marrandy wrote:
On Friday 05 December 2003 09:47 pm, marrandy wrote:
In the meantime, I've pulled information that may, or may not be correct.
If
people can verify or add to this, it would be appreciated.
-
Codec
Here is the original article reference.
http://lists.digium.com/pipermail/asterisk-dev/2003-August/001435.html
In your sip.conf entry for the phone set
qualify=no
So far for how I use the phone I haven't really seen a downside to this setup,
but we are still really in a test mode at this point.
Hi,
- Original Message -
From: Gary [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, December 06, 2003 3:19 AM
Subject: [Asterisk-Users] USB - FXS for Windows
Yup there is the tigerjet unit...
has anyone found any others BUT which there might be a hope that
iaxclient
Hi,
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, December 05, 2003 8:22 AM
Subject: Re: [Asterisk-Users] x100p/hangup detection issues?
On Thu, 4 Dec 2003, Jonathan Tew wrote:
We're testing with an X100P card. When the caller on the POTS line
Message: 7
From: Juan J. Sierralta P. [EMAIL PROTECTED]
To: Asterisk Users [EMAIL PROTECTED]
Organization: Telefonica CTC Chile
Date: 06 Dec 2003 02:14:22 -0300
Subject: [Asterisk-Users] MGCP IADs
Reply-To: [EMAIL PROTECTED]
Hi,
For MGCP users. Is there any success stories with any MGCP
Hi all,
In order to get the CallerID from PSTN (X100P) I have modified callerid.c
file like that:
callerid.c [line 256]
from:
case 3: /* Number (for Zebble) */
to
/*case 3: Number (for Zebble) */
Without this modification my own number was displayed as the inoming call
CallerID.
Now I want to
On Thu, 4 Dec 2003, Andrew Gillham wrote:
Well as far as I can tell, the only version I have on the box is 2.4.22-1.
I certainly only have 'kernel-headers-2.4.22-1' installed and 'linux'
symlinked
to that directory in /usr/src.
Are you saying my /usr/include would be skewed? Since I
Is anyone other than me having trouble dialing out via IAXTEL? I havn't
changed my config files in weeks but seems that IAXTel calls (800 and FWD)
stopped working in the past week sometime.
Robert
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Hi all,
In order to get the CallerID from PSTN (X100P) I have modified callerid.c
file like that:
callerid.c [line 256]
from:
case 3: /* Number (for Zebble) */
to
/*case 3: Number (for Zebble) */
Without this modification my own number was displayed as the inoming call
CallerID.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Dan
Sent: Saturday, December 06, 2003 6:51 AM
To: Asterisk Users
Subject: [Asterisk-Users] CallWaiting CallerID
Hi all,
In order to get the CallerID from PSTN (X100P) I have
Hi Dan,
[..]
to help, so... if another producer who read this list is interested that his
own product to be fully supportde by DIAX, then I kindly ask him to send me
a mail.
I am very interested in this support within DIAX. Please keep us posted
as to any kind of contact you might receive from a
Is anyone other than me having trouble dialing out via IAXTEL? I havn't
changed my config files in weeks but seems that IAXTel calls (800 and FWD)
stopped working in the past week sometime.
Yes, I've been having problems as well but had not taken the time to diagnose
the problem. Just did
Todd Wallace wrote:
Anyone ever have the Ethernet port on a Budgetone phone quit working. For
some reason, it stopped link'ing up and I can't get an address from DHCP or
when I set a static address, it would ping. I have reset to factory
defaults and nothing seems to work. Feels like the port
Hi ,
I picked up a x100p the other day and thaught I'd havea go at getting the
driver going for linux 2.6, things have gone pretty, two basic problems.
1. makefiles, with 2.6 you can't get away with using the old makefile to build
the kernel modules, they will build but you'll get an error
Hi Richard,
- Original Message -
From: Richard Alexander [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, December 06, 2003 3:31 PM
Subject: RE: [Asterisk-Users] CallWaiting CallerID
.
Do you have:
callwaiting=yes
callwaitingcallerid = yes
In Zapata.conf ?
Now I
I am new to * and after getting some much needed help from someone on the
IRC channel with my config files I got my system up and going.
I have several dlink DG 104s's and they seem to work fine on my LAN the only
trouble I have is trying to get them to work thru double Nat by that I mean
on both
Hi,
- Original Message -
From: Peer Oliver schmidt [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, December 06, 2003 3:35 PM
Subject: Re: [Asterisk-Users] USB - FXS for Windows
Hi Dan,
[..]
to help, so... if another producer who read this list is interested that
his
anyone with some succes with the drivers for the TE410p ?
joachim.
