For those who don't wake up at 5.00 am and start reading /.
http://searchnetworking.techtarget.com/originalContent/0,289142,sid7_gci935769,00.html
--
Dave Cotton <[EMAIL PROTECTED]>
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Here is the scenario.
i have a meeme conference extension at 8000.
now i want to dial out and call tom (9000).
Once this call is established, i want to connect him to the conference
extension, so that
he can participate in the conference.
How can i do this in asterisk? what is the sequence of c
On Fri, 2003-11-07 at 23:52, Eric Wieling wrote:
> As I understand it for large numbers of channels the TelCo usually
> provides a DS-3 or higher to the customer. The customer either connects
> that directly to their equipment (if their equipment supports it) or
> breaks the DS-3 out into multiple
On Fri, 2003-11-07 at 23:30, Darren Martz wrote:
> Thanks Brian, and thanks again for the included definitions - that
> helped too. Your comments are really helping clear many questions.
>
> I suppose our intensions are to become an IXC.
>
> So if my local carrier is sporting old technology, the
On Friday 07 November 2003 19:50, Michael Koehler wrote:
> 1. can someone please quote the text from this restricted page which
> is linked below to the list. could be helpful for some.
It's a patch that I wrote to allow MOH to access an MP3 stream.
> 2. just for the stats, i prefer html
So what
As I understand it for large numbers of channels the TelCo usually
provides a DS-3 or higher to the customer. The customer either connects
that directly to their equipment (if their equipment supports it) or
breaks the DS-3 out into multiple T-1 channels.
As for the actual signaling it's either M
Thanks Brian, and thanks again for the included definitions - that
helped too. Your comments are really helping clear many questions.
I suppose our intensions are to become an IXC.
So if my local carrier is sporting old technology, they'll provide TDM
services. So if I understood you correctly,
Thanks Brian, and thanks again for the included definitions - that
helped too. Your comments are really helping clear many questions.
I suppose our intensions are to become an IXC.
So if my local carrier is sporting old technology, they'll provide TDM
services. So if I understood you correctly,
Hi,
I'm using the SPA-2000 with firmware 1.06 on the Asterisk PBX, which works
great for taking and placing calls, but for for some
reason I can't seem to clear the stutter dialtone by either calling the
extension I'm on, or the voicemail system on the Asterisk PBX.
If I call my voicemail acc
> I was working in an env that had some Snom 200 with release 'r'
> firmware i beleive, when thy put the phone in DnD, mode then
> released it from DnD mode, the phone does not re-register with *
>
> Does anybody else observe the behavior ?? Is is a config issue
Doing a "sip debug" suggests the D
> > > >My Snom 200 (about 2 months old) is running v2.02q from a couple of
> > days ago,
> > > >and the speakerphone is fine. Have not tried a headset with it though.
> > > >
> > > >Just ran a short test from a 7960 to the 200, both on the same wire, to
> > > >validate the quality.
> > > >
> > > >
May be I'm missing something here; there seems to be two different
approaches to MusicOnHold. The first is simply any station pressing
the hold button and the holding party hearing music. That seems to be
rather automatic and relies on a single musiconhold.conf statement.
The second approach is s
Darren,
The answer (unfortunately) is "sort of" and "it depends"
- Definitions -
LEC = Local Exchange Carrier
CLEC = Competitive LEC
IXC = Interexchange Carrier (LD company)
---
In most cases a traditional phone company is going to want to hand off
TD
It takes 5 seconds or less to log into that site as an anonymous user.
Just for the record I generally delete HTML mail on a mailing list
without reading it.
On Fri, 2003-11-07 at 19:50, Michael Koehler wrote:
> 1. can someone please quote the text from this restricted page which is
> linked bel
1. can someone please quote the text from this restricted page which is
linked below to the list. could be helpful for some.
2. just for the stats, i prefer html
John Todd wrote:
Hi All,
I keep asking things as they come into my head.
Is there any way to grab an audio stream and pipe it out as t
Hi All,
I keep asking things as they come into my head.
