I have the similar problem but with IAX.I have two servers. First is primary with dynamic IP and open 4569 port. Second is behind firewall.If first server is being disconnected for some time, the second server cannot reconnect. I have to either restart asterisk or stop it, wait for some time and st
>
> Today I had a brief power outage which caused the Asterisk server and
> DSL modem to reboot. The Asterisk server came up before the internet
> connection was working, so it failed when try to look up some of the
> hosts for my outbound voip providers in sip.conf.
>
> Asterisk never recovered
Updating Asterisk is worth a go - I know of someone else who contacted
us with a distorted music on hold problem, and an Asterisk updated fixed
it.
PaulH
AsteriskIT
www.asteriskit.com.au
On Mon, 2006-08-21 at 16:14 +0800, Nathan Alberti wrote:
> On 20/08/2006, at 8:38 PM, Paul Hales wrote:
>
>
Today I had a brief power outage which caused the Asterisk server and
DSL modem to reboot. The Asterisk server came up before the internet
connection was working, so it failed when try to look up some of the
hosts for my outbound voip providers in sip.conf.
Asterisk never recovered from that,
As a couple of people have pointed out already, unless we doing
something wrong, there seem to be no jobs. Haven't you any comment on
that, other than to post another announcement about how great it is
now the employers have to pay?
On 20/08/06, Matt Gibson <[EMAIL PROTECTED]> wrote:
Hello All,
Hi, all. I'm using TE411P card and asterisk in Newzealand as a VOIP gateway.
At the begining, all works fine. I have 50 concurrent calls in busy time.
But recently I found some users can't send DTMFcorrectly to my gateway.
I found some of them sent less DTMF digits than they acturely dialed, whi
I don't know if it is a bug or a feature, it just surprised me.
Dial into an empty meetme. Alison says you are the only person in the
conference. While she is speaking, dial into the same conference from
another phone. When you enter the conference the new caller hears that
there is 1 other pa
ngrep is also good if you only want to see SIP traffic and filter all the lower
level stuff.
-Original Message-
From: Brandon Galbraith [mailto:[EMAIL PROTECTED]
Sent: Mon 8/21/2006 8:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hey all,
I've done some peeking around and can't find a GOOD listing of what the
currently supported SIP headers are that Asterisk supports. My main reason
is to get the CallerID/RPID settings for whether or not to display, but
there's others as well.
Anyone have a link?
SKM
_
Try Ethereal (I think it's called WireShark now). Does nice decoding of the packet stream to show you what's going on. Supports SIP for sure, not so sure about IAX though.-brandon
On 8/21/06, Leo Ann Boon <[EMAIL PROTECTED]> wrote:
Christopher Aloi wrote:> Hello List ->>> I'm a big fan of call
Christopher Aloi wrote:
Hello List -
I'm a big fan of call traces to diagnose a problem; I often use
"pri set debug file X" to write PRI traces out to a file, anyone
know of a similar method of saving IP traces (SIP,IAX) to a file?
Anyone have any ngrep scripts that do the tric
Hello List -I'm a big fan of call traces to diagnose a problem; I often use "pri set debug file X" to write PRI traces out to a file, anyone know of a similar method of saving IP traces (SIP,IAX) to a file?
Anyone have any ngrep scripts that do the trick?Thanks!-- --Christopher T Aloi--
--
Hi,
What's wrong with T3 timed out?
I use asterisk-1.2.10 package from ScopServ
http://www.scopserv.com/v2/home.php?section=news (ScopServ Telephony
Server 1.2.20).
Here there are four pages of the scanned report from our telco
http://www.flickr.com/photos/[EMAIL PROTECTED]/
==
Hi all,
Anyone has information on how Chinese equipment makers are encrypting
the SIP signaling + media packets to avoid ISP firewalls? Recently, I
was sent a sample FXS/O gateway with support for 3 flavors (Seawolf,
etc) of such encryption. I don't believe they're using SIP/TLS and SRTP.
