CW_ASN wrote:
Here is my example. I'm using a lot of times a day.
?php
$socket = fsockopen(192.168.0.53,5038, $errno, $errstr, $timeout);
fputs($socket, Action: Login\r\n);
fputs($socket, UserName: admin\r\n);
fputs($socket, Secret: blabla\r\n\r\n);
fputs($socket, Action: Command\r\n);
Actually asterisk.org has
all the info you need.
Just install the linux
distrib with CVS, kernel sources and openssl-devel and all their dependencies.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew England
Sent: Monday, November 03, 2003
3:56 AM
To:
Look into www.digium.com.
Digium's cards are you best choice.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of brez
Sent: Monday, November 03, 2003 4:03 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Newbie Questions
hello,
I am completely new to
Robert Mann wrote:
Problem I have is this. outside firewall (extension 2003) can call me
inside firewall (extension 2000) and all is fine. If I call from
inside firewall (extension 2000) to outside firewall (extension 2003)
I hear no ringing and person at other end can pick up and I hear for
Hi Brian,
- Original Message -
From: Brian West [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, November 02, 2003 11:54 PM
Subject: Re: [Asterisk-Users] New IAX software phone (for WIndows platform)
...
Its a start but having to restart when you change registration isn't very
WipeOut wrote:
So the basic rules where NAT is involved are..
Asterisk server must always be on a public IP address..
You keep saying this, but it is not correct.
I have several asterisk servers running behind NAT servers, and they
function perfectly.
I won't say configuring them was as easy
Please provide your feedback about the application
Only in that way it can be improoved.
Thanks!
Dan
- Original Message -
From: Senad Jordanovic [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, November 03, 2003 12:13 AM
Subject: RE: [Asterisk-Users] New IAX software phone (for
Thanks Jose/Tom for responding to my Newbie questions. its much clearer
now. anyhow on to the next [unrelated question] here's the use case:
i will need one machine that will answer incoming calls - store the
caller's number [caller ID] and then prompt the caller to answer a
question by using
Hi ,
- Original Message -
From: Masakazu Nakano [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, November 03, 2003 5:05 AM
Subject: Re: [Asterisk-Users] New IAX software phone (for WIndows platform)
...
thanks for good application!
and I wish 'no with installer' package about
On Sun, Nov 02, 2003 at 06:20:15PM -0500, Andrew Kohlsmith wrote:
So here's my humble request.
Can someone who has implemented a live production Asterisk deployment,
preferably between two sites (HQ and a branch office, connected over the
internet) spare the time and contact me here, or to
Look into AGI, there a re some examples out there, but it's very much
doable.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of brez
Sent: Monday, November 03, 2003 11:30 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Another newbie question
Thanks
Hi Matthew,
On 2 Nov 2003, Matthew Enger wrote:
exten = _004,1,Dial(modem/g1/V${EXTEN:1})
Try this Dial command:
Dial(Modem/ttyI0:${EXTEN:1})
msn=0397468733L*
Try removing L* from the MSN, it looks wrong to me.
You might find that ttyI0 and ttyI1 are both channels of the first
On 31/10/03 12:11, Senad Jordanovic wrote:
You are right, but what if each * server had a single source for all
of its configuration files from a file server over NFS or similar.
Single point of failure at the file server. Better to rsynch all the
machines config files or similar.
--
Alastair
I cannot seem to get the software to work on my machine. I am multihomed
running windows XP home. Perhaps the software is binding to the card not
connected to asterisk. If I turn on debugging in asterisk I see no IAX stuff
coming in from the IP.
Thanks,
Will
- Original Message -
If your DSL link is the bottleneck, rather than earlier hops back
through the providers network, the provider could also prioritize VOIP
packets going up the DSL line. That requires a cooperating provider,
of course.
You may also setup a linux box (or another QoS supporting router) on the
Shoval Tom wrote:
Isn't putting asterisk on the public IP network a bad idea?
Is it a bad idea?, Not really if you take the right precautions..From
how you described your setup you have connected your server directly to
the internet anyway.. If you nominated you Asterisk box as the DMZ host
Hello Olle,
Olle, I can't reach the faq page, and haven't been able to for the
last four days.
I'm getting 504 gateway timeout errors.
Gateway timeout indicates something with your web proxy ...or?
I've been able to reach the Wiki all weekend, I've updated and created
several pages...