Is there any way to grab an audio stream and pipe it out as the MOH?
I am a helper at a local Charity Hospital Radio Station and thought
it would be nice to pipe the studio output to waiting callers.
Dave
Dave -
1) Please don't post HTM
You hit the FLASH button on your phone. You will then get a dialtone
(assuming you have threewaycalling enabled in zapata.conf. You can then
call another person talk to them, or whatever. If you hit flash a
second time you'll be on an threeway call, if they hangup and you do a
flash you should s
Pardon my ignorance, but I was hoping someone could clear up something for
me.
- For a few POTS lines, digium has a single port card for that, or a T1 card
to a channel bank.
- For 10 or more lines, digium has a T1 or E1 card for that too based on PRI
channels
- For 100's to 1000's of lines, I sus
On Fri, Nov 07, 2003 at 04:41:09PM -0600, Steven Critchfield wrote:
> ztdummy and ztrtc don't provide a full zapata device only because they
> don't make a telephony interface, but the implement the timing needed
> for asterisk.
Alas, neither did for me :/ usb-uhci loaded and detected 2 ports fine
Look again, this time with the cvs code.
Jeremy McNamara
Paul Cheng wrote:
THERE IS AN INCONSISTENCY IN THE README FILE THAT IS OUT OF DATE:
Follow the instructions on line below and do NOT issue a "make clean
install" in asterisk/channels/h323 as indicated elsewhere, just issue
a "make" and
Hi All
I was working in an env that had some Snom 200 with release 'r'
firmware i beleive, when thy put the phone in DnD, mode then
released it from DnD mode, the phone does not re-register with *
Does anybody else observe the behavior ?? Is is a config issue
At 03:31 PM 11/7/2003, you wrote:
> >My Snom 200 (about 2 months old) is running v2.02q from a couple of
days ago,
> >and the speakerphone is fine. Have not tried a headset with it though.
> >
> >Just ran a short test from a 7960 to the 200, both on the same wire, to
> >validate the quality.
> >
>
Title: Leterhead
Hi All,
I keep asking things as they come into my head.
Is there any way to grab an audio stream and pipe it out as the MOH?
I am a helper at a local Charity Hospital Radio Station and thought it
would be nice to pipe the studio output to waiting callers.
Dave
Ok.. Example.. I can put them into extension 123 playing MusicOnHold,
but how would I retreive the call when I need to get the caller back?
This is to be done on a analog phone.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ernest W.
Lessenger
Posted At
Thanks all,
Will try to get it up and running this weekend.
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Michael Manousos
Sent: 07 November 2003 10:46
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] H323 Gateway
Alternatively, you may use aste
On 07-11 13:17, John Todd wrote:
From what I can understand of the issue you describe, it sounds like
the problem resides on the remote side, and not Asterisk's side.
You are sending an invalid request in your first query, and the
remote side is sending "Unauthorized", meaning that it believes
> >My Snom 200 (about 2 months old) is running v2.02q from a couple of days ago,
> >and the speakerphone is fine. Have not tried a headset with it though.
> >
> >Just ran a short test from a 7960 to the 200, both on the same wire, to
> >validate the quality.
> >
> >
> Looks like Snom have just rel
Success at last!!! See below for details.
On Thursday, November 06, 2003 1:16 PM, Steven Critchfield
[SMTP:[EMAIL PROTECTED] wrote:
> You need to have an exten entry that matches the number of digits the
> telco is sending. You should be able to see this vrom the CLI with a
> couple of verbose fl
At 03:12 PM 11/7/2003, you wrote:
Is there a way to put a call on hold and play music on hold with out
using the park app?
There is a "MusicOnHold" extension that is like park, except that you can
never take them off hold.
Most SIP phones also have the ability to put a call on hold and tell * to
> On Fri, 2003-11-07 at 11:43, Alexandru Coseru wrote:
> > Can I use a modem and a soundcard as an fxo ?
