On Mon, 2005-05-30 at 18:21, Armin Schindler wrote:
> On Mon, 30 May 2005, Mike Price wrote:
> > On Thu, 2005-05-26 at 23:21, Armin Schindler wrote:
> > > On Thu, 26 May 2005, Mike Price wrote:
> > > > Yes libcapi is installed. Here is a sample of the errors I am getting:
> > > >
> > > > In file i
Hmmmyes but last time I played with my FXO module on the TDM400 could
not detect hangup properly (that is on a London BT line). Has this been
fixed? I keep an eye on the CVS but I have not seen any fixes for that.
Maybe I missed it.
Vassilis
Well, the official line is as Mr. Spencer has
On Mon, 30 May 2005, Mike Price wrote:
> On Thu, 2005-05-26 at 23:21, Armin Schindler wrote:
> > On Thu, 26 May 2005, Mike Price wrote:
> > > Yes libcapi is installed. Here is a sample of the errors I am getting:
> > >
> > > In file included from chan_capi.c:38:
> > > chan_capi_pvt.h:92: syntax er
Hi All
From where we can get the data for
1) ASR on various countries
2) Average Call drop on VoIP
3) Average Call Quality
This we require to get an idea of what types of
problem normally users use to face on voip and what is the average percentage of
those problems.
Pls. help me if
Luki wrote:
OK -- follow up, for the record.
After looking at the patch code, it looks like the documentation is wrong:
[default]
1000:sip-reg-srv-001 => 1000,John Doe,[EMAIL PROTECTED]
should be:
[default]
[EMAIL PROTECTED] => 1000,John Doe,[EMAIL PROTECTED]
... and then extensions.conf can
Hello *'s,
I have question regarding Quintum Tenor AXT800 VOIP gateway can anyone
integrate it with asterisk if anyone what is the scenerio i have
scenerio which is quite simple but i am confused about it whether it
is possible or not :
I integrate it with asterisk for interanet no PSTN at all just
This sounds remarkably like an IM problem
We're in the process of building a CRM frontend that uses Jabber as the
IM mechanism. The Asterisk server sends the URL via Jabber (PCs
authenticated as extension number). The Jabber client (custom, written
in Flash) receives the URL and automagica
Does it support pre-paid billing?
On 5/30/05, Darren Wiebe <[EMAIL PROTECTED]> wrote:
> El Flynn wrote:
>
> > Darren Wiebe wrote:
> >
> >> Good Day,
> >> I'm finally getting around to officially announcing ASTPP. For the last
> >> 6 months, I've been working on converting ASTCC into a decent bil
On Thu, 2005-05-26 at 23:21, Armin Schindler wrote:
> On Thu, 26 May 2005, Mike Price wrote:
> > Yes libcapi is installed. Here is a sample of the errors I am getting:
> >
> > In file included from chan_capi.c:38:
> > chan_capi_pvt.h:92: syntax error before "_cword"
> > chan_capi_pvt.h:92: warning
Let it be officially stated that I wanted Harrison Ford to play my role as
digium staff member at ISPCon when the Asterisk movie comes out.
C. Savinovich
ITN-Telecom
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Kristian
Kielhofner
Sent: Sunday, May
Ever tried alot of sip devices on one asterisk box? You will see the
need real fast :)
Preston Garrison
direct: 877-748-4142
fax: 310-774-3901
cell: 623-748-4140
-Original Message-
From: Matt Riddell <[EMAIL PROTECTED]>
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sen
In extensions.conf.
You should rather read a little bit about what file does what, and how
extensions and contexts work.
http://www.asteriskdocs.org/
http://www.voip-info.org/
http://www.digium.com/index.php?menu=documentation
If you ask questions on the mailing list that you could answer by
rea
Wilson Pickett wrote:
dialplan is :
exten => 0,1,Record(recordyourmessage.gsm)
exten => 0,2,Playback(recordyourmessage)
exten => 0,3,Hangup
Though I pressed # after I finished recording my voice, but application
record() did not stop.
Two possible causes:
1 Your phone uses # as the
So I don't see the need.
Just forward 5060 and 1-2 to Asterisk from the firewall and put
nat=yes and canreinvite=no in sip.conf
--
Cheers,
Matt Riddell
___
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http://www.sineapps.co
OK -- follow up, for the record.
After looking at the patch code, it looks like the documentation is wrong:
[default]
1000:sip-reg-srv-001 => 1000,John Doe,[EMAIL PROTECTED]
should be:
[default]
[EMAIL PROTECTED] => 1000,John Doe,[EMAIL PROTECTED]
... and then extensions.conf can have the usual
After a lot of hours I'm getting close to finishing up a multi tenant
interface to asterisk based on the static database. I could use a bit
of help in one area if someone would like to contribute. The code
will be released under the bsd license.