At 14:06 6/12/2003 +, you wrote:
Hi ,
I picked up a x100p the other day and thaught I'd havea go at getting the
driver going for linux 2.6, things have gone pretty, two basic problems.
1. makefiles, with 2.6 you can't
Yes, I've been having problems as well but had not taken the time to
diagnose
the problem. Just did some looking and it appears iaxtel.com has removed
the iax v1 support. iax2 seems to be working fine.
Rich,
That solved the outbound problem.. Thanks for the hint... 800 numbers are
accessable
To add to this problem, we have been fighting a denial of service
attack on the toll-free gateways. We have installed new code to deal
with this problem hopefully. I will never understand why someone would
attack a free service.
John Lodden
Telesthetic
On Sat, 2003-12-06 at 07:45, rnc Info
On Fri, Dec 05, 2003 at 04:40:42PM +, Michael T Farnworth wrote:
On Fri, 5 Dec 2003, Nicolas Bougues wrote:
On a slightly different topic : does somebody know of a NAT-friendly
(as Grandstream means it) tftpd server ? It seems theirs replies from
port 69, which is the only thing their
On Friday 05 December 2003 15:51, Ray Burkholder wrote:
On Friday 05 December 2003 14:44, mattf wrote:
Has anyone out there had the freezing problem(where they have to
kill asterisk with kill -9) on any linux distro other than
RedHat?
What other distros do people out there use
John,
Did you shut down iax v1 (udp/5036) on purpose, or are we all suppose to
change to iax2 (udp/4569) forever?
Rich
To add to this problem, we have been fighting a denial of service
attack on the toll-free gateways. We have installed new code to deal
with this
Hi,
I have a RH9 system with an onboard VIA sound chip. According to the
archives, VIA won't work for asterisk.
So, I disabled the VIA and I purchased a Creative Labs Soundblaster PCI
128-Voice soundcard ($13). This card is on the approved RedHat list.
However, the documentation inside the
Rich Adamson wrote:
On Sat, 2003-12-06 at 07:45, rnc Info Lists wrote:
Is anyone other than me having trouble dialing out via IAXTEL? I havn't
changed my config files in weeks but seems that IAXTel calls (800 and FWD)
stopped working in the past week sometime.
To add to this problem, we
Have you tried just the DG104s behind nat?
Have you had any issues with call waiting?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of James Schenck
Sent: Saturday, December 06, 2003 9:26 AM
To: [EMAIL PROTECTED]
Subject: RE:
I thought Mark was willing to help you with the S100U, what ever
happened with that?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Dan
Sent: Saturday, December 06, 2003 3:49 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users]
Hello,
I have a friend that is asking if he can use his Ericsson 3413
H.323 IP phone with Asterisk. I can't seem to find any reference to this
phone on the Wiki...
--
Vice President of N2Net, a New Age Consulting Service, Inc. Company
http://www.n2net.net Where everything
Great interview with Nicolas-Peter Pohland CEO of SNOM
Chief Executive Officer
- Original Message -
From: Steve Totaro [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, December 06, 2003 1:01 PM
Subject: FW: snom in Wallstreet report
-Original Message-
From: Robert
Hello.
I did an update a couple of days ago. Between Wednesday Friday.
I have just started getting problems. The first time since I installed and
played with asterisk in August. I don't know why it is indicating three way
call.
When I stopped and restarted asterisk, the problem went away.
Hi all,
Does anybody know a way to set callerid to a channel
by searching the incoming cid to a database and return
the name of the caller and set the callerid to the
channel.
I tried it with agi script in C but the problem is
that agi command SET CALLERID did not work for me with
a variable.
What
On Sat, 2003-12-06 at 12:36, Olle E. Johansson wrote:
I seem to get a G729 codec on calling 1800 numbers on IAXTEL...
Temporary fault or do we have to buy a license?