At
First off, I don't work for Ericsson, but it is my impression that normally the
call control is done in Erlang and the media processing is done ineither hardware orC.I never said you work in Ericsson ;)
I You really are interested I recommend You to ask on[EMAIL PROTECTED]I did but they stopped r
I found this indication that Shared Line Appearance is possibly in SVN. Is
it or is this just an indication that it is up and coming?
http://bugs.digium.com/view.php?id=7701
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users ma
All I can find for Flite is for AAH, does it work as well with plain
Asterisk? Is the setup the same?
Never tried it, but it should be the same.
Have a look here : http://dialogpalette.sourceforge.net/extras.html
hth
___
--Bandwidth and Colocation pro
If you are originating a call with a Local/ channel you cannot use the
uniqueID alone to track it. The only field that will follow all legs
of a Local/ channel originated call is the CallerID, and that is only
if you add the "o" flag to your Dial string.
It's a very messy prospect to track calls
Hello,
I am not so really familar with asterisk at the moment, but i am working
hard on it. Please could anybody advise me how to write a call file for the
queue to do 2 outbounds call and connect both via my SIP interface.
Thanks in advance.
Daniel
__
Yes there is but only in Bristuff asterisk.In bristuff when you enter Originate command you receive feedback with uniqueid of created call.So than you can trace uniqueidgreetingsmk
2006/8/21, Tony Mountifield <[EMAIL PROTECTED]>:
In article <[EMAIL PROTECTED]>,Nic Bellamy <[EMAIL PROTECTED]> wrote:
Hello,
I am not so really familar with asterisk at the moment, but i am working
hard on it. Please could anybody advise me how to write a call file for the
queue to do 2 outbounds call and connect both via my SIP interface.
Thanks in advance.
Daniel
___
In article <[EMAIL PROTECTED]>,
Nic Bellamy <[EMAIL PROTECTED]> wrote:
> Hi,
> I'm having a bit of trouble matching up Newchannel (and Newexten,
> etc. etc.) events with the Originate that created them.
>
> Basically, what I want to do is have software automatically initiate a
> call, and th
Anytime I try and specify a voice when there is more than 1 voice in my
voices directory...it has an error with the syntax you show here...
Like I was saying in a previous post...
- Original Message -
From: "Shane Young" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commerc
Douglas Garstang wrote:
Does anyone know if realtime extensions support the use of labels?
I don't believe so.
As I understand it, the dialplan parser internally converts n-type and
labeled priorities to a straight numeric format, which is then used
internally.
Becuase the Realtime engine
You might try runtime Dectalk for Linux available from
http://www.fonix.com -- its not free, but it sounds quite nice.
on Monday 08/21/2006 Time Bandit([EMAIL PROTECTED]) wrote
> N.B.: Please use plain text when sending to this list
>
> > Can someone recommend a good text to speech engine that
Thank you very much Carlos, you are absolutely right. Now it works!
Thanks again.
---
Dominique Dartois
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Carlos Chavez
Envoyé : lundi 21 août 2006 23:09
À : Asterisk Users Mailing List - Non-Commercial Di
Hi,
I'm having a bit of trouble matching up Newchannel (and Newexten,
etc. etc.) events with the Originate that created them.
Basically, what I want to do is have software automatically initiate a
call, and then track the status of that call through to completion.
I can match to some degr
Quoting Kevin Savoy <[EMAIL PROTECTED]>:
> Can someone recommend a good text to speech engine that is usable by
> Asterisk? I have tried the Festival one and it just doesn't cut it for
> commercial applications.
I like Cepstral.
Using the information here:
http://www.oldskoolphreak.com/tfiles/vo
On Mon, 2006-08-21 at 23:03 +0200, Dominique Dartois wrote:
> I want to hear french messages. I put language=fr in the [globals] section
> of extensions.conf and in the [general] section of sip.conf.
>
> The french messages are at the right place :
> /var/lib/asterisk/sounds/fr/digits/*.gsm
I want to hear french messages. I put language=fr in the [globals] section
of extensions.conf and in the [general] section of sip.conf.