> >
> > I've read in the documentation something , but how can I do that ?
> >
>
> NO YOU CAN'T. Read the archives, it isn't implemented. Read the flame
> wars to see why it most likely will nev
Is there a way to put a call on hold and play music on hold with out
using the park app?
Thanks,
-gcc
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On 07-11 13:17, John Todd wrote:
> From what I can understand of the issue you describe, it sounds like
> the problem resides on the remote side, and not Asterisk's side.
>
> You are sending an invalid request in your first query, and the
> remote side is sending "Unauthorized", meaning that it
Asterisk was wrong. Every SIP message can be challenged with 401 or 407,
depending on who is challenging.
If you send a REGISTER message then you can get "407 Proxy
Authentication Required" from any proxy along the path of the message.
You can also get "401 Unauthorized" from registrar.
The same
On Fri, 2003-11-07 at 16:04, Olle E. Johansson wrote:
> Steven Critchfield wrote:
>
> >>We have to rename "Zaptel timing" to "Asterisk timer", which is more correct
> >>since there are several ways of getting a timer to work, only one of them
> >>is by using Zaptel cards.
> >>
> >>http://www.voip-
Rich Adamson wrote:
Brian,
My Snom 200 (about 2 months old) is running v2.02q from a couple of days ago,
and the speakerphone is fine. Have not tried a headset with it though.
Just ran a short test from a 7960 to the 200, both on the same wire, to
validate the quality.
Looks like Snom have
First of all great job on diax. I downloaded it and tried it, could not
connect, got an authentication rejected,but I have not had a chance to
figure out why yet - tried with a working gnophone setup in the
configuration files.
Is there any way to pass command line arguements to the program ? W
I haven't done this, but
Use the Linux high availability project's heartbeat software
Configure both servers to watch the same IP.
T each of the incoming PSTN lines to dual x100p cards (one on each server)
Configure one asterisk server to pick up the line on incoming calls a few
seconds
Hi People,
Let's take a look in this diagram :
Part A - Server running VPN IP ie.192.168.10.1
Part B - Client running over the VPN with internal IP ie. 192.168.10.2
--
From network A i can reach B.
Use all programs - Share Printers , aplications, using Netmeeting etc..
Then i make this in t
Hi!
First, I want to let everyone that I am new to the Asterik world and
looking forward to have fun with it.
I am interested in setting up a small pbx. The following are some
pre-requisites:
i.) Would like to have three extension with the capability of expanding
using Hard VoIP phones. (How
Brian,
> Yep, we have 8 SNOM200's in an installation and none of them have a usable
> speakerphone. There is something frelled up with the voice detection on
> those phones when using speakerphone. We talked to kevin in technical
> support for the American Distributor and he told us to try the lat
Steven Critchfield wrote:
We have to rename "Zaptel timing" to "Asterisk timer", which is more correct
since there are several ways of getting a timer to work, only one of them
is by using Zaptel cards.
http://www.voip-info.org/tiki-index.php?page=Asterisk+timer
Actually it needs to be zapata ti
Mireia Munoz de jesus wrote:
Hi!
I am trying to know well asterisk. For that I would like to know the exact role
for each config file. Can someone tell me what is the role of the next ones or
a web where I could find this information? That will be very helpful.
- alsa.conf
- enum.conf
- modem.conf
Dragan Mickovic wrote:
I have couple of questions about the following. Currently I have 2 phone lines going
into my house, and I would like to have both of those coming into asterisk. I also
want to have a backup asterisk, so here are the main questions (I am knew to this so
I apologize if I ask s
mtm spm wrote:
Maybe this is a silly question but I am a beginer with
Asterisk.
I want now to be able to write a script or something
so that I can dial out a number and when the call is
answered to play a .gsm file or an output from
festival.
I need to call this numbers on demand(from another
progr
mtm spm wrote:
Hi Olle,
--- "Olle E. Johansson" <[EMAIL PROTECTED]> wrote:
The first Invite is without credentials, since
digest authentication needs input
from the server to create credentials.
