On Wed, 2005-02-09 at 18:07 +0100, Remco Barende wrote:
> Hi list!
>
> Sorry for this slightly off-topic message but does anybody know if the
> standard for ISDN BRI is the same in Spain as it is in the rest of Europe
> (or the Netherlands).
>
> Will a standard HFC-S card work?
Afaik Telefonic
- Original Message -
From: "Pedro" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Thursday, February 10, 2005 4:08 AM
Subject: Re: [Asterisk-Users] Zombie SIP channels
> Thanks for the tip. They both seemed to go away on their own after a
> whi
Dave Green wrote:
Following a top posted thread is a pain.
not trimming the useless part of a reply is another pain...
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As my previous try on getting an answer was hijacked I thought I would
try again.
Ok I have a challange that I can't seem to find a way to fix it.
My Voicemail in * timesout after 30secs without fail everytime no matter
what I do.
I have incomming calls comming in through Freshtel IAX2, if it goe
So I'm having trouble getting call-pickup working. Got a few different
SIP phones (cisco 7940's and SPA-841s) all with pickupgroup=0 in sip.conf.
I can't seem to get it working. This *is* possible from SIP phones,
right? Do I need to add anything to my dial-plan?
--
Paul A. Dugas
Hi,
I have a polycom reboot script which sends a NOTIFY with check-sync. It
worked fine with 1.3.4. After I upgrade to 1.4.1, it stopped working. Anyone
has the same problem?
Thanks,
Richard
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Jerry,
Thanks a lot for the feedback!
By the way, how long did it take to replace the faulty 10% of phones by RMA?
What company did you use to buy it from?
Jerry wrote:
On Feb 9, 2005, at 9:14 PM, Sergey Kuznetsov wrote:
Hi there,
Does someone can share the experience with Cisco and Polycom Phones
On Feb 9, 2005, at 9:14 PM, Sergey Kuznetsov wrote:
Hi there,
Does someone can share the experience with Cisco and Polycom Phones?
How rock solid are they? And who will win in sound quality contest?
I heard that Cisco phones is a Polycom replicas with changed design.
Is that true?
What else phone
Anton Krall wrote:
Guys, Im new to asterisk and voip but Im have a couple of questions
regarding the initial setup.
1. Im going to install an asterisk server at home, where I have 2 phone
lines, what kind of card do I need to get? I was thinking about 2 X100P
Cards, so 1 can have 2 FXO ports and re
Hi,
http://www.voip-info.org/tiki-index.php?page=Asterisk%20consultants
Best regards,
Chris HARIGA
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hariharan
Gopalan
Sent: Wednesday, February 09, 2005 10:19 PM
To: Asterisk-Users@lists.digium.com
Subject
Hi
Was wondering if there is any directory of Asterisk Consultants.
I have some project ideas and need help and am looking for asterisk consultants
Thanks
Hari
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Hi there,
Does someone can share the experience with Cisco and Polycom Phones?
How rock solid are they? And who will win in sound quality contest?
I heard that Cisco phones is a Polycom replicas with changed design. Is
that true?
What else phones is better to implement to the medium sized busines
Thanks for the tip. They both seemed to go away on their own after a
while with no action on my part. I am not sure what caused it (there
is nothing in the log file). This is the first time I have seen it on
any of my asterisk machines (and I have been working with asterisk for
a year now).
Any
Pedro wrote:
No one is on a call - how can I get rid of this without restarting asterisk?
soft hangup in Asterisk console.
It'd pay to try and find out why you're getting them though.
:)
--
Cheers,
Matt Riddell
___
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Shaun Ewing wrote:
You need to do an incremental upgrade - eg: SIP 2.3 -> SIP 4.4 -> SIP 7.3.
The ealier images can be obtained from Cisco if you have a valid CCO login.
Actually, you need to go from earlier version to version 5. This is the
version Cisco started using an encrypted firmware
Hi all,
If you're in Melbourne Australia and interested in Asterisk, you're
invited to join us for a casual evening to talk about Asterisk, VOIP,
networks, and just generally get geeky about IP phone stuff.