If I call an unavailable number, the digits are read in english even if the
trace says french :
-- Executing VoiceMail("SIP/103-6441", "[EMAIL PROTECTED]") i
All I can find for Flite is for AAH, does it work as well with plain
Asterisk? Is the setup the same?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Time Bandit
Sent: Monday, August 21, 2006 3:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discuss
Hi,
I have site using only softphones (SJPhone under Windows). Once in a
while the users complain that they hear double and triple dial dtmf when
they dial out.
What could be causing that on the asterisk side?
Andre Courchesne
___
--Bandwidth and
Hi Simon,
I did "yum update" last week and here is my current kernel:
# uname -vr
2.6.17-1.2174_FC5smp #1 SMP Tue Aug 8 16:00:39 EDT 2006
#
# ls -l /usr/src/kernels
total 12
drwxr-xr-x 18 root root 4096 Jul 8 19:43 2.6.17-1.2145_FC5-smp-i686
lrwxrwxrwx 1 root root 26 Jul 8 19:43 2.6.17-1.214
Cepstral seems to sound descent...But if you have
more than one voice installed (Example: different languages)
I can't say it in realtime in the dialplan...I have
to do a little trick like:
exten => 1,1,System(/opt/swift/bin/swift -n
Diane-8kHz "Hello World" -o /var/lib/asterisk/sounds/swif
N.B.: Please use plain text when sending to this list
Can someone recommend a good text to speech engine that is usable by Asterisk?
I have tried the Festival one and it just doesn't cut it for commercial
applications.
We are willing to pay for a good one that works. Anyone tried the AT&T s
Can
someone recommend a good text to speech engine that is usable by Asterisk? I
have tried the Festival one and it just doesn’t cut it for commercial
applications.
We
are willing to pay for a good one that works. Anyone tried the AT&T speech
engine? The IBM ViaVoice sounds no better t
> -Original Message-
> From: Tzafrir Cohen [mailto:[EMAIL PROTECTED]
> Sent: Sunday, August 20, 2006 5:08 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Apache for FastAGI
>
>
> On Sun, Aug 20, 2006 at 04:18:11PM -0600, Douglas Garstang wrote:
> > I'm not sure th
Well, for one, you could set something like CID = ${CALLERID(number)} in the
dialplan, and then GetVar CID
>-Original Message-
>From: [EMAIL PROTECTED]
>[mailto:[EMAIL PROTECTED] On Behalf Of
>Warren (mailing lists)
>Sent: Monday, August 21, 2006 3:54 PM
>To: Asterisk Users Mailing
Ricardo Carvalho wrote:
I've been searching for sound files in Portuguese language to use in
Asterisk for example for voicemail, but I couldn't find anything...
Does anyone know where I could find them for download, if there is such
thing already?
Brazilian Portuguese only...
http://www.google
But how do you get that with GetVar? I am trying to do this through the
API. I tried:
Action: GetVar
Variable CALLERID(227)
and I tries:
Action: GetVar
Variable ${CALLERID(227)}
Neither returned anything.
How can I do this? Alternately... Is there a way to have a program
fired off when an
Can realtime be used with hints? How would you get the following into the
database given that the priority column is numeric, and that there is no
application for the first entry?
exten => 2944006,hint,SIP/2944006
exten => 2944006,1,Dial(SIP/2944006)
Every time I touch realtime I hit obstacles.
David Gagnon wrote:
Finally, in the trunk all the states of my device are broken. If I
downgrade to 1.2.10, everything is fine. The device get busy and
ringing. But in the current trunk Asterisk SVN-trunk-r40632M none of my
hints works.
Anyone could confim this bugs ?