This is also what I understood too from rfc.
I was just confused becouse in the Asterisk code
there
Whenever I have had this problem it was certainly a SIP/Firewall issue.
Calls will go through but audio (RTP) will not go through.
Second, using iconnect I have to add an option r (I think) to my dial
command in extensions.conf to make asterisk send ring back to the
originating phone, because I th
C M wrote:
hi,
i saw the cdr file called Master.csv and i want to
know what these represent. examples
"","","4","incoming","","Zap/1-1","Zap/4-1","Voicemail","u","2003-11-07
17:43:04","2003-11-07 17:43:04","2003-11-07
17:43:22","ANSWERED","DOCUMENTATION"
"","","19373693874","incoming","","Zap/
Yes,
ASTERISK1 has 2x TDM400P and ASTERISK2 has 3 x100P cards in it.
I'll try to dork with the timer, but as long as wcfxo or wcfxs is loaded
shouldn't that take care of these issues?
- Original Message -
From: "Olle E. Johansson" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday,
Greetings. I'm having a bit of a problem using the sipdtmfmode app. I have two
incoming paths to * from pstn via FWD that use differing dtmfmode. IPKall
wants rfc2833, libretel wants inband. If I set dtmfmode= in the fwd peer
config in sip.conf each works seperately, and I'm trying to use gotoi
On Fri, 2003-11-07 at 15:02, Olle E. Johansson wrote:
> Louis-David Mitterrand wrote:
> > On Thu, Nov 06, 2003 at 10:41:15PM -0500, Brian Schrock wrote:
> >
> >>Hello,
> >>
> >>I have searched google, read everything on the mailing list, read
> >>/usr/src/asterisk/README.iax and /usr/src/asterisk/
Yep, we have 8 SNOM200's in an installation and none of them have a usable
speakerphone. There is something frelled up with the voice detection on
those phones when using speakerphone. We talked to kevin in technical
support for the American Distributor and he told us to try the latest beta
firmwar
Hi Olle,
--- "Olle E. Johansson" <[EMAIL PROTECTED]> wrote:
The first Invite is without credentials, since
digest authentication needs input
from the server to create credentials.
This is also what I understood too from rfc.
I was just confused becouse in the Asterisk code
there was something l
Thank you guys very much,
it work great based on sample.call.
Have a nice weekend.
MTM
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Asterisk-Users mai
Michael Manousos wrote:
when i try to make a call with "netmeeting" through * ( * dial out with
"Dial,OH323/[EMAIL PROTECTED]" ) the call will be blocked.
This is a problem of OpenH323 1.12.0. Use this dial string:
Dial,OH323/h323:[EMAIL PROTECTED]
Or, even better, use the latest (it has been fixed
Steven Critchfield wrote:
On Wed, 2003-11-05 at 15:03, Steve Murphy wrote:
Everyone--
Here's a cost analysis, rather crude and inspecific, of using Asterisk
to implement a phone system. I'm really quite naive and new to all this,
so I'd appreciate any corrections, tips, pointers,
On Fri, 2003-11-07 at 14:39, WipeOut wrote:
> Steven Critchfield wrote:
>
> >On Fri, 2003-11-07 at 13:55, WipeOut wrote:
> >
> >
> >>Hi,
> >>
> >>I have been playing around with AGI scripting..
> >>
> >>I have worked out how to initiate a call using "EXEC Dial
> >>channel/number" the problem wi
On Fri, 2003-11-07 at 14:38, Yelson Vivas wrote:
> HI Asterisk Gurus
>
> I'm connecting my * server to a zplex 10B, I'm using a slot on my TE410P
> card as T1 to connect my server to the channel bank (i use a cross cable),
> the zplex doesn't show any alarm neither the server. I'm trying to ma
Louis-David Mitterrand wrote:
On Thu, Nov 06, 2003 at 10:41:15PM -0500, Brian Schrock wrote:
Hello,
I have searched google, read everything on the mailing list, read
/usr/src/asterisk/README.iax and /usr/src/asterisk/doc/iax.txt(?), asked on
the IRC channel and I cannot figure out what is wrong w
THERE IS AN INCONSISTENCY IN THE README FILE THAT IS OUT OF DATE:
Follow the instructions on line below and do NOT issue a "make clean
install" in asterisk/channels/h323 as indicated elsewhere, just issue a
"make" and then in /usr/src/asterisk (or wherever you source is), issue
a "make install"
Interesting. Can you point to where this is documented? I rooted around
thru the Digium online manual, whitepaper, etc, couldn't find any doc.