Ultimately, I think it would be interesting and useful to turn this
into a monthly get-toge
I had similar distortion issues and shutdown problems, I had better results
using madplay instead of mpg123 as per the wiki:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20musiconhold.conf
Cheers
Walt
_
FREE pop-up
On Wed, 09 Feb 2005 12:20:01 -0800, Max Clark <[EMAIL PROTECTED]> wrote:
> Which lead me to believe that there was an 8.3 naming issue when using a
> windows based tftpd server. So I changed the file names of my image to
> an 8.3 structure, updated the configuration files and rebooted. After I
> do
I tried to send this earlier but does not look like it went through
for some reason. If you get this twice - my appologies.
Does anyone know how to kill a zombie channel (and why do they pop up)?
Here is what I see on a show channels:
--
show chann
remember to use the ^ to indicate matching at the beginning of the number.
i.e ^01144 should be all you need to match any international call going to
country code 44.
Karl Putz
>-Original Message-
>From: [EMAIL PROTECTED]
>[mailto:[EMAIL PROTECTED] Behalf Of Jason
>Kawakami
>Sent: Wedne
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Well, a bit of a newbie using Asterisk and trying to get some Sipura SPA-841
phones to talk to the Asterisk server. Not doing something right
apparently
If I turn on debug for the extension in the CLI (sip debug ip xxx.xxx.xxx.xxx)
I see freque
So I have a route with [1-9][0-9][0-9][1-9][0-9]* as a base route that
should match NXXNX. Right?
I built another route 01144[0-9]* that I thought would match 01144X. and
send the call to the UK but the script is matching 01144207108 With the
first route.
Can someone smarter than me help with
> > I just have one problem with that. I have _never_ (that I know of) seen
> > a phone with A-D on it!
> >
>
> They were (are?) mostly used for military installations.
And used by some Operators (when you dialed "0") in the olden days.
___
Asterisk
> > > Can anybody confirm if IAX on FWD is down again?
> > > I can not register IAX with FWD.
> > >
> >
> > I got fed up with the yo-yo, which then led me to dump fwd and install
> > asterisk and start playing with inter-asterisk routing via e164.org...
> >
>
> I wander what is causing the pro
I'm able to register sip friends with asterisk
but i wish to use presence and instant messaging.
asterisk sent back "Proxy Authentication required
407" when SUBSCRIBE is sent to asterisk.
harry
--- Noah Miller <[EMAIL PROTECTED]> a écrit :
> Hi Harry -
>
> > Can you get SUBSCRIBE registration
Can anyone provide a good manufacturer of
echo cancellation equipment for a PRI?
--
Richard Cook
[EMAIL PROTECTED]
T: 705-497-9320
x2010
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Andrew Thompson wrote:
Greetings,
Sorry, wrong list :-D
--
Andrew Thompson
http://aktzero.com/
http://dev.asteriskdocs.org/
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On Wed, 9 Feb 2005 20:54:51 +0200, Walid Azab <[EMAIL PROTECTED]> wrote:
> Just want to understand the difference between Asterisk Versions and please
> correct me if I am wrong, I understand they are:
>
> Stable
> CVS
> CVS Head
As I've noticed several posts regarding Asterisk versioning r
1 – 2000 UDP is wrong try 1-2UDP and try port forwarding
rather than opening ports . It probably has to do with the ip of the
server.
mohammed
Jeff R Glassman wrote:
I had my [EMAIL PROTECTED] server working fine SPA-8841 SPA-2100. It was
on an open IP no fire wall. I moved the ser
it probably has to do with the handset of your phone. Try using a good
mp3 from the web and test it using that
mohammed
Ousmane Doukara wrote:
Est ce que quelqu'un peut me dire pourquoi le volume de mes enregistrement
sont tout le temps trop bas.