Right now the only part that is not done is the con
> Though I pressed # after I finished recording my voice, but application
> record() did not stop.
Two possible causes:
1 Your phone uses # as the SEND key - try hitting # twice
2 DTMF setting is wrong try using RFC or INFO
___
Asterisk-Users mailing l
Hello,
I am new asterisk user. I am trying to setup asterisk locally. I have installed Red Hat 9.0 on my PC and I installed asterisk on it.
Then i configured sip.conf, Extensions.conf, voicemail.conf for two users. I am using Soft dialer to make calles.
I have two another PC's. So all the thre
Hello.
Though I pressed # after I finished recording my voice, but application
record() did not stop.
dialplan is :
exten => 0,1,Record(recordyourmessage.gsm)
exten => 0,2,Playback(recordyourmessage)
exten => 0,3,Hangup
I don't want either silence and maxduration time.
Can you help?
Regards.
hello David,
> [EMAIL PROTECTED] linux]# cp /boot/config-2.4.20-31.9
> /usr/src/linux/.config
> cp: overwrite `/usr/src/linux/.config'? y
after you copy this file, did u do make mrproper
before
u do make menuconfig?
coz before this i have error when i want to compile
zaptel and now its working
Hi,
I liked to have a pre paid
card in my asterisk Server.
I saw some applications in
the voip-ifo site, but all are not complete.
Somebody has some tested and
functioning solution ?
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El Flynn wrote:
Darren Wiebe wrote:
Good Day,
I'm finally getting around to officially announcing ASTPP. For the last
6 months, I've been working on converting ASTCC into a decent billing
package for asterisk.
The link in the original email opens a page that says
"Download the latest ve
Aidan Van Dyk wrote:
Browsing through the new website...
* Q - Does Asterisk Business Edition contain any additional features, fixes,
or enhancements not found in the open source versions of Asterisk?
* A - Digium remains committed to the open source model, and has based
Asterisk B
On Sunday 29 May 2005 20:59, Aidan Van Dyk wrote:
> 1) Simply CVS head (as of some point in time) with certain features or
>bug fixes "backed out"
>
> 2) In addition to CVS head, some important features and bug fixes.
I think it's simply #2. They are taking HEAD and maintaining a version wher
> There is a patch on Mantis (http://bugs.digium.com/view.php?id=4371)
> ...
Can somehow who got it to work help me out? Patches applied against
CVS-HEAD from 5/28, compiles and runs fine. It seems to do "something"
as well as I see messages like:
ERROR[22276] app_voicemail.c: Can't start the voi
Darren Wiebe wrote:
Good Day,
I'm finally getting around to officially announcing ASTPP. For the last
6 months, I've been working on converting ASTCC into a decent billing
package for asterisk.
The link in the original email opens a page that says
"Download the latest version of the code fr
Browsing through the new website...
* Q - Does Asterisk Business Edition contain any additional features, fixes,
or enhancements not found in the open source versions of Asterisk?
* A - Digium remains committed to the open source model, and has based
Asterisk Business Edition entire
[EMAIL PROTECTED] wrote:
On Sun, 29 May 2005, Steve Underwood wrote:
It isn't a one way scale. Music on hold over G.729 is often
unrecognisable. Over GSM 06.10 it is usually just poor. Is that a big
issue for you, or totally irrelevant? GSM 06.10 and G.729 at 8kbps offer
fairly similar qu
Good Day,
I'm finally getting around to officially announcing ASTPP. For the last
6 months, I've been working on converting ASTCC into a decent billing
package for asterisk. I'm just finishing up fixing a few bugs before
the 1.0 release and would appreciate if there would be a few who would
be w
- Original Message -
From: "Michiel van Baak" <[EMAIL PROTECTED]>
To:
Sent: Sunday, May 29, 2005 10:41 PM
Subject: Re: [Asterisk-Users] Peer to Peer calls
On 00:32, Mon 30 May 05, Cenk Yabas wrote:
Can anybody please answer this.
Both clients are behind different NAT's.
One of the
On 5/29/05, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
> On Sat, May 28, 2005 at 06:34:23AM +1000, Gonzalo Servat wrote:
[snip]
> > If Asterisk allowed me to configure up to 10 ringing patterns, I could
> > probably cover most of the ringing patterns being detected, but
> > unfortunately there is a l
Andres,
move on to the latest unicall-0.0.2pre16
it has a lot of fixes from pre2
Hector.