http://www.voip-info.org/wiki-Asterisk+variables
/O
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http://l
HI Asterisk Gurus
I'm connecting my * server to a zplex 10B, I'm using a slot on my TE410P
card as T1 to connect my server to the channel bank (i use a cross cable),
the zplex doesn't show any alarm neither the server. I'm trying to make
calls from (ext 1) from the channel bank, to other (ex
Steven Critchfield wrote:
On Fri, 2003-11-07 at 13:55, WipeOut wrote:
Hi,
I have been playing around with AGI scripting..
I have worked out how to initiate a call using "EXEC Dial
channel/number" the problem with this is that the script then completes
and does not wait for the call to end..
I have following setup:
AnalogPhone_1--TDM400P--Asterisk---SIP---[Softswitch]POTS-AnalogPhone_2
I can call from AnalogPhone_1 to AnalogPhone_2 and all is fine.
When I call to AnalogPhone_1 from AnalogPhone_2, AnalogPhone_1 rings BUT
I hear no ringing tone AND when someone picks up AnalogPh
So, after playing with 6.0 on the Cisco 7960 and 7940 platforms, I
have the following gripes, which I've sent to a very clueful Cisco
person already. Mind you, I love the Cisco 79xx series phones, and
currently they are what I recommend to anyone who wants a 'real' IP
phone. I just cringe
-
John Todd wrote:
exten => 514777,1,Dial,Zap/2|10
Try:
exten => 514777,1,Dial(Zap/2,10)
I think these two versions of giving arguments are confusing. Reading docs
and "show application " texts, both variants are used, sometimes even in the
same text.
Is the first syntax old, to be repl
On Fri, 2003-11-07 at 13:55, WipeOut wrote:
> Hi,
>
> I have been playing around with AGI scripting..
>
> I have worked out how to initiate a call using "EXEC Dial
> channel/number" the problem with this is that the script then completes
> and does not wait for the call to end..
>
> Is there a
Hi,
I have been playing around with AGI scripting..
I have worked out how to initiate a call using "EXEC Dial
channel/number" the problem with this is that the script then completes
and does not wait for the call to end..
Is there an alternate way to dial the call and then when the call is
co
At 09:20 AM 11/7/2003, you wrote:
I though to it also, but really I don't know how can I get the pid of a
process ran by asterisk.
I mean, the only think I do it's :print "EXEC MP3Player \"$key\"\n";
Then asterisk take the hand with mp3player applications that will launch
mpg123, etc...
You're righ
> Thanks for that - but how does that plug into a sip client - all this
> will do as I understand it is if I forward a call to that extension it
> will play music - how do I get it back - and how do I tie it into the
> hold button on a sip client??
The call parking feature may be more what you wa
Does everything work fine now? I am still having
problems with SayUnixTime. Voicemailmain2 works
though. The one simple AGI script I wrote doesn't do anything. Asterisk starts
playing and the grandstream just rings. Both work fine on other channels/sip
phones.
Thanks,
Will
A simple question.
I found an old Xten Xphone Beta 1.01 that has Sip capabilities.
So i try with my asterisk, but he always try to login using this format:
sip:[EMAIL PROTECTED]:5060
And say that registration failed.
Someone see this kind of thing before ?
I try using Xten Lite and everythi
Thanks William,
Works fine now.