/O
The toll-free gateways allows the use of G729, if you don't have that
CODEC don't show you can use it. The gateways can
On Saturday 06 December 2003 19:01, Greg Boehnlein wrote:
Hello,
I have a friend that is asking if he can use his Ericsson 3413
H.323 IP phone with Asterisk. I can't seem to find any reference to this
phone on the Wiki...
you can either use chan_h323 or chan_oh323 (the latter is contrib
I can setup one 104s with a Nat router between it and *
dg104slan---natlan-asterisk-lan--dg014s
That works great
But if I try
dg104s-lan---nat---internetnat--internet--asterisk--
---dg104s
a 1 2
b
I cannot connect to IAX1 either
What I would like to know is icmp needed for IAX1?
If it is that is the cause of my trouble as my service provider is blocking
icmp and port 135
Do to welchia and blaster
But IAX2 works fine
James Schenck
Egraph Design Inc.
Arkansas Online Internet Services
Thnx, now its working perfectly ...
Azhernathan [EMAIL PROTECTED] wrote:
On Friday 05 December 2003 13:45, Azher Amin wrote: Hi, I have a setup using Xlite SIP phone and X100P card and I am using the Dec 1 CVS. I have put the incomming limit in both zapata.conf for the FXO card and in the
Tristan 'Minty' Colgate wrote:
I'd like to thank everyone on #asterisk for all the support they gave to a
fellow linux enthusiast... absoutely none.
That was really a nice post until right at the end here.
I hope you understand that cheap shots like this just make *you* look
like an
Roy Sigurd Karlsbakk wrote:
I know people are running h.323 in production (or so I've heard), but as (AFAIK) there still are some
unsolved issues, YMMW.
Why not list out the specific problems and they can be addressed if they
are still a problem? You are quick to bash my channel driver but
I am trying to setup a Sipura SPA-2000 with asterisk and do not hear any
audio.
I have set NAT=yes in sip.conf, is there anything else to look at?
hilpert*CLI sip show channels
Peer User/ANRCall ID Seq (Tx/Rx) Lag Jitter
Format
192.168.0.6 000E08AAFB fb865a36-f7
Hi,
- Original Message -
From: Jim Paraschou [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, December 06, 2003 8:13 PM
Subject: [Asterisk-Users] Caller ID from Database
Hi all,
Does anybody know a way to set callerid to a channel
by searching the incoming cid to a database
Hi,
Anyone knows if the S100U USB interface can be used on the same box as *
server?
It can pass the CallerID to the connected phone?
Best regards,
Dan
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
On Sat, 2003-12-06 at 14:42, Dan wrote:
Hi,
Anyone knows if the S100U USB interface can be used on the same box as *
server?
Currently I think this is the only option.
It can pass the CallerID to the connected phone?
At least if the phone adheres to the same callerid spec as is used in
-- Executing unixODBCput(SIP/10-cc1b, BLAH/blah=bkw) in new stack
-- unixodbcput: family=BLAH, key=blah, value=bkw
-- Executing unixODBCput(SIP/10-cc1b, BLAH/blah=bk2) in new stack
-- unixodbcput: family=BLAH, key=blah, value=bk2
-- Executing unixODBCget(SIP/10-cc1b, testingget=BLAH/blah) in new
- Original Message -
From: Steven Critchfield [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, December 06, 2003 11:39 PM
Subject: Re: [Asterisk-Users] S100U and *
On Sat, 2003-12-06 at 14:42, Dan wrote:
Hi,
Anyone knows if the S100U USB interface can be used on the same
Hi!
-- Executing unixODBCput(SIP/10-cc1b, BLAH/blah=bkw) in new stack
-- unixodbcput: family=BLAH, key=blah, value=bkw
Just to get everyone hot and bothered! :)
Cool!! Excellent, incredible! And I thought you were angry at me
suggesting this... ;-
Cheers, Philipp
The FWD bridge is currently down, as I am looking to try to find someone
else to host the IAX2 to SIP gateway so that iaxtel can remain strictly
IAX2.
Mark
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Accually i'm going to rename everythign from unixodbc to just odbc
bkw
On Sat, 6 Dec 2003, Philipp von Klitzing wrote:
Hi!
-- Executing unixODBCput(SIP/10-cc1b, BLAH/blah=bkw) in new stack
-- unixodbcput: family=BLAH, key=blah, value=bkw
Just to get everyone hot and bothered! :)
Hi,
My apologies for those on the channel who may take offense to this, atleast
the ones to whom it was not aimed, but the fact is that after making a simple
enquiry on the IRC channel I was in absolute shock...
I asked one simple question, are there any known issues with the zaptel
modules
Hi Nicolas,
Thanks for the file.
I would appear to have some of the file missing that the BT-100 is looking
for.