On 5/29/05, Andres Maduro <[EMAIL PROTECTED]> wrote:
Hi,I have successfully installed libunicall and mfcr2 in Venezuelan variant. We are able to receive, make calls without any problems.
When we place a ca
All,
I'm trying to set up Asterisk with an AVM C4 ISDN card, I've got stuck very
early on -- I'm usually pretty good at this stuff but why is the
documentation for Asterisk/CAPI/ISDN/Fedora so bad (almost non-existent)?
I have the following:
- Hardware: P3 3.2GHz, 1Gb PC3200 RAM, Intel S865W
All,
I'm trying to set up Asterisk with an AVM C4 ISDN card, I've got stuck very
early on -- I'm usually pretty good at this stuff but why is the
documentation for Asterisk/CAPI/ISDN/Fedora so bad (almost non-existent)?
I have the following:
- Hardware: P3 3.2GHz, 1Gb PC3200 RAM, Intel S865W
Hi,
I have successfully installed libunicall and mfcr2 in Venezuelan variant. We
are able to receive, make calls without any problems.
When we place a call from a snom sip phone through chan_unicall and press a
dtmf (either using info, rfc2833 or inband) Asterisk seems to get the order of
Be aware to the codec compatibility between peers. Direct calls to the
peer has to be under the same codec and initiation protocol.
And, yes, if you have (eg.) SIP and GSM, and careinvite=yes, the media
path dont pass through Asterisk.
Denis Galvao.
On 29/05/2005, at 18:32, Cenk Yabas wrote:
I've been using AMP to manage my * test system. I've been trying to
activate an extension that I don't want AMP to manage. It would appear
that the extesion definitions are placed in the appropriate "custom"
files which are then added with an include command to the appropriate
master file (sip.conf
What is the status with this?
On 4/8/05, trixter http://www.0xdecafbad.com <[EMAIL PROTECTED]> wrote:
> On Fri, 2005-04-08 at 11:45 -0500, Parker, Blake (MIS) wrote:
> > I will also donate to a good accounting/billing package.
>
> I will accept donations to write such a package if the following a
Greg Boehnlein wrote:
On Sun, 29 May 2005, Kristian Kielhofner wrote:
Dean Collins wrote:
Great booth guys, looks really interesting - can you cull out some of
the more lousy photos though.
Anything else you've seen at the event that's looks interesting?
Dean,
I could cut out some of th
On 00:32, Mon 30 May 05, Cenk Yabas wrote:
> Can anybody please answer this.
> Both clients are behind different NAT's.
> One of them starts a SIP call to the other through Asterisk.
> Asterisk sets up the call.
> Issues reinvite and connects them together.
> After this point does the media stre
Can anybody please
answer this.
Both clients are
behind different NAT's.
One of them starts a
SIP call to the other through Asterisk.
Asterisk sets up the
call.
Issues reinvite and
connects them together.
After this point
does the media stream flow through Asterisk or Peer to Peer?
Does
Do you have a sample h323.conf file? I finally got my * and CCM talking
to each other, but I had to use a gatekeeper and have them both go
through that. Without a GK, they both just sat there like they didn't
know what to do. My guess was always that my h323.conf file was wrong,
but there is
On Sun, 29 May 2005, Steve Underwood wrote:
> It isn't a one way scale. Music on hold over G.729 is often
> unrecognisable. Over GSM 06.10 it is usually just poor. Is that a big
> issue for you, or totally irrelevant? GSM 06.10 and G.729 at 8kbps offer
> fairly similar quality for clean voice
On Sun, 29 May 2005, Kristian Kielhofner wrote:
> Dean Collins wrote:
> > Great booth guys, looks really interesting - can you cull out some of
> > the more lousy photos though.
> >
> > Anything else you've seen at the event that's looks interesting?
>
> Dean,
>
> I could cut out some of
Hi Marcel,
I know.
I want to use one of my own msn numbers other then the default outgoing msn.
Regards,
Steven Lam
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marcel van Kaam,
Fonètica
Sent: zondag 29 mei 2005 21:08
To: 'Asterisk Users Mailing List
The Dutch KPN only send a callerID that the line recognize.
So you can only send the MSN of the line itself to an outside line.
Met vriendelijke groet,
Best regards,
Marcel van Kaam
Fonetica Teleservices
Bankstraat 88
3000 Leuven
Belgium
tel BE: +32 16 297270
tel US: +1 206 8660502
tel UK:
hi
i'm a newbie in asterisk...i installed asterisk but when i tried to
dial 1000 for the first time i got the following error messages and i
don't hear anything...