I also now
On 03/11/03 00:25, Mark Spencer wrote:
As a side note, I strongly would like to see someone implement a
client using libiax2 which implements IAX2 instead of the (now
obsolescent) IAX version 1.
I'm implementing a Java-based IVR server (and yes, I know Asterisk does
IVR, and no, it's not flexible
Hi Will,
- Original Message -
From: William Carlson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, November 03, 2003 12:31 PM
Subject: Re: [Asterisk-Users] New IAX software phone (for WIndows platform)
I cannot seem to get the software to work on my machine. I am multihomed
Hello,
With help from Adam, I managed to get it working. I have put in the bits
which might help others in the future below:
Kernel:
I had to compile a fresh kernel source and apply the voice patch
available from www.traverse.com.au.
Since I had 2 cards, I did a modprobe in my boot scripts of
I did set it up to register here is my iax.conf config.
[blah]
type=friend
user=blah
secret=blah
context=default
host=192.168.5.200
This is what I am seeing in asterisk.
NOTICE[32773]: File chan_iax.c, Line 2708 (register_verify): Peer 'blah' is
not dynamic (from 192.168.5.200)
Rx-Frame
Hi,
- Original Message -
From: William Carlson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, November 03, 2003 1:15 PM
Subject: Re: [Asterisk-Users] New IAX software phone (for WIndows platform)
I did set it up to register here is my iax.conf config.
[blah]
type=friend
I made few successful calls (in/out).
However, the application did crash few times during conversation, and
now while trying to start it the application shows this error message:
Run Time error '341':
Invalid control array index
I am using XP-PRO, Service pack 1.
So here's my humble request.
Can someone who has implemented a live production Asterisk deployment,
preferably between two sites (HQ and a branch office, connected over the
internet) spare the time and contact me here, or to my email directly?
As a lurker, I would very much
WipeOut wrote:
Shoval Tom wrote:
And how will all us newbies make the linux box as secure as possible?
The quickest way is to setup an IPTABLES firewall.. You will need ports
5060 and 1 to 2 open for a default Asterisk install using SIP
only..
Visit the Wiki page
Hi,
- Original Message -
From: Senad Jordanovic [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, November 03, 2003 1:39 PM
Subject: RE: [Asterisk-Users] New IAX software phone (for WIndows platform)
I made few successful calls (in/out).
However, the application did crash few
Peer Oliver schmidt wrote:
I have not been able to reach www.voip-info.org as well using my default
settings.
Upon researching the problem I tried
nslookup www.voip-info.org
which returns an IP address of
192.168.168.3
which is obviously wrong. This is the answer from my local DNS server
Robert Hajime Lanning wrote:
quote who=WipeOut
Sharing the config files is the smallest problem.. its sharing SIP
session and reistration information that is more of an issue..
And managing the data flows..
I wouldn't really worry about that. What happens when current PBX's
fail. Does
Hi Dan,
Another problem I am seeing is I cannot delete any
phone book entrys.
This is very strange...
Someone else with this issue?
I cannot reproduce it here
Just tried to delete Entry 12. Same problem here. And afterwards I can't
start DIAX anymore, except by manually editing diax.cfg and
2. legal issues (what happens when an employee needs to call
emergency personnel and the phone system doesn't work for whatever
reason)
I was worried about this until I realized that commercial systems were _no_
different in this regard. My current Meridian system is a PC in a fancy
box
On Mon, 2003-11-03 at 11:27, Rich Adamson wrote:
A couple of items to consider (in addition to the technical * implementation
issues) are:
Many thanks for your input, Rich - fortunately many of these issues
don't pertain to our own environment, but they're still highly worth
pointing out, and
Hello,
On Fri, 31 Oct 2003 10:24:32 -0600, David Gomillion wrote
I can understand the size concerns for putting it in an
appliance or what-not. However, my opinion is that, due to
the low cost of hard disk space, it is cheaper for the
company to go out and buy another hard disk to replace
Hi,
- Original Message -
From: Peer Oliver schmidt [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, November 03, 2003 2:28 PM
Subject: Re: [Asterisk-Users] New IAX software phone (for WIndows platform)
Hi Dan,
Another problem I am seeing is I cannot delete any
phone book
Problem I have is this. outside firewall (extension 2003) can call me
inside firewall (extension 2000) and all is fine. If I call from
inside firewall (extension 2000) to outside firewall (extension 2003)
I hear no ringing and person at other end can pick up and I hear for
maybe a
WARNING!!! (from bilbo.inter.net.il)
The following message attachments were flagged by the antivirus scanner:
Attachment [2.1] , scan failed: Internal error (0x11). Action taken: incomplete scan
My asterisk server is inside my LAN. Our branch office is connected to here
via VPN tunnel,
That is correct. I'm able to get to your site using the IP address provided
below.
