck.
I'm able to register with them and originate a call which looks like its
getting connected but no audio passes.
I've got this in sip.conf
[fwd]
type=friend
secret=thepassword
username=theusername
host=fwd.pulver.
I think he means if the script is available(would you be willing to
email it to the list and have someone else host the file for you?)
The application sounds very interesting. I'm sure you would get a lot of
positive feed back and help developing it.
-edwin
-Original Message-
From: [EMAI
Hi Folks,
Took delivery of 3 of these today and am having problems. Pulver tech
support is pretty much non existant so I came here.
Problem is this. Phone configured to talk to * as extn 3405. * shows that
3405 has registered and a sip show peers reveals its ip address etc. Make
a call from the W
Gurus,
I currently have two separate asterisk networks running but am having
problems on one and not the other with identical configs (other than
transport, one if over the internet one is a managed network, ironically the
over the Internet pass thru is working).
I have both systems configured id
+++ Ernest W. Lessenger [16/03/04 10:03 -0800]:
> At 09:52 AM 3/16/2004, you wrote:
> >I need to setup asterisk so that users can dial into asterisk using normal
> >phone lines and and enter a number when prompted then this number should be
> >accessable to a backend app. is this possible in asteri
> >
> Use yast2 from suse and configure isdn:
He's using redhat, not suse :)
btw, I'm using capi with avm under redhat 7.x, 8.x , 9 and
fedora.
What're your problems?
please be more descriptive and we'll help you.
Matteo
--
Brancaleoni Matteo <[EMAIL PROTECTED]>
Espia - Emmegi Srl
__
At 09:52 AM 3/16/2004, you wrote:
I need to setup asterisk so that users can dial into asterisk using normal
phone lines and and enter a number when prompted then this number should be
accessable to a backend app. is this possible in asterisk. any pointer would
be helpfule
Yes, this is possible. Yo
I need to setup asterisk so that users can dial into asterisk using normal
phone lines and and enter a number when prompted then this number should be
accessable to a backend app. is this possible in asterisk. any pointer would
be helpfule
--
regards
Vikram (http://www.vicramresearch.com)
__
Web Service is not available publicly. One of the reasons is that I do not
have a coumputer outsde of firewalls I could deploy it on.
Serge
From: "C. Johnson" <[EMAIL PROTECTED]>
Reply-To: [EMAIL PROTECTED]
To: <[EMAIL PROTECTED]>
Subject: RE: [Asterisk-Users] Web service to start a conference
Il 18:24, martedì 16 marzo 2004, Nick Grindley ha scritto:
> Hi All,
>
> Sorry to be a pain nut I have spent three days with no joy at installing
> the AVM C2 ISDN card. I even tried to install it in a Suse box!
>
> So please (in child like terms for me please) what are the steps and what
> softwar
Hi Greg,
There are a lot of installation instructions that I have to write before it
can run somewhere else then my development system. It requires Apache Axis,
Java Runtime and Eclipse platform. I will write that down at some point.
Then it will be possible to share the code.
There is not much
Hi All,
Sorry to be a pain nut I have spent three days with no joy at installing the
AVM C2 ISDN card. I even tried to install it in a Suse box!
So please (in child like terms for me please) what are the steps and what
software package do I need to install the AVM C2 into the following
config: -
I have a 3802 and I was told by their support the SIP version doesn't support CallerID
from the PSTN side.
Also - mine was "freezing" occasionally on calls. I sent several debugs to technical
support, but didn't get any response.
My experience has not been that pleasant - please let me know wha
Hi,
As far as is know, there is no perfect solution for this problem.
Some people worked out an math algorithm that tries to detect an
answering machine.
But this just bases on the fact, that an answering machine takes more
time to answer (speak the first tone),
than a real person. And therefore it
Title: Message
Hello,
We
currently have 30 of the sipura devices in production in a corporate
environment for over 2 months now. They are all running .31 firmware. We
have had problems with a couple that did not upgrade as described by Senad, but
one call to Sipura tech support and we had
Unfortunatly, no yet. The units are at customs at this moment.
Keep you informed.
Jorge
Michael Graves wrote:
Anyone make progress using the Welltech FXO adapters with *?
Michael
--
Michael Graves [EMAIL PROTECTED]
Sr. Product Specialist www.pix
Serge,
Are you talking about calls to the PSTN? If so, there is no universal
standard for indicating the call is answered be voice mail. This is
possible with GSM; however, you must have SS7 signalling (not directly
available on * at this time.)
I would recommend that you write your app so that
is it publicly available?