May 29 20:46:03 WARNING[262160]: chan_oss.c:413 soundcard_setinput:
Unable to re-open DSP device: Device or resource busy
May 29 20:
Hi to all,
I have a question about the callerid (msn number) off calls comming FROM
a queue.
This is my setup:
- ISDN using zap (zaphfc)
- Incomming calls arrive in a queue
- One of the members of the queue is my cell phone... member =>
Zap/g1/XXX73XX19 (X to protect my privacy ;-))
The problem
This is not the CLI output. Please reproduce the problem and paste the
CLI output, from both, when it's set to 10 seconds, and when It's set
to 60.
On 5/29/05, PistolPete <[EMAIL PROTECTED]> wrote:
> Show it sent call to vm. But outside call is terminated from PSTN
>
> -Original Message-
Show it sent call to vm. But outside call is terminated from PSTN
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C F
Sent: Sunday, May 29, 2005 12:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] 60 second ti
Known issue when using dhcp and cisco routers. Use dhcpd from linux server
and problem is gone forever... we do it constantly with 100% success.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C F
Sent: Sunday, May 29, 2005 1:06 PM
To: Asterisk Users Maili
New versions are great multiple lines and better control on buttons.
We are a Certified Reseller for Polycom in both IP telephony and video
products and have access to all Polycom's software.
We offer special discounts to SIP / Open source needs IP500 and 501 @ 189.00
on singles and 180.00 on m
I'm having the same issues with the polycom phones, as well as with
Sipura ata's. I am also using on another natted network a sipura ata,
that I changed the settings on the sipura that might help, and it did
help, I havn't had an unreachable message since.
I'm not sure if on the second network the
You can get the firmwares from the resellers that sold you the phone.
Accoding to Polycom they only give it to resellers.
As for your dhcp problems, check the release notes on www.polycom.com
they should give you clue.
On 5/27/05, Jeff Ramsey <[EMAIL PROTECTED]> wrote:
> I am using version 1.4.1.0
What is the CLI output?
On 5/29/05, PistolPete <[EMAIL PROTECTED]> wrote:
>
>
>
> If I try to execute this dialplan, and nobody picks up at any of the
>
> three extensions (7780 7781 and 7782), it's supposed to go to voice
>
> mail; instead, it hangs up and gives me a busy signal:
>
If I try to execute this
dialplan, and nobody picks up at any of the
three extensions (7780 7781
and 7782), it's supposed to go to voice
mail; instead, it hangs up
and gives me a busy signal:
exten =>
2001,1,Dial(sip/7780,20)
exten =>
2001,2,Goto(2001,102)
exten =>
2001,102,
I just noticed, looks nice
--
Andrew Latham - AKA: LATHAMA (lay-th-ham-eh)
WWW: http://lathama.com
Email: [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED]
If any of the above are down we have bigger problems than my email!
___
Asterisk-U
Dean Collins wrote:
Great booth guys, looks really interesting - can you cull out some of
the more lousy photos though.
Anything else you've seen at the event that's looks interesting?
Dean,
I could cut out some of the more lousy photos, but I would rather leave
them in case someone finds t
Jim Duda wrote:
The MENU key on the BT100 would work as I was attempting to "reboot"
the phone. I had to give the phone a hard power-cycle to restore it
to normal.
Has anyone experienced this problem with a BT100?
Yes, a couple of times on my BT102s. Once even had something like a
s
On Sun, 29 May 2005 14:51:08 GMT, Mohamed M Moustafa wrote:
>
>Hi,
>
>I need to upgrade my HOP-1002 ip phones (i am currently running
>ver.1.35), i need the source of the latest software and the
>steps to do it.
www.aredfox.com
>Thanks in advance.
>
>I am also configuring Asterisk as my SIP serv
Hi,
I need to upgrade my HOP-1002 ip phones (i am currently running
ver.1.35), i need the source of the latest software and the
steps to do it.
Thanks in advance.
I am also configuring Asterisk as my SIP server, any
recommendations ?
Regards,
Mohammed Mahmoud
__
I've been using
Asterisk for a few weeks now. I have a (1) BT100 phone and a Sipura-2000
for all my analog phones. All has worked rather flawlessly, until
today.
I was on the BT100
phone today. During my phone conversation, the BT100 disconnected and went
into a "click" mode. 2 "clic
Great booth guys, looks really interesting - can you cull out some of
the more lousy photos though.
Anything else you've seen at the event that's looks interesting?