Hi,
I am trying to figure out why my recorded fil
Greetings,
I'd like to thank everyone that has responded to my original email. I
have received information from several companies, and will be testing
several of them.
I also would like to update a statement from my original message to
clarify it:
>My strikelist: nufone, voicepulse, iax/sixtel
no this is not normal and the codec has very slight effect on the delay.
Delay is a function of two things:
1) Transcoding
2) Routing (dialplan routing + network latency)
Network latency >> Transcoding+dialplan routing.
If you are using two sip client which are using the same codec then no
transco
We are experiencing problems with FXO modules on TDM400P. From time to time
they stop responding to incoming rings although they work fine if we use
them to dial out. It's been verified at least in two different installations
(using different mainboards) in two different locations.
The only soluti
i want the to find out the delay between two events:
1) the instance a call is recieved on an FXO port and the
2) the instance a SIP INVITE is sent to the SIP destination.
i need to attach timestamps to the events before logging them. How can i:
1) log `ALL' events.
2) Attach timestamps to them?
t
I would say yes, even I have not tried too many of them, just VoicePulse
Connect and iax.cc.
-D
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Robert Goodyear
> Sent: Wednesday, February 09, 2005 10:55 PM
> To: Asterisk Users Mailing List - No
Does anyone know how to kill a zombie channel?
Here is what I see on a show channels:
--
show channels
Channel (ContextExtensionPri ) State Appl.
Data
SIP/frontdesk-72c7 (customercontext 1 ) Up
Bridged
Steve Blair wrote:
I'm not sure of your configuration but last time I looked there
wasn't a SIP v8.3. Unless that has changed I'd suggest getting
the correct version of SIP if that is what you'd like to test.
He was talking about 8.3 filename formats not firmware versions.
Following a top posted
Steve,
No, not SIP version 8.3, but the SIP firmware in an 8.3 formatted file
name (i.e. DOS).
-Max
Max Clark
max [at] clarksys.com
http://www.clarksys.com
Steve Blair wrote:
I'm not sure of your configuration but last time I looked there
wasn't a SIP v8.3. Unless that has changed I'd sug
Any idea if that's happening with other VoIP providers?
I too have about a one-second dropout right when the destination number
answers.
Cisco 7960 > * > IAX2 > VoicePulse Connect
/rg
On Feb 9, 2005, at 1:04 PM, David Hajek wrote:
Hi,
I'm experiencing some voice delay (2-3 sec) after outgoing cal
Hi,
On Wed, 2005-02-09 at 13:43 -0800, Marco Gonzalez wrote:
> I'm trying to make a video conference with sip. I have
> the Eyebeam from Xten, a video sip phone. I have a
> good audio conection, but nothing about the video
Dit you enable videosupport in the sip.conf and allow the proper
vide
Oops, my fault!
Disclaimer: Where is my favorite coffee mug ?!
Dana Olson wrote:
And for Windows, a minimum purchase of 1 unit... Is he using Mac or
PocketPC? If not, then he doesn't have to worry.
On Wed, 09 Feb 2005 15:59:45 -0500, Sergey Kuznetsov
<[EMAIL PROTECTED]> wrote:
Hi everyone!!! My Name is Marco, I'm from
Caracas,Venezuela. I'm a new Asterisk user...
I'm trying to make a video conference with sip. I have
the Eyebeam from Xten, a video sip phone. I have a
good audio conection, but nothing about the video
Now I'm trying to do the same with h323, but don
I'm not sure of your configuration but last time I looked there
wasn't a SIP v8.3. Unless that has changed I'd suggest getting
the correct version of SIP if that is what you'd like to test.
Max Clark wrote:
What I don't understand is that the phone will request and download
the SIP/SEP*.cnf file
What I don't understand is that the phone will request and download the
SIP/SEP*.cnf files which are well over 8.3, and if I change the file
name to an 8.3 format the phone doesn't even attempt to download the
firmware.
-Max
Max Clark
max [at] clarksys.com
http://www.clarksys.com
BJ Wesch
Some things that I might check:
1) make sure you have set the jumper to E1 or T1
Mine came configured for E1 and I needed T1.