Wim
- Original Message -
From:
William Carlson
To: [EMAIL PROTECTED]
Sent: Thursday, November 06, 2003 9:43
PM
Subject: Re: [Asterisk-Users] Grandstream
problem
try
disallow=all
allow=ulaw
under the general
The broken code sends audio directly to the NAT address. For instance:
When I place a call from the grandstream to * with the broken code
* sees the grandstream.
* according to tcpdump, sends a bunch of UDP (audio?) directly to the PRIVATE IP
(192.168.0.100) of the grandstream even though it is
Hi!
> I want now to be able to write a script or something
> so that I can dial out a number and when the call is
> answered to play a .gsm file or an output from
> festival.
Look at sample.call and the Playback() application.
Cheers, Philipp
___
Ast
On Fri, 2003-11-07 at 11:57, Mireia Munoz de jesus wrote:
> Hi!
>
> I am trying to know well asterisk. For that I would like to know the exact role
> for each config file. Can someone tell me what is the role of the next ones or
> a web where I could find this information? That will be very helpfu
On Fri, 2003-11-07 at 11:43, Alexandru Coseru wrote:
> Can I use a modem and a soundcard as an fxo ?
>
> I've read in the documentation something , but how can I do that ?
>
NO YOU CAN'T. Read the archives, it isn't implemented. Read the flame
wars to see why it most likely will never be imple
server:/usr/src/linux/drivers/isdn# patch -p0 <
../../../isdn-kernel-dtmf-dsp-patch.diff
patching file isdn_tty.c
patch: malformed patch at line 9: (info->emu.vpar[1]))
what can be this??
Matthew Enger wrote:
And a working patch for linux kernel.
On Fri, 2003-11-07 at 09:30, Matthew Enger
Alexandru Coseru wrote:
Can I use a modem and a soundcard as an fxo ?
I've read in the documentation something , but how can I do that ?
Don't bother, you will waste more time which equates to money. Buy a
X100P from Digium and support Asterisk.
Jeremy McNamara
___
see sample.call in the Asterisk source directory
On Fri, 2003-11-07 at 11:52, mtm spm wrote:
> Maybe this is a silly question but I am a beginer with
> Asterisk.
>
> I want now to be able to write a script or something
> so that I can dial out a number and when the call is
> answered to play a .g
Hi!
I am trying to know well asterisk. For that I would like to know the exact role
for each config file. Can someone tell me what is the role of the next ones or
a web where I could find this information? That will be very helpful.
- alsa.conf
- enum.conf
- modem.conf
- modules.conf
- oss.conf:
Maybe this is a silly question but I am a beginer with
Asterisk.
I want now to be able to write a script or something
so that I can dial out a number and when the call is
answered to play a .gsm file or an output from
festival.
I need to call this numbers on demand(from another
program), since th
Thanks for that - but how does that plug into a sip client - all this
will do as I understand it is if I forward a call to that extension it
will play music - how do I get it back - and how do I tie it into the
hold button on a sip client??
Thanks again.
Nick
___
Can I use a modem and a soundcard as an fxo
?
I've read in the documentation something , but how
can I do that ?
Regards
Alex
I have been trying to get the DTA310 to work properly with Asterisk for the
last couple of weeks. It seems to connect but it does not play back any
sound and I
cannot dial it by using x-lite. Sip debug looks pretty good. I was
wondering if someone has a working config that they could post so that i
Hi Olle,
--- "Olle E. Johansson" <[EMAIL PROTECTED]> wrote:
> The first Invite is without credentials, since
> digest authentication needs input
> from the server to create credentials.
This is also what I understood too from rfc.
I was just confused becouse in the Asterisk code
there was somethi
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Dan
> Sent: 03 November 2003 19:18
> To: Asterisk Users
> Subject: [Asterisk-Users] DIAX Soft phone v0.9.1 is available for
> downlaod...
>
> as promise, at:
>
> http://www.laser.com/dante
> or
> http://www.geocities.com/tdanro
I though to it also, but really I don't know how can I get the pid of a
process ran by asterisk.