Since I get the same address (192.168.168.3) from four different ISPs (home,
HQ, branch office, and dial-up to another one) I think it's safe to say your
DNS configuration is what should be looked at first.
On Mon, 2003-11-03 at 13:07, Roy Sigurd Karlsbakk wrote:
hi all
anyone that've heard of any working IAX hardphones yet?
There is an unofficial firmware for the SNOM phones:
http://www.marko.net/asterisk/archives/0208/0158.html
although that thread seemed to go nowhere - may be worth chasing
On Mon, 2003-11-03 at 07:07, Roy Sigurd Karlsbakk wrote:
hi all
anyone that've heard of any working IAX hardphones yet?
During Phreaknic, Mark showed the IAXY(sp?) a small maybe 2x3 board
that was a single analog FXS port to IAX adapter.
It was impressive that he was able to come into the
Dear Listers,
I have the following problem:
me and my father try to use 2 gnophones to talk to each other. We both
registered at Iaxtel and both can call other numbers --- say FWD
numbers, but when one of us tries to ring the other's gnophone, we get
the the party you are trying to call is
I'm using * under RH9...
When I go into production, I'll probably be changing distros, though.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David
Sussman
Sent: Saturday, November 01, 2003 7:16 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Quick
2. legal issues (what happens when an employee needs to call
emergency personnel and the phone system doesn't work for whatever
reason)
I was worried about this until I realized that commercial systems were _no_
different in this regard. My current Meridian system is a PC in a fancy
On Mon, 3 Nov 2003, Senad Jordanovic wrote:
The voicemails servermail and fromstring variables should change
default
values when email voicemail notification gets received by user.
I change it, but received mail still shows Asterisk PBX in place of
fromstring.
Anyone knows is there anything
Hi All,
I am attempting to setup Asterisk with sipcall.co.uk. They use Intertex
kit to provide the SIP service. Unfortunately Asterisk cannot seem to
authenticate against Intertex. Having provided SIP debug info the
provider has informed me that Asterisk does not appear to support 'qop',
'nc' and
Did you reload after you made the change?
dave
Yes, many times.
BTW, The servermail variable works fine.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
On Mon, Nov 03, 2003 at 11:32:11AM +0100, Roy Sigurd Karlsbakk wrote:
If your DSL link is the bottleneck, rather than earlier hops back
through the providers network, the provider could also prioritize VOIP
packets going up the DSL line. That requires a cooperating provider,
of course.
Senad Jordanovic wrote:
The voicemails servermail and fromstring variables should change
default
values when email voicemail notification gets received by user.
I change it, but received mail still shows Asterisk PBX in place of
fromstring.
Anyone knows is there anything else needs changing?
Hi all,
can somebody tell me where i can get the g.723 codec for * ?
Thanks.
Regards,
Thomas.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
AFAIK these only work with voicemail2.. check your extensions.conf and
make sure you are using voicemail2 and not just voicemail..
Yap, that did it. :)
Ta
Senad
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
On Mon, 2003-11-03 at 14:28, Thomas Haeger wrote:
Hi all,
can somebody tell me where i can get the g.723 codec for * ?
http://store.yahoo.com/asteriskpbx/asteriskg729.html
$10 per channel. I looked into the licensing costs for another product,
and this is damn cheap.
Cheers,
Gavin.
Hello all,
I have a half working configuration:
I have an asterisk box with one GS101 register to it in SIP mode and an
IP10S in MGCP mode.
I can dial IP10S from my GS101 and everything seems fine.
But from my IP10S I can't dial any number (GS or anything else).
All the version I use are the
This is the g.729 codec, but i want the g.723
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Gavin
Hamill
Gesendet: Montag, 3. November 2003 15:44
An: [EMAIL PROTECTED]
Betreff: Re: [Asterisk-Users] Where can i get the g.723 codec?
On Mon,
Andrew Kohlsmith wrote:
tested the 911 capability of * and, using an extension trick given from the
#asterisk IRC channel, dialling 911 just plays You will dial 911 in 5
seconds. If this was done in error, hang up now before actually zapping a
trunk line (if all are busy) and dialling out.