David,
I haven't hea
Does anyone know if realtime extensions support the use of labels?
ie:
exten => acdpause,1,Answer
exten => acdpause,n,Wait,1
exten => acdpause,n,PauseQueueMember(|Agent/${CALLERIDNUM})
exten => acdpause,n,GotoIf($["${PQMSTATUS}" =
"
${CALLERID(number)}
>-Original Message-
>From: [EMAIL PROTECTED]
>[mailto:[EMAIL PROTECTED] On Behalf Of
>Warren (mailing lists)
>Sent: Monday, August 21, 2006 1:41 PM
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>Subject: [asterisk-users] Variable to show caller id for
I have tried it with exten => 5481,2,DIAL(IAX2/5480,,D(w1)) and it also does
not work.
I have since moved it to an analog extension on a legacy PBX.
I have tried:
exten => 5481,3,DIAL(Zap/g2/5110,,D(1))
and a macro with SendDTMF.
It works fine if I dial 5110, then enter the number of the zo
What is the status of it anyway? I followed the "bug" for it and it
appears that the bug was closed and maybe it was incorporated into
Trunk. Is this true? And should it be (fully) functional now?
PA
___
--Bandwidth and Colocation provided by Easy
Steve Underwood wrote:
> marek cervenka wrote:
>
>> hi,
>>
>> bounty for t.38 is $11,750. that looks good!
>> http://www.voip-info.org/wiki/view/Asterisk+T.38+Bounty
>>
>> how high must be bounty for Digium to hire programmer for this?
>>
>> thanks
>
> Do you really think T.38 can be implemented
Is there a variable that can be gotten with GetVar to show the callerid
of the current incoming call in progress at a sip extension?
For instance, a caller from 516-922-9463 calls extension 234. I would
like to be able to be able to get back the 516-922-9463 if I pass 234.
Also, can this be
Hi,
I've been searching for sound files in Portuguese language to use in
Asterisk for example for voicemail, but I couldn't find anything...
Does anyone know where I could find them for download, if there is such
thing already?
Regards,
Ricardo.
_
Hi Doug -
Let me start by saying when I first plugged it in, I didn't have the
files set up on my ftp server yet, and the phone used it's default
settings and it completed bootup. Now...
I started with sip v1.6.6b and bootrom 3.1.3 on the ftp server. Phone
boots, d/l's files, reaches "Welcome
Hi,
Tomer Horn wrote:
Are there any known (bad) issues / experience running Asterisk inside
Xen VM? Has anyone experienced running Asterisk inside a Xen VM with PCI
access to PRI adapter?
We do this a lot, although I believe our engineers are still using Xen2
for systems with BRI/PRI adapter
Is there a way to find out if a channel is currently being
recorded/monitored via the Asterisk Manager API.
Currently, if I issue a "Action: Status", it lists all channels as
"unmonitored", regardless if they're being recorded or not.
(In my setup, I'm not doing automatic monitoring, I have a
How where you able to interact with the callee after they had answered the
call? You lose control of the dial plan after someone answers, until they hang
up.
> -Original Message-
> From: Roy Kidder [mailto:[EMAIL PROTECTED]
> Sent: Monday, August 21, 2006 5:05 AM
> To: Asterisk Users Mai
Hello!
Are there any known (bad) issues / experience running Asterisk inside
Xen VM? Has anyone experienced running Asterisk inside a Xen VM with PCI
access to PRI adapter?
Regards, Tomer.
___
--Bandwidth and Colocation provided by Easynews.com --
Hi all,
I'm trying to use a bigger appdata column for realtime, the reason being
that I'm moving to a new setup where the SIP devices are named according
to the name of the user and some of my dial/page commands need to dial a
goodly number of phones which then exceeds the 255 max size of the
colu
I am trying to track down a problem which is occurring on
about 1% of the phone calls through a customer’s system.
Layout looks like this:
PSTN ß PRI ß Asterisk A ß IAX Trunk over point to point T1
ß Asterisk B ß SIP over LAN ß Polycom
IP501
1) The user on
the Polycom IP5
Hi,
we have a setup with an Asterisk, an openser and a Cisco 5400 in place.