2) try to see if you can get it working with a different
configuration.
i.e. My configuration for a wcte11xp configured for a
T1 channel bank is:
loadzone = us
defau
Tks
- Original Message -
From: "Tim Greiser" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Wednesday, February 09, 2005 5:16 PM
Subject: Re: [Asterisk-Users] SIP / IAX ActiveX
Colin Anderson wrote:
Actually, I have an application for a IAX Acti
There is the red asterisk near of G.729a which is footnotes to this note.
Adrian Chapman wrote:
Sergey Kuznetsov wrote:
Adrian,
Have you ever read the note for that?
=head Comment
* NOTE
G.729a for X-PRO Pocket PC is available with a minimum
order of 20,000 units,
G.7
Thanks ED, initially I suspected that IAX/FWD was down but I realized
that I couldn't connect to VOICEJET service either so I realized that
something was wrong with my system.
I'm using Gentoo ebuild *-1.0.5 and after recent upgrades the IAX
registration stopped working.
I re-emerged * again and
BJ,
I tried removing the SEP* files and changing the OS79XX.TXT to the 7.3
version of the firmware and then got the message below in my tftpd server.
How do I check the version of the Firmware that the phone is running?
Based on what you said I have a feeling that I need to go way back in
versi
And for Windows, a minimum purchase of 1 unit... Is he using Mac or
PocketPC? If not, then he doesn't have to worry.
On Wed, 09 Feb 2005 15:59:45 -0500, Sergey Kuznetsov
<[EMAIL PROTECTED]> wrote:
> Adrian,
>
> Have you ever read the note for that?
>
> =head Comment
>
> * NOTE
>
Sergey Kuznetsov wrote:
Adrian,
Have you ever read the note for that?
=head Comment
* NOTE
G.729a for X-PRO Pocket PC is available with a minimum
order of 20,000 units,
G.729a for X-PRO Mac OSX is available with a minimum
order of 10,000 units,
G.729a
On Wed, 2005-02-09 at 20:15 +, Gary Stimson wrote:
> On Wednesday 02 February 2005 23:38, Jens Vagelpohl wrote:
> > On Feb 3, 2005, at 0:03, Miguel Ruiz Velasco Sobrino wrote:
> > > --- [EMAIL PROTECTED] wrote:
> > >> The DNS approach does not handle single or multiple system failures,
> > >>
I need to determine the state of extensions/port based on active
channels on my asterisk. For SIP and AIX extensions this is not a real
problem because the channel names they create relate to the SIP/AIX
extensions (just strip the '-' sign and anything after). For ISDN
extensions addressed by Z
You're probably going to want to go ahead and remove the SEP.cnf.xml file from the TFTP server and leave the SIP.cnf file in place.
I think with version 6 or 7 of the firmware (don't recall which one -
probably the one that starts using the Universal App Loader) the phone
starts looking for the
Hi,
I'm experiencing some voice delay (2-3 sec) after outgoing call is setup. It
means during the first 2-3 secs, audio is very choppy or nothing. So usually
I can't hear the 'Hello".
I use IAX2 for my ougoing calls with Grandstream phone as a client. Any
hints to prevent this?
Thanks,
David
_
Adrian,
Have you ever read the note for that?
=head Comment
* NOTE
G.729a for X-PRO Pocket PC is available with a minimum
order of 20,000 units,
G.729a for X-PRO Mac OSX is available with a minimum
order of 10,000 units,
G.729a for LindowsOS i
On Wed, 9 Feb 2005 14:59:44 -0500, Noah Miller <[EMAIL PROTECTED]> wrote:
> Hi Walid -
>
> > I need to use Asterisk to call out PSTN numbers via an analogue line. I
> > understand Digium manufactures these kinds of cards, but can someone
> > tell me
> > which model number it is. I really only need
Daniel Eboa wrote:
Sir,
I think when somebody asked a question, is because he doesn't know the
> answer. Even maybe when for some people like you, the answer is
> evidence.