I am new to this so smack me if I am wrong but shouldn't it be serveremail not
servermail?
Maybe serveremail being wrong causes the fromstring not to function and the
default * is using just happens to be the same thing your serveremail is set to.
Robert
- Original Message -
From:
On Mon, 2003-11-03 at 14:51, Thomas Haeger wrote:
This is the g.729 codec, but i want the g.723
Apologies - was too quick to jump :)
I'm not aware of there being any G.723.1 codec pre-licensed for use with
Asterisk.
The code won't be hard to find, and will probably be publically
Hi ,
I even think to avoid using an installer mainly because the installer
part is bigger that the application himself.
What do you think?
Dan,
I agree that if an installer or registry entries are not needed then it
makes an automated rollout much easier. Also makes it possible to run
Ji,
-Original Message-
I have an asterisk box with one GS101 register to it in SIP
mode and an
IP10S in MGCP mode.
I can dial IP10S from my GS101 and everything seems fine.
But from my IP10S I can't dial any number (GS or anything else).
Is the callmanager setting on the
Licensing info for the G723.1 codec, direct from the holding company
that licenses the codec.
http://www.dspg.com/technology/LicensePricing.html
As you can see they want a LOT of money. This is why I doubt there will
ever be G.723.1 codec available for Asterisk.
--Eric
On Mon, 2003-11-03 at
Can you post the trick, as far as the zapping a channel and what not?
That's something I've been looking for...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Monday, November 03, 2003 6:25 AM
To: [EMAIL PROTECTED]
Subject: Re:
On Mon, 2003-11-03 at 15:14, Eric Wieling wrote:
Licensing info for the G723.1 codec, direct from the holding company
that licenses the codec.
http://www.dspg.com/technology/LicensePricing.html
From what I remember when I looked into this about a year ago, this
isn't even the end of it,
On Sun, 2003-11-02 at 06:43, Scott Stingel wrote:
Can anyone please tell me their experiences with the Tyan i7501 series
(Xeon-basd), or recommend an alternate motherboard?
I'm using a TE410P card in a Tyan S2721 motherboard (a.k.a Thunder i7500
Pro). I've had no problems whatsoever with the
Hi Robert,
- Original Message -
From: rnc Info Lists [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, November 03, 2003 5:08 PM
Subject: Re: [Asterisk-Users] New IAX software phone (for WIndows platform)
Hi ,
I even think to avoid using an installer mainly because the
The makers of hardphones prolly get their G72x licensing by using a DSP
that already has a license. The DSP can't be that expensive. I wish
someone would make a PCI card with something like 8 of these chips on it
and sell it cheap. Should be pretty easy to build a codec for Asterisk
that uses
Hi,
Florian Overkamp a crit:
Ji,
-Original Message-
I have an asterisk box with one GS101 register to it in SIP
mode and an
IP10S in MGCP mode.
I can dial IP10S from my GS101 and everything seems fine.
But from my IP10S I can't dial any number (GS or anything
This is a really cool phone, except one problem, searched the archives
and this was brought up before. Just wondering if anyone figured out
how to solve it.
I'm having the same problem as these previous posts...
---
posted 06/09/03
Whenever I try using the voicemail through my ADSI
I have had this problem for a while where my fxs device has an answering machine on it
and getting a call will hang up for no reason.
I timed how long it took to hang up on me , 6 seconds , so I greped 6000 in zaptel src
and found some code about OHT which was not present before sept back from
Hi Thomas,
Unless you have a *very* specific need to use G.723.1 for compatibility
with someone else, forget it. It is pretty much an obsolete product.
Licencing is also a pain, as there is not patent pool for it. G.729 is
expensive to licence, but at least it is relatively strightforward. If
Lists wrote:
I am having a wired issues with the outgoing calls here is my queue file
Channel: IAX2/[EMAIL PROTECTED]/NUM
MaxRetries: 1
RetryTime: 600
WaitTime: 300
Context: playoutstart
Extension: s
Priority: 1
If someone picks up the phone, it works great, if it gets a voicemail, it
plays the
On Mon, 2003-11-03 at 15:47, Eric Wieling wrote:
The makers of hardphones prolly get their G72x licensing by using a DSP
that already has a license. The DSP can't be that expensive. I wish
someone would make a PCI card with something like 8 of these chips on it
and sell it cheap. Should be
I must agree with Eric on this one. I did testing with g723.1 pass thru
between two cisco ATA's and you can fit two calls in the same bandwidth as
one g729 call. But without a codec in * its pretty much pointless. Also
I have emailed these guys about the g723.1 lic they NEVER email back.