Ala,cfg.txt
sipp.bin
ring.bin
After the tftp update the program is still showing 1.0.3.81.
Any thoughts.
Regards
Dave
-Original Message-
From: [EMAIL
On Sat, Dec 06, 2003 at 04:35:55PM -, David J Carter wrote:
Hi Nicolas,
Thanks for the file.
I would appear to have some of the file missing that the BT-100 is looking
for.
Ala, cfg.txt
sipp.bin
ring.bin
After the tftp update the program is still showing 1.0.3.81.
Thats odd. We have the firmware on our Linux RH7.3 tftp server. The GS
phones can download it just fine on the LAN.
We would like that NAT-Friendly tftp though:)
- Original Message -
From: Nicolas Bougues [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, December 06, 2003 6:36
Mark,
I cannot connect to IAX1 either
We are phasing out support for IAX1. Are the people who still *need*
IAX1?
Given that iax2 has been around and stable for a rather lengthy period,
I would doubt there is any real need for it other then to coordinate production
systems that can't be
I have just started laying out the plans for my first project using
Asterisk. I am very interested at this stage in getting much needed
feedback, critiquing my approach. What are the ups and downs going to
be if I develop this project as follows:
-The client wants to connect some phone reps
Hi!
using NetMeeting, or GnomeMeeting, or some other H323, or SIP
compatible client
For H323 you'll need to install a gatekeeper next to Asterisk and fiddle
with h323 or oh323 (I love to live dangerously, hit me Jeremy). :-
Moreover NetMeeting doesn't work through NAT.
SIP is easier to set
I've used the tftp-hpa 0.34 with this as the startup and it has worked fine
in the past with GS 101 and 102s:
/usr/local/sbin/in.tftpd --daemon --port=69 --verbose=6 --user unpriv.unpriv
/usr/local/tftp
MATT---
-Original Message-
From: TeleSIP [mailto:[EMAIL PROTECTED]
Sent: Saturday,
On Saturday 06 December 2003 09:10, Michael Welter wrote:
Hi,
[...]
says Ensoniq, ES1371 (AudioPCI-97), module es1371, which seems normal.
However, there is no sound.
Does anyone have experience with this?
Thanks,
Mike
___
Asterisk-Users
If setting up multiple users in the same location overseas, I would
strongly suggest setting up another asterisk box at that location. IAX
can transverse nat and you will have the change to use some low
bandwidth codecs that will not cost you anything more, as I have found
that most devices do not
Sorry, but would someone mind giving a brief explanation to newbies as
to why this is cool? I am interested in creating call trees from a
postgres database, so this looks like it might be useful, but I still
don't understand much of what's going on here.
Thanks,
Carl Youngblood
On Dec 6,
Well think about the ability to share that data among a cluster of
asterisk servers. Or interface to that data from the web. Or even GASP
lookup extensions with it.
bkw
On Sat, 6 Dec 2003, Carl Youngblood wrote:
Sorry, but would someone mind giving a brief explanation to newbies as
to why
What kind of stability / reliability are people currently experiencing
with the Linux / Asterisk combination? We will be running 3-10 SIP
phones from India to US using nothing more than regular cable / dsl
connections from both locations.
Also, what make / model SIP phone do you recommended
Would have probably been more appropriate to at least announce that
iax was going to disappear at some specific date, as opposed to folks
randomly discoverying it and chasing problems. (Kind of related to why
there isn't a marketing plan.)
Sorry, it was something of a side effect of some
na, rather 3 days in London with stay in Mayfair :)))
On Tuesday 02 of December 2003 16:18, Scott Stingel wrote:
Must have included a week in Amsterdam
Scott M. Stingel
Emerging Voice Technology Inc.
URL:www.evtmedia.com
-Original Message-
From: [EMAIL PROTECTED]
On Sat, Dec 06, 2003 at 11:14:40PM +, Tristan 'Minty' Colgate wrote:
Hi,
My apologies for those on the channel who may take offense to this, atleast
the ones to whom it was not aimed, but the fact is that after making a simple
enquiry on the IRC channel I was in absolute shock...
...
On Sat, 6 Dec 2003, Jeremy McNamara wrote:
Roy Sigurd Karlsbakk wrote:
I know people are running h.323 in production (or so I've heard), but as (AFAIK)
there still are some
unsolved issues, YMMW.
Why not list out the specific problems and they can be addressed if they
are
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