Asterisk is the frontend to the users, providing registering and RTP
proxy functionality and openser is the gate-keeper of the Cisco.
I can call in and out, everything is fine so far.
But there is one strange fact: wh
Hello list,
Was wondering if anyone knows how to get DTMF to work on voipjet..
Tried,
dtmf=rfc2833
dtmfmode=rfc2833
doesn't seem to work...
Any clues?
Cheers!
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To
marek cervenka wrote:
hi,
bounty for t.38 is $11,750. that looks good!
http://www.voip-info.org/wiki/view/Asterisk+T.38+Bounty
how high must be bounty for Digium to hire programmer for this?
thanks
Do you really think T.38 can be implemented on a contract basis for
$11,750? Besides, thes
This is a commercial activex you may want to evaluate: http:/.www.vaxvoip.com It worksLennie De Villiers <[EMAIL PROTECTED]> wrote: Hi, I'm looking for a SIP ActiveX component to use in Visual
Basic/Delphi. Thanks Kind Regards, Lennie De Villiers
> -Original Message-
> From: Benny Amorsen [mailto:[EMAIL PROTECTED]
> Sent: Monday, August 21, 2006 3:39 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Re: Asterisk 'Hosting'
>
>
> > "JM" == Jeremy McNamara <[EMAIL PROTECTED]> writes:
>
> JM> Why do you need mu
www.zhone.com. Their MALC can handle >500 POTS lines in a 23" shelf
with POTS -> VoIP (SIP/MGCP). 'Telco quality' and the per port cost
for high density isn't that bad.
You could probably also go with a bunch of CAC AccesBanks connected
to a CAC Widebank, connected to a Lucent TNT and
This is what I am trying to do, yes, as to do all DID administration
myself without contacting the switch monkey.
It's quite possible, it seems, by sending a cause 34, lying about no
bchans being available to handle the call.
Thanks for reporting back, I like this idea :) thanks again.
___
On 8/18/06, Shidan <[EMAIL PROTECTED]> wrote:
I don't know if I responded to the original poster before but if you are
looking for a python fastAGI server, there already is one, its called
starpy.
Anders, since you know Erlang, do you know of any media processig
libraries in Erlang, do the eric
Sorry. It sould say SIP-to-Zap not the other way around. Meaning that the Zap
user is heard fine, but the external-SIP user is choppy when calling out on Zap
(not when calling SIP-to-SIP though).
-Ursprungligt meddelande-
Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För [EMAIL PROT
hi,
bounty for t.38 is $11,750. that looks good!
http://www.voip-info.org/wiki/view/Asterisk+T.38+Bounty
how high must be bounty for Digium to hire programmer for this?
thanks
---
Marek Cervenka
===
On Mon, Aug 21, 2006 at 09:52:23AM -0400, Ferguson, Michael wrote:
>
> G'Day List,
>
> I am looking for documentation on how to configure sendmail to deliver
> asterisk voicemails to the recipient's mailbox.
Nothing special about sendmail. Basically any standard MTA: sendmail,
postfix, or wha
Hi guys,
Does anyone know whether is it possible to call System (Execute a system
Linux shell command) dialplan application via AMI? If so, how?
Thanks in advance,
*
___
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asterisk-users mailing lis
Hi,
I have a strange problem about the cpu consumption of a IAX trunk.
I have two asterisk connected by a IAX trunk.
The asterisk number 1 is installed on a Soekris Box
Asterisk 1 Asterisk 2
IAX T
| --> |
I use another asterisk to generate some traffic
This feature was supposed to be in 1.2, in fact Kevin promised me that
it would be since I had it in before the feature freeze for 1.2. It
did not go in. Since then I have had to move on to other things and
others have tried to keep it going. This is really a very basic
function that should be in
G'Day
List,
I am looking for
documentation on how to configure sendmail to deliver asterisk voicemails to the
recipient's mailbox.
I Googled it but
found many many references to the fact that asterisk can do that but no
How-To's.