Thanks Steve,
there is no special reason for me for using g.723.
I will take g.729. It seems to be easier :-)
Regards,
Thomas.
-Ursprungliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Steve
Underwood
Gesendet: Montag, 3. November 2003 17:14
An: [EMAIL
Alastair Maw wrote:
On 03/11/03 00:25, Mark Spencer wrote:
As a side note, I strongly would like to see someone implement a
client using libiax2 which implements IAX2 instead of the (now
obsolescent) IAX version 1.
I'm implementing a Java-based IVR server (and yes, I know Asterisk does
IVR,
On Mon, 2003-11-03 at 15:02, nathan wrote:
Hi All,
I am attempting to setup Asterisk with sipcall.co.uk. They use Intertex
kit to provide the SIP service. Unfortunately Asterisk cannot seem to
authenticate against Intertex. Having provided SIP debug info the
provider has informed me that
You can port forward the 5060 SIP port and use externip keyword in
sip.conf to have it working behind a NAT.
Martin
On Mon, 3 Nov 2003, WipeOut wrote:
Robert Mann wrote:
Problem I have is this. outside firewall (extension 2003) can call me
inside firewall (extension 2000) and all is
Eric Wieling wrote:
The makers of hardphones prolly get their G72x licensing by using a DSP
that already has a license. The DSP can't be that expensive. I wish
someone would make a PCI card with something like 8 of these chips on it
and sell it cheap. Should be pretty easy to build a codec for
QuickNet certainly did this with their Windows PhoneJack LineJack, but
interestingly the Linux LineJack had the hardware DSP facility removed
IIRC - I'm guessing the 'open'-ness of Linux just frightened the legal
people :(
IIRC the DSP is still enabled in Linux; it's just the g.729a codec
Martin Pycko wrote:
You can port forward the 5060 SIP port and use externip keyword in
sip.conf to have it working behind a NAT.
Martin
Martin,
Is externip and new parameter??
Does it do a similar thing for the server as what nat=yes does for the
phone?
Later..
Maybe you need the straight through cable.
Martin
On Mon, 3 Nov 2003 [EMAIL PROTECTED] wrote:
Hi,
At least I have one E1 to test my E100P.
My telco company in Spain has installed one LiteSpan 1540 NT (UTR 2M)
I make a crossover cable between E100P and UTR.
1 - 4
2 - 5
after loading
And Visual Basic? Please.
What precisely is the problem with it? Or are you just a language nazi? I
don't like VB any more than you do but if the thing works, who cares what
it was written in. Nobody's asking me to maintain it.
Andrew
___
I would be happy to do so for the advantages of G723.1. i.e. great
sounding calls at a very low bandwidth. I suspect that the cost of
running the data over the PCI bus multiple times would be more than
offset by the faster compression/decompression provided by the DSP.
On Mon, 2003-11-03 at
Jeremy McNamara wrote:
If not registered, nothing works and the application closes by himself.
This is a very bad behavior. You only need to register if you plan on
receiving a call from Asterisk and your IP is dynamic or you need to
punch thru a NAT/Firewall edge device.
An error message would
On Monday 03 November 2003 06:17, Dan wrote:
P.S. I'll post later today a new prerelease (0.9.1) with some bug
fixes and some users requested improovements. Keep on eye on this
list!
As this has become quite popular and is taking up a significant number
of postings on this list, might I
I hope this doesn't show up twice (posted from wrong mail adr.)
Hi Eric,
You can actually get boards like this already from companies like
Mapletree.
Its a hardware pci carrier card where you add the number of DSP modules
that you need.
This hardware may be a bit 'high end' for most users on
Let's keep this positive. Somebody took the time to try to make
something useful. He's not charging for it.
If you don't like it, don't use it. If you have a problem with VB, port
it to C.
My pair of pennies,
David Gomillion
-Original Message-
From: [EMAIL PROTECTED]
It's new. It prevents asterisk from putting the private IP in the messages
that asterisk sends with SIP.
Martin
On Mon, 3 Nov 2003, WipeOut wrote:
Martin Pycko wrote:
You can port forward the 5060 SIP port and use externip keyword in
sip.conf to have it working behind a NAT.