I mean, the only think I do it's :print "EXEC MP3Player \"$key\"\n";
Then asterisk take the hand with mp3player applications that will launch
mpg123, etc...
How can I get this pid of the good mpg123 p
C M wrote:
hi,
i saw the cdr file called Master.csv and i want to
know what these represent. examples
Take a look in the root of the source (/usr/src/asterisk) at the
README.cdr and in the /cdr/cdr_csv.c file for descriptions of the fields..
please help me. i want to store these into mySQL
data
mtm spm wrote:
Hello,
I have a problem with asterisk when dial out to a SIP
provider.
Asterisk send a INVITE with no credentials, the
provider reply with a 401 Unauthorized.
However, Asterisk DOES NOT resend the invite again
with credentials. But it hangs there (maybe waiting
for a ok)
It is thi
hi,
i saw the cdr file called Master.csv and i want to
know what these represent. examples
"","","4","incoming","","Zap/1-1","Zap/4-1","Voicemail","u","2003-11-07
17:43:04","2003-11-07 17:43:04","2003-11-07
17:43:22","ANSWERED","DOCUMENTATION"
"","","19373693874","incoming","","Zap/1-1","IAX
At 07:01 AM 11/7/2003, you wrote:
Hi !
Now I can hear nice mp3 through my phone... Great :P
And many thanks for your posts. Now it's working fine... hmmm almost !!!
In fact, I m using DialenMP3.agi. It's a real nice agi script...
For those which would not know, you will find it here :
http://aster
anyone around here have the ability to terminate
a .NL phone number to IAX or SIP ??
if so please contact me off list.
thank you
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Hello,
Probably that would help you :
http://www.voip-info.org/wiki-Asterisk+cmd+Musiconhold
Cheers,
Areski
On Fri, 2003-11-07 at 16:37, Nick Knight wrote:
> Hello all,
>
>
>
> I am trying to suss out music on hold feature - hopefully to integrate
> nicely with SIP phones and the hold butto
Or maybe like this:
(I do not know)
Exten => 514777,1,Dial,Zap/22|10|r
- Original Message -
From: "Lal, Deepak (Contractor)" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, November 07, 2003 12:07 PM
Subject: RE: [Asterisk-Users] No ringing tone
> I did put "r" as in (in
Dan,
You are doing a fine job with your DIAX project. Just in case your
English is a second language and you haven't heard this colloquialism
On Wed, 2003-11-05 at 12:12, Dan wrote:
> Hi Gary,
>
> - Original Message -
> From: "Gary" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent:
>minutes, hours, days... ?
It happens after about 6-7 hours. The problem is very consistent.
I am also running 1.0.3.81 firmware on the phones.
Perhaps this is a config problem with ntpd?
here is my ntp.conf:
server clock.isc.org
server time.nist.gov
restrict clock.isc.org mask 255.255.
To be more accurate: Can I use asterisk as a Call Agent/Media Gateway Controller
for a third party Media Gateway? E.g. if I have
PABX (not *)---[ Some 3-party Media Gateway ] -- AAL2 -->to somewhere
Can I use Asterisk to "manage" the 3-party Media Gateway using MGCP ? In other
words:
PABX (no
Hello all,
I am trying to suss out music on hold feature - hopefully to integrate
nicely with SIP phones and the hold button you find on most - can some
give me some points on how to get started?
Regards
Nick
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I did put "r" as in (in extensions.conf):
Exten => 514777,1,Dial,Zap/2r2|10
But this still does not help!!
Thanks - DL
-Original Message-
From: Bartosz Jozwiak [mailto:[EMAIL PROTECTED]
Sent: Friday, November 07, 2003 9:59 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] No
Hello,
I have a problem with asterisk when dial out to a SIP
provider.
Asterisk send a INVITE with no credentials, the
provider reply with a 401 Unauthorized.
However, Asterisk DOES NOT resend the invite again
with credentials. But it hangs there (maybe waiting
for a ok)
It is this a bug in aste
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