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED]
On Behalf Of
> Serge Mankovski
> Sent: Tuesday, March 16, 2004 10:45 AM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Web service to start a
conference
> and voice mail
>
> Hi
>
Title: Message
Hi,
Essentially these are general issues I have with Sipura
SPA 2000:
If SPA 2000 is behind NAT, calls are not
hanged up when receiver is replaced. I think asterisk does not get
hung-up signal from SPA so called party user agent is ringing until timeout
expires or
Hi
I have written a web service that starts a conference call and then monitors
call progress on the manager interface. It works nicely until conference in
a voice mail system. It would be better if I could intercept the fact that
the answering side is a voice mail and not to conference it in. T
ooo, sounds like a bug. Maybe a counter is overflowing somewhere!
Scott M. Stingel
Emerging Voice Technology Inc.
Palo Alto, California and London, England
Email: scott "at" evtmedia.com
URL:www.evtmedia.com
>-Original Message-
>From: [EMAIL PROTECTED]
>[ma
Thomas Gallaway wrote:
Hi
I am working on this since a while now and seem to be stuck. Here is my
issue:
I have a bunch of Budge Tone 101's. Asterisk is set up. 4 Incoming PSTN
lines.
It all works fine just the DTMF is not working. I am not beind a NAT so
the phones
can talk directly to the as
wangji wrote:
>That's what Intel want to do, too. They guys have released a hardware
>emulation software works like an four channel IP board. And they want to use
>only interface board + host CPU instead of Dialogic products, so that they
>needn't use DSP etc.
>
>
Have you actually tried playin
On Tuesday 16 March 2004 08:58, Thomas Haeger wrote:
> if i answer a call on my astbox and go into an AGI script... then
> there is somthing happens.(play music or something like that)...and
> the person who called to the box hangs up the script will never be
> terminated. The process hangs around
Maciej,
and what about "AVM FRITZ! card PCI v 2.0" or "AVM ISDN controller C2" ?
I am using AVM Fritz PCI 2.0 It works fine. (I just put it in service some
days ago. Asterisk runs stable with it)
what does it mean active vs. pasive controller, where is difference?
Am I need active controlle
Tony Mountifield wrote:
In article <[EMAIL PROTECTED]>,
Thomas Gallaway <[EMAIL PROTECTED]> wrote:
Hi
I am working on this since a while now and seem to be stuck. Here is my
issue:
I have a bunch of Budge Tone 101's. Asterisk is set up. 4 Incoming PSTN
lines.
It all works fine just the DTMF
Hi all,
I should have said you will need TIFF version 3.5.7 or later installed
on your machine. 5.5.7 or later is a mistake.
Regards,
Steve
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发件人: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] 代表 Nicolas Bougues
发送时间: 2004年3月16日 22:36
收件人: [EMAIL PROTECTED]
主题: Re: [Asterisk-Users] about voice conference system. I need suggests
> On Tue, Mar 16, 2004 at 10:10:01PM +0800, wangji wrote:
> >
> >I am trying to deploy a con
I just had the same exact problem this morning. The only thing I've done in the last
couple of days is update update zaptel. I rolled back my zaptel to 2/11/04 from
3/8/04. And kept my libpri from 3/8/04. I never had this error before updated. I
had other issues, but not this one.
-sb
On Monday 15 March 2004 17:28, Darren Nay wrote:
> I am just looking into Asterisk as a viable voicemail solution for
> our phone service. In order to use it though I will need to make
> extensions.conf dynamic (ie. Via MySQL). Is this possible?
Not yet, although you can do something similar by
In article <[EMAIL PROTECTED]>,
Thomas Gallaway <[EMAIL PROTECTED]> wrote:
> Hi
>
> I am working on this since a while now and seem to be stuck. Here is my
> issue:
>
> I have a bunch of Budge Tone 101's. Asterisk is set up. 4 Incoming PSTN
> lines.
> It all works fine just the DTMF is not work
Hi,
yes. I answer the call first:
exten => _.,1,Answer
exten => _.,2,SetVar(FPIN=0)
exten => _.,3,AGI(FCall.agi)
exten => _.,4,Hangup
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Areski
Gesendet: Dienstag, 16. März 2004 16:27
A
Hi
I am working on this since a while now and seem to be stuck. Here is my
issue:
I have a bunch of Budge Tone 101's. Asterisk is set up. 4 Incoming PSTN
lines.
It all works fine just the DTMF is not working. I am not beind a NAT so
the phones
can talk directly to the asterisk server.
When I
Polycom 600 phones will do this. They're about $300, including the
power supply.
I wrote a little how-to at
http://www.voip-info.org/wiki-Polycom+auto-answer+config
I also wrote an all-call script for system wide paging which you're
welcome to, if you decide to use these phones.
John
[EMAIL
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