I believe sendmail
is running on my asterisk box as:
Is there a way to initiate 2 different calls and connect them together with
Asterisk, using the manager.api or the AGI system? I want to link the calls
without using DTMF, such as with an SMS or web triggered script.
The only way right now is using meetme. There is a patch with a
'bridge' functi
Is there a way to initiate 2 different calls and connect them together with
Asterisk, using the manager.api or the AGI system? I want to link the calls
without using DTMF, such as with an SMS or web triggered script.
I thought the call files would be able to set the necessary AGI variables for
t
Hi,
I have lately noticed that we sometimes get choppy sound when recieving
calls from the PSTN (on a TE410P-card) that get sent to an external SIP
extension (over the internet) who has a somewhat bad connection.
The strange thing is that it still sounds good when calling internally
to the SIP-to
On Sun, Aug 20, 2006 at 06:20:42PM -0300, Danko Miocevic wrote:
> Hello, is there any way to send signals to asterisk, for example, I send a
> sign to a parallel port and it calls an extension. I can´t modify asterisk
> code to make it. Any ideas?
> Thanks for your time,
>
Allan Kamau schrieb:
> I would like to run a fast-agi application in the
> background.(cmd agi())
> This is because I would like to implement a
> "disconnect after so many seconds" feature or at least
> a log of the duration of the call.
What about using an Option of the Dial-App instead ??
S(n):
On Sunday 20 August 2006 10:55, Roy Sigurd Karlsbakk wrote:
> - If the call received by asterisk from the PRI is sent to a number
> not in the dialplan, what will asterisk do? Will the call be
> cancelled, or will asterisk signal something back to the switch to
> indicate "dunno about this, try a
Hi,
I'm using the polycom branch and have been trying to get the
outboundproxy=xxx to work. Is this something that should work in the version
of software?
Thanks,
Dean.
___
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asterisk-users mailing
In which case your best bet is probably to install with an rpm --
rebuilt on the source rpm.
simon
On 21 Aug 2006, at 12:36, Tomislav Parčina wrote:
In article <344F8B3D-6591-4001-9DE6-
[EMAIL PROTECTED]>, [EMAIL PROTECTED]
says...
I managed to get zaptel to compile reasonably easily on
2.
hi,
we've got a Diva Server BRI-2M PCI, SN:3485 card in our asterisk server.
we use the latest divas4linux-melware-3.0.g-106.628.1-1 driver for it.
the card is connected to a Bosch Integral33 PBX. the two system
connected with an S0 line in order the two pbx be able to call
each other. when we call
In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] says...
> I managed to get zaptel to compile reasonably easily on
> 2.6.17-1.2157_FC5smp. However, the yum repo sites do not provide
> devel packages for 2.6.17-2174 for some reason last time I checked,
> hence couldn't get it to build on that
I couldn't find 2.6.17-1 for download but this is what I used to install the
kernel source
http://download.fedora.redhat.com/pub/fedora/linux/core/5/source/SRPMS/GFS-kernel-2.6.15.1-5.FC5.17.src.rpm
Dennis Clark
DENPRO
WRK 207.618.1998
CEL 443.415.0527
FAX 1.888.811.8809
[EMAIL PROTECTED]
I would like to run a fast-agi application in the
background.(cmd agi())
This is because I would like to implement a
"disconnect after so many seconds" feature or at least
a log of the duration of the call.
When the call is answered, the application checks to
see the number of seconds (talk time)re
I did something along these lines, but I was playing the caller ID back to
the caller, not after a transfer. In a perl AGI script. I split the caller
ID number into an array, seperated by '//' so each number was an element.
Then I played digits/$array[0]... digits/$array[1]...etc.
coolbreeze wrot
Yeh but as [EMAIL PROTECTED] is now called Trixbox so go to www.trixbox.com
Cheers,
Dean
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Paul A Brown
> Sent: Monday, 21 August 2006 3:58 AM
> To: Asterisk Users Mailing List -
In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] says...
> Hi!