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Kristian Kielhofne
Title: Database Usage with Asterisk
If you mean, can I access data files
through various database properties such as CTMF digits, time, CLI then the
answer is yes.
Do you have more information on what you
are trying to achieve?
From:
[EMAIL PROTECTED]
[mailto:[EMAIL
Hi everybody.
A few weeks ago I wrote to this list asking about anyone being able to
compile asterisk on a via EPIA 5000 board. I received some reply saying
like "no problem, just change to PROC=i586 in the Makefile".
So I did. :-)
Now I try to compile and I get to a certain point where I get th
Isamar,
Please don't take offense on what I said. I wasn't doubting your
comments, I was just curious. I will look into the other options you
provided.
Thanks,
Waldo
On May 28, 2005, at 10:18 PM, Isamar Maia wrote:
Waldo,
The external material quality is "business-level". It looks like
Hi:
After an Asterisk client behind a nat is registerd
with a remote server, how the remote server context in
the IAX file define Asterisk client so it can receive
calls.
Thanks.
__
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On 20:31, Sat 28 May 05, Gavin Hamill wrote:
> On Saturday 28 May 2005 20:21, Rusty Shackleford wrote:
>
> > D'oh!
> > I had misread the PP's statement and assumed he meant a "bareback"
> > browser window.
> > You are, of course, quite right. A Java app could handle this, but we
> > are still left
Title: Database Usage with Asterisk
Hi;
Can I connect asterisk to a database throw the dial plan
Regards;
Ghassan M. Lama'
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Michael George wrote:
On Sat, May 28, 2005 at 11:10:30AM -0400, Steve Totaro wrote:
qualify = yes is what is causing the messages. You can assign a value
rather than yes. like 1000 or something or you can remove the qualify
statement alltogether. The message is just a warning. Eliminating
On Sat, May 28, 2005 at 11:10:30AM -0400, Steve Totaro wrote:
> qualify = yes is what is causing the messages. You can assign a value
> rather than yes. like 1000 or something or you can remove the qualify
> statement alltogether. The message is just a warning. Eliminating the
> warning does
Hi Nathan,
Nathan E. Pralle wrote:
Greetings all.
Well, the first Asterisk Sound List in HTML was so popular, I did some more
fiddling around to make it even more useful. Here's an updated page:
- One master list with all sounds, sorted alphabetically by filename
- The old lists are linked
Hi All, Newby with problem when
attempting to make Zaptel on Red Hat linux 9.0.
I have seached Google, VoIP forums etc and cannot resolve
issue.(Red Hat 9.0) Linux version 2.4.20-31.9
([EMAIL PROTECTED]) (gcc version 3.2.2 20030222 (Red Hat Linux
3.2.2-5)) #1 Tue Apr 13 18:04:23 EDT 2004 Al
Thank you for the quick reply, but as I'm VERY NEW to Asterisk, where do I
make these changes?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Jean-Christophe Heger
Sent: Sunday, 29 May 2005 18:21
To: Asterisk Users Mailing List - Non-Commercial Discussio
On Fri, May 27, 2005 at 06:47:40PM -0500, Benjamin West wrote:
> Has anyone tried or had success statically linking Asterisk? I'd like
> to do this to deploy to smaller boxes that don't have the toolchain
> and libraries to build the thing.
You don't need all the toolchain on the target box just
Greg,
I was never able to automatically restart asterisk in fedora core 2
using asterisk/contrib/init.d/rc.redhat.asterisk.
I don't know if this happens only to me.
# make config
# /etc/rc.d/init.d/asterisk start
# kill -11 `cat /var/run/asterisk.pid`
asterisk doesn't restart.
I have to take
On Sat, May 28, 2005 at 06:34:23AM +1000, Gonzalo Servat wrote:
> Hi All,
>
> I've recently got a "second" number installed on my PSTN line,
> trusting the Asterisk distinctive ring detection would work as
> expected. It appeared to work fine at the start, as the second number
> generated a differ
exten => _0.,1,Dial(Zap/g1/9${EXTEN:1})
exten => _0.,102,Dial(Zap/g1/6129${EXTEN:1})
:1 strip out the digit on the left.
Claudio Angeloni a écrit :
> Ladies and Gents
>
> Please be patient as I try to explain what I am trying to achieve..
>
> I have a PSTN line and a Freshtel account, w
Ladies and
Gents
Please be patient as
I try to explain what I am trying to achieve..
I have a PSTN line
and a Freshtel account, what I want to do is have the PSTN line as the first
choice for outgoing calls for local calls and Freshtel as the second choice. The
problem is that it's
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