> Thinking that I know the answer of the question I asked, suppose that
> I'm stupid, while I'm not.
> I you feel offence by
On Wed, 9 Feb 2005 13:46:41 -0600, Anton Krall
<[EMAIL PROTECTED]> wrote:
> Guys, Im new to asterisk and voip but Im have a couple of questions
> regarding the initial setup.
>
> 1. Im going to install an asterisk server at home, where I have 2 phone
> lines, what kind of card do I need to get? I
Hello List,
i am trying to set up an * box with a debian 2.6.10 Kernel.
the compilation of asterisk and the zaptel drivers worked almost perfectly.
so i did
modprobe zaptel
modprobe wcte11xp (which is the correct driver for my card)
ztcfg -v
and this results in
ZT_SPANCONFIG failed on span 1: No su
On Wed, 9 Feb 2005, [EMAIL PROTECTED] wrote:
> Peter,
> Do you know what the current status of app_intercept is?
No, not really. See below for the errors listed.
> I got it working on 1.0.2, but can't get it to complie on
> CVS-HEAD-01/26/05-02:14:44
> I get:
> app_intercept.c: In function `int
try adding something like this to 'zap-in'
exten => i,1,Background(invalid)
On Wed, 2005-02-09 at 10:44, Liaan vd Merwe wrote:
> Hi all
> I'm being really stupid today.
> i simply want asterisk to answer a incomming call, then wait for
> digits dialed. and then dial that extenstions
> but i kee
Very interesting,
I had disabled that previsouly on my SPA but if I am the midpoint
between an inbound SIP connection and outbound IAX connection any
silence detection is not on my end it would be with the company
providing the DIDs, correct?
Or is IAX using it and I'm not aware.
Cheers,
On W
Hi all,
So I have been reading through the docs available online and the
different threads on this list, but I cannot seem to get this phone to work.
I have configured the OS79XX.TXT and SIP/SEP*.cnf files (see attached),
when I configure the phone to point to my tftp server and reboot it I
get
On Wednesday 02 February 2005 23:38, Jens Vagelpohl wrote:
> On Feb 3, 2005, at 0:03, Miguel Ruiz Velasco Sobrino wrote:
> > --- [EMAIL PROTECTED] wrote:
> >> The DNS approach does not handle single or multiple system failures,
> >> only very elementary load balancing over a lengthy period of time.
I just have one problem with that. I have _never_ (that I know of) seen
a phone with A-D on it!
They were (are?) mostly used for military installations.
Jeff
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I have been qualityhaving problems although I don't have all the
details at this point - I currently suspect mine are due to Latency
issues as both ends are colo boxes which don't seem to drop packets in
testing (ping).
Latency right now seems to be in the 80ms range on each end of the server.
Th
Hi,
For various reasons a customer of mine is moving from a SER-based to an
Asterisk-based installation, mostly because of problems with SIP devices
behind NAT trying to reach each other and because it's easier to do
accounting when all calls go through Asterisk (canreinvite=no is the idea).
Ty Carter wrote:
Anyone using a MAX TNT for VoIP terminsation and handing off, via SIP to
asterisk? Any experiences would be appreciated.
None of the 49 results from Google were helpful to you?
http://www.google.com/search?hl=en&q=site%3Alists.digium.com+%22MAX+TNT%22&btnG=Google+Search
__
Hi Walid -
I need to use Asterisk to call out PSTN numbers via an analogue line. I
understand Digium manufactures these kinds of cards, but can someone
tell me
which model number it is. I really only need a card with one or 2
analogue
ports max.
You'd be looking for the TDM400P. You can get it i
Peter Svensson wrote:
Perhaps the app_intercept patch would work better? It is a lot less of a
kludge and more flexible than the *8 that is in Asterisk.
Peter,
Do you know what the current status of app_intercept is?