>
> This patch does passthrough the AOC information from on ZAP channel to
> another ZAP channel. There is no support yet for storing the AOC value
> as CDR, but I think this may be easily added.
Hi Klaus!
I'm not programmer so I
I managed to get zaptel to compile reasonably easily on
2.6.17-1.2157_FC5smp. However, the yum repo sites do not provide
devel packages for 2.6.17-2174 for some reason last time I checked,
hence couldn't get it to build on that kernel. You could probably
create the devel package without too
I'm trying to install Zaptel 1.2.7 on Fedora Core 5 with 2.6.17-1.2174_FC5
kernel from source code. When I untar Zaptel and execute this is error that I
get.
cc -o ztmonitor ztmonitor.o
cc -o ztspeed.o -c ztspeed.c
cc -o ztspeed ztspeed.o
cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DST
Lennie De Villiers wrote:
Hi,
I'm looking for a SIP ActiveX component to use in Visual Basic/Delphi.
Thanks
You can find a proof of concept at
http://www.pernau.at/kd/voip/bookmarks-sip-phones.html
It's called ActXPhone
regards
klaus
___
--Bandw
>> So, a few questions:
>>
>> - If the call received by asterisk from the PRI is sent to a
number
>> not in the dialplan, what will asterisk do? Will the call be
>> cancelled, or will asterisk signal something back to the switch to
>> indicate "dunno about this, try another"?
>
> Asterisk wil
Tomislav Parčina wrote:
In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] says...
Hi Marco, as good?
Well, you are use libpri-1.2.3?
Believe that this is a bug of this version. Look at link´s below, contains
patchs for this "problem".
I wait to have helped.
Best Regards
Josué
http://bugs.digi
> "MR" == Matt Riddell (NZ) <[EMAIL PROTECTED]> writes:
MR> And so you're thinking it would be better to run several hundred
MR> Asterisk instances?!
Why not? As long as you stay away from the things that need zap
timing, asterisk is really not much of a load.
/Benny
_
Hi all,
we do have the following configuration
(non-Asterisk PBX) - T1 - ZAP (Asterisk PBX) - ZAP - T1 - (GSM Gateway)
-> GSM Enduser
The call is originated on the (non-Asterisk PBX) - gets send over a T1
connection to the asterisk server (which does least cost routing) - the
asterisk serve
> "JM" == Jeremy McNamara <[EMAIL PROTECTED]> writes:
JM> Why do you need multiple instances? Just setup your Asterisk
JM> configuration to separate the various 'customers' or 'tenants'.
The configuration files balloon to unmanageable sizes, and changing
them means that you risk breaking tele
U need 2 give more info on your setup i.e wherether u have
sipclient<<>>asterisk<<>>nat<<>>sipclient or whatever the situation is .
Anyway in the mean time just rtp dubug on and see wherether there r rtp
packets sent back and forth
>
> there's actually no audio b/w sip to sip calls.
>
> I jus
Hi,
I'm looking for a SIP ActiveX component to use in Visual Basic/Delphi.
Thanks
Kind Regards,
Lennie De Villiers
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On 20/08/2006, at 8:38 PM, Paul Hales wrote:
Does anything pop up on the Asterisk screen?
Does music on hold work fine?
PaulH
On Fri, 2006-08-18 at 13:13 +0800, Nathan Alberti wrote:
Nothing strange on the asterisk console... just stopped and started
hold on channel.
If I repeatedly
Hi all,Could anyone help me, why my calls of some clients disconnecting with the following error message: i have more than 500 IAX users but it is happening with very few customers.I will be appricate for u kind of help.Error::Auto fallthrough, channel 'IAX2/2001@2001/1' status is 'UNKNOWN'
Stay
Thanks in advance
Paul
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asterisk-robert wrote:
I need to send some information to our German HQ regarding my experiences with
VoIP.
> Asterisk is very prominent in those experiences. I would like to
include
> information about installations of Asterisk at
> German companies/universities.
We have installed Asterisk
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