Martin
Hi List,
Additional with the latest tries from the below
I get a nice random seg fault when I hangup on PSTN.
(With obviously no sound on x-lite, still!)
asterisk -gc
results after hanging up the pstn line in:
-- Executing Hangup(SIP/1087997-d79f, ) in new stack
== Spawn extension
Gentlemen
We are attempting to use * in a simple switching application:
+- office lines
|
V
LEC --PRI-- * --PRI-- modem bank (56k dialup modems)
The problem is that (even with no office lines active) the modems
have difficulty establishing a
Peer Oliver schmidt wrote:
And Visual Basic? Please.
What is wrong with Visual Basic?
I always thought, it is the solution that counts, not the programming
language. Am I missing something?
1) Bloat
2) Borgware
3) Try running it on Linux/*BSD
Not only is a really good win32 iax2 solution
Hi Jeremy,
- Original Message -
From: Jeremy McNamara [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, November 03, 2003 6:38 PM
Subject: Re: [Asterisk-Users] New IAX software phone (for WIndows platform)
Dan wrote:
If not registered, nothing works and the application closes by
David Gomillion wrote:
Let's keep this positive. Somebody took the time to try to make
something useful. He's not charging for it.
If you don't like it, don't use it. If you have a problem with VB, port
it to C.
I don't plan on using it. I will use mine, which is created in wxWindows
Hi,
- Original Message -
From: Peer Oliver schmidt [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, November 03, 2003 7:19 PM
Subject: Re: [Asterisk-Users] New IAX software phone (for WIndows platform)
Jeremy McNamara wrote:
If not registered, nothing works and the application
On 03/11/03 16:35, Jeremy McNamara wrote:
I'm implementing a Java-based IVR server (and yes, I know Asterisk does
IVR, and no, it's not flexible enough to do what I want and no, it
doesn't integrate well with the Java systems we have, etc. hence my
doing this).
Are you mad? What is not flexable
At 12:11 PM 11/3/2003, you wrote:
And Visual Basic? Please.
What precisely is the problem with it? Or are you just a language nazi? I
don't like VB any more than you do but if the thing works, who cares what
it was written in. Nobody's asking me to maintain it.
I was willing to give vb a
Hey all,
For those of you who are really worried about asterisk performance, I
thought I might alert you to a toy you might play around with. The Intel
Performance Primitives contain a number of optimized functions for use in
digital signal processing that could help with echo cancellation,
Hi. I am looking for the proper syntax for the Cut application. I am
working on a Feature Code extension that drops a caller directly into
a voicemail box. Here is what I have:
exten = _55.,1,Answer()
exten = _55.,2,Cut(VMEXT=EXTEN|55|2)
exten = _55.,3,Voicemail(u${VMEXT})
exten =
Thorsten Trapp wrote:
Hi List,
Additional with the latest tries from the below
I get a nice random seg fault when I hangup on PSTN.
(With obviously no sound on x-lite, still!)
asterisk -gc
results after hanging up the pstn line in:
-- Executing Hangup(SIP/1087997-d79f, ) in new stack
==
Hi!
I don't think that is what keeping the original poster's system from
working. The issue is one extension is configured for canreinvite=no
and the other is canreinvite=yes. One extension believes all RTP must
be passed through * while the other is attempting to negotiate a
phone-to-phone
Hi!
The voicemails servermail and fromstring variables should change
default
values when email voicemail notification gets received by user.
I change it, but received mail still shows Asterisk PBX in place of
fromstring.
Same here - please open a bug report on this.
Cheers, Philipp
I consider good examples to be those of John Todd and Zac Sprackett, viz:
http://www.loligo.com/asterisk/current/extensions.conf
http://sprackett.com/asterisk/conf/extensions.conf
If you lop the filename off each of those, you also get a directory of
*all* their .conf files, also good reading.
quote who=WipeOut
You are right in what you are saying.. I was thinking back to the
original message that started this thread that talked about load
balancing VoIP clients accross multiple servers.. Thats where my
comments came from.. :)
My goof, for not reading the start of the thread.
Hi!
How to use that externip new parameter?
Where in sip.conf and what is the format?
thanks
- Original Message -
From: Martin Pycko [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, November 03, 2003 3:34 PM
Subject: Re: [Asterisk-Users] Asterisk behind LinkSys NAT Routing
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