I got it working on 1.0.2, but can't get it to complie on
CVS-HEAD-01/26/05-0
I get a little confused about IAX and SIP, are those 2 diff. protocols? For
example, I know IAX2 works behind NAT or firewalls right? So that users with
IAX softphones or phones can register into asterisk boxes behind firewall or
them been behind a firewall?
Is this true? Then what is SIP, another
Hi,
On Wed, 2005-02-09 at 14:16 -0500, Andrew Thompson wrote:
> > Actually, A-D digits would be excellent to use instead of # and whatever
> > for transfer. Having a separate code to do '#'-transfer without having
> > to use any regular key on the phone would take away my objections to the
> > ent
They also have a .tar.gz file if you don't want to use the iso
and my understanding is that the .tar.gz file does not nuke
your harddrive.
Jon.
On Wednesday 09 February 2005 11:04 am, Brett, Gary wrote:
> Thanks Dean, you say that " [EMAIL PROTECTED] is just an automated way of
> installing AMP
Guys, Im new to asterisk and voip but Im have a couple of questions
regarding the initial setup.
1. Im going to install an asterisk server at home, where I have 2 phone
lines, what kind of card do I need to get? I was thinking about 2 X100P
Cards, so 1 can have 2 FXO ports and regarding phones, wh
Anyone using a MAX TNT for VoIP terminsation and handing off, via SIP to
asterisk? Any experiences would be appreciated.
Also anyone using HP Blade servers??? BL20g2?
Thanks.
Ty Carter, President
Strategic Network Consultants, Inc.
524 East 9th Street
Washington, NC 27889
252-946-0351 - Voice
Hi Vince -
My next goal is to setup 1 SIP channel, and be able to call the
Asterisk PBX
from a softphone.
Then setup 2 SIP channes and be able to call one from another.
What is the best open source softphone software available for this?
And what is the best documentation source for finding out h
On Wed, 9 Feb 2005 18:42:27 -, Mike Wright <[EMAIL PROTECTED]> wrote:
> Unfortunately I seem to have another problem!
>
> I am using sipgate for the incoming line - and it appears that you cannot
> get DTMF to work in that configuration. Unless anyone knows anything
> different of course!!
I'
Michael Manousos schrieb:
...
Use Asterisk-1.0.3 with asterisk-oh323-0.6.5.
Hi,
may I ask, whether that combination runs really stable
at your machine?
I have now those versions installed.
I have asterisk crashes at least once every hour, when
several simultanious calls take place.
Roger.
On February 9, 2005 02:16 pm, Andrew Thompson wrote:
> I just have one problem with that. I have _never_ (that I know of) seen
> a phone with A-D on it!
Don't you have the Kracker Jax whistle?
-A.
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> Why not share with the community?
I do, like I do with my IAX2 softphone. It's just that I haven't took
the time to make a webpage that explains what it does and provide a
link to download it.
I already send it to peoples on this list that asked for it.
Anybody want it, just email (privately, s
Yeah, I wouldn't mind having a look at it as well.
--
Dana
On Wed, 9 Feb 2005 09:36:27 -0800, Michael Levenson
<[EMAIL PROTECTED]> wrote:
> Why not share with the community?
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Tom Chandler
> Sent: Wed
> All I want is the web config tool ! Apologies if I am misunderstanding you
> here, as I say I am quite new to this and need to get up to speed fast
>
> For an Admin only web based product, is AMP my only option ??
If you want a web based config files editor, I have done one. But it's
not advance
Hello there,
I've successfully compiled the G729 DLL for FireFly. However, I'd
like to hook up some customers using this, and I'm not sure how I can
license it. Everything seems to point to buying thousands of G729 licenses
at a time (from Sipro), and the Virbiage site doesn't mention anyt
Colin Anderson wrote:
Actually, I have an application for a IAX ActiveX control myself. Rather use
IAX because of the whole NAT-traversal issue. I found this:
http://lists.digium.com/pipermail/asterisk-users/2002-August/003766.html
But little else, and I can't find the source. Anyone else found thi
Florian Overkamp wrote:
Hi,
On Wed, 2005-02-09 at 09:53 -0700, Kevin P. Fleming wrote:
They don't. Most phones (99.9%) don't have any way to generate DTMF A
through D. There are test sets that do, and of course softphones could
easily do so. These tones could also be generated by automated
appli
I am using X100P and the files that come with the
samples (asterisk 1.0.5):
zaptel.conf:
-
;
; Zapata telephony interface
;
; Configuration file
[trunkgroups]
;
;
For
softphones, I used SJPhone, is a very good SIP phone, and I have it working on
asterisk. The setup is kind of tricky, 'cause you must remember to set the sip
register in your local DNS.
For a
good overview and introductory tutorial to asterisk, go to the asterisk home
site (www.asterisk
On Wed, 9 Feb 2005 11:44:30 -0500, Paul Rodan <[EMAIL PROTECTED]> wrote:
>
>
> I know FWD uses a SER/Asterisk combo, and I keep hearing about the massive
> benefits, however, my initial playing around in SER's configuration
> indicates it's NOTHING like Asterisk at all, and almost 5x as difficult
Apologies for the double post in re my SPEEX issues, but can anyone
report successfully installing libOgg and getting SPEEX to work on
Asterisk?
Details: Fedora Core 3, Asterisk 1.0.4, VoicePulse Connect.
Here's my original query with the CLI log
http://lists.digium.com/pipermail/asterisk-user
Hello all.
I'm still investigating the cause of freezes on my
asterisk server. It's a minimal installation: the only
things I remember running are httpd, sshd, sendmail
and asterisk itself. I have a DID from Voicepulse. No
telephony cards or SIP phones ... I'm just trying to
figure out the voicema
Thanks Noah
I got the source with CVS to a Windows machine, this is the
source causing the problem, although I suspect that getting the files to
Windows and then copying them to Linux was not a good idea.
I then got the tarball files, unzipped them on Linux and
compiled and everythi
No quite sure what you mean by timimg source. But I have a T100P card
in the box and zaptel drivers loaded.
On Wed, 9 Feb 2005 13:49:15 -0500, Paul Rodan <[EMAIL PROTECTED]> wrote:
> That jitter buffer has caused nothing but problems for me.
>
> But that was a few months ago, haven't tried it la
Hi,
Just want to
understand the difference between Asterisk Versions and please correct me if I
am wrong, I understand they are:
Stable
CVS
CVS
Head
I am a newbie and
about to install Asterisk on SUSE Server. Can someone please advise what is the
best version type and number should
Sorry all
figured out one needs to include the context that
the extensions to be dialed into the current context
thanks
Liaan
__Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com _
That jitter buffer has caused nothing but problems for me.
But that was a few months ago, haven't tried it lately.
What are you using as your timing source?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian Dingman
Sent: Wednesday, February 09, 200
Unfortunately I seem to have another problem!
I am using sipgate for the incoming line - and it appears that you cannot
get DTMF to work in that configuration. Unless anyone knows anything
different of course!!
> > I just want one of my incoming numbers to go to an IVR service that
> > will allo
Hi all
I'm being really stupid today.
i simply want asterisk to answer a incomming call,
then wait for digits dialed. and then dial that extenstions
but i keep on getting: WARNING[3314]: pbx.c:2017
ast_pbx_run: Invalid extension '5', but no rule 'i' in context
'zap-in'
my config:
exten => 0
Guys,
I need to use
Asterisk to call out PSTN numbers via an analogue line. I understand Digium
manufactures these kinds of cards, but can someone tell me which model number it
is. I really only need a card with one or 2 analogue ports
max.
Walid
_
Joseph,
FWD's experimental IAX service has been very stable for the last
several months. The problem that occurred during November
(caused by using an unstable build) has been corrected.
Here is your entry:
[EMAIL PROTECTED] asterisk]# asterisk -rx 'iax2 show peers' | grep
491581
1 - 100 of 221 matches
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