On Tue, Sep 09, 2003 at 07:57:20PM -0400, Jeremy McNamara wrote:
Travis Johnson wrote:
I've called NuFone and was not impressed by their voicemail answering
system (choppy) and was unable to even leave a message before the phone
call was disconnected (in the middle of the
recording).
Hi
I really need caller id to work in the UK, I understand that the X100p
uses a US chipset,two questions
1) is that a product that converts UK to US caller id in line
Not really the answer you were looking for, but if you get a line from a
non-BT supplier (e.g. NTL or Telewest) you are quite
- Original Message -
From: Linus Surguy [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, September 10, 2003 9:46 AM
Subject: Re: [Asterisk-Users] UK Caller ID and X100p
Hi
I really need caller id to work in the UK, I understand that the X100p
uses a US chipset,two
If T38 could be ported to IAX/IAX2 that would be great
Is it correct that currently fax does not work over IAX/IAX2?
David Luyens
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lee Goodman
Sent: Tuesday, September 09, 2003 8:50 PM
To: [EMAIL
hi,
I hav a softPBX setup. Our set up has 2 servers,
one is connected to an ISDN PRI E1 coming from PSTN central office and the other
server is connected to another E1 which is coming from a Nortel PBX. and 2
servers are connected to a LAN. So when a Nortel PBX users want to get an out
Louis-David Mitterrand wrote:
So your going to judge our system by making one phone call into my home
asterisk system that runs on a fully saturated ADSL connection.
Wait... of course people are going to judge you by that! If putting your
company's answering machine on your (saturated) dsl
Hi,
Where can I disable silence detection with
chan_capi ?
Is there option in capi.conf ?
Regards
Rattana
At 23:53 9-9-2003 +0200, you wrote:
On Tuesday 09 September 2003 17:54, Tais M. Hansen wrote:
How do I optionally hide the caller id on outgoing calls on chan_zap? Ie.
calling h323 - asterisk - chan_zap - isdn provider.
Problem solved. I made app_dial.c take an option to change
OK
When I call Netmeeting by my phone. I have silence (the sound is choppy) in
my phone but not with netmeeting.
And I don't know why ?
How can I set it ?
If chan_capi not support silence detection perhaps asterisk do it ...
any tips ...
Thanks
Rattana
- Original Message -
From:
At 23:53 9-9-2003 +0200, you wrote:
On Tuesday 09 September 2003 17:54, Tais M. Hansen wrote:
How do I optionally hide the caller id on outgoing calls on chan_zap? Ie.
calling h323 - asterisk - chan_zap - isdn provider.
Problem solved. I made app_dial.c take an option to change
do you see DATA_B3_REQ errors on the * console? if yes, then
netmeeting has a bad timing (or no timing?) for sending audio
or your network produces that jitter. chan_capi needs to get
the outgoing audio in a rather strict timing (there is no
possibility to use a jitterbuffer on a Bchannel! 64
I think I have found the problem .
The guilty is Netmeeting !! Netmeeting make silence detection.
Whatever I put silence detection into Min I have this.
I make some test with openphone and I have good result.
Rattana
- Original Message -
From: Klaus-Peter Junghanns [EMAIL PROTECTED]
To:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Wednesday 10 September 2003 10:47, Florian Overkamp wrote:
How do I optionally hide the caller id on outgoing calls on chan_zap?
Ie. calling h323 - asterisk - chan_zap - isdn provider.
Problem solved. I made app_dial.c take an option to
Hi,
I have an installation connecting two machines through IAX2. Each
machine has 3 FXS and 4 FXO ports.
Everything seems to work fine, except on one FXO port, where I
constantly get a strange locomotive noise when I use it to terminate
an IAX2 incomming call. Usually after a while the strange
I assume it manages the signal part of the RTP stream but not the RTP voice
stream at the codec level?
Maybe someone else can comment on the translation methodologies within
Asterisk?
Regards,
Steven Thomas
Luckily, I have a E100P.
could you tell me how to get the dial status within the extension logic or
in AGI script?
- Original Message -
From: Eric Wieling [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, September 10, 2003 1:39 PM
Subject: Re: [Asterisk-Users] Dial + disconnect
Hello,
There are 2 leds at the back of the E100P card. I
search the mailing list and digium website. There seems to beno
documentation about them.
On the card itself, 1 led is labeled D1 and the
other is labled D2.
Can someone explain or point me to the right
resources about these
George Pajari wrote:
FYI I asked them:
Your website talks about configuring the Rhino channel bank as 24xFXS.
Is it possible to mix FXO and FXS modules? What affect does that
have on pricing?
They replied:
We will have FXO and the ability to mix both FXS FXO within 60-90
days. Our RD
On Tue, 9 Sep 2003 09:15:32 +0100, Skuse, Phil wrote
What's the secret to getting sound through Xlite? The SIP
messages all look OK to me, but the sound isn't coming through.
It was trying to use GSM, so I searched the archive and tried:
disallow=gsm
allow=ulaw
Now it says that it's
On Tue, Sep 09, 2003 at 06:13:02PM -0500, denon wrote:
With regards to Asterisk on FBSD, I for one would love to see it happen. I
prefer FreeBSD over Linux in almost every case.
However, personally I have a few concerns:
Namely, the primary developer is a Linux nut .. (sorry Mark, I mean
messages all look OK to me, but the sound isn't coming through.
It was trying to use GSM, so I searched the archive and tried:
disallow=gsm
allow=ulaw
Now it says that it's using ULAW but I still get no sound in
either direction.
Have you tried setting Send Internal IP to Yes?
On Wed, Sep 10, 2003 at 11:02:59AM +0100, Alastair Maw wrote:
As importantly, we'll get cross-unix portability for less effort. There
won't be a *BSD port - the autoconf stuff will sort it all out for you.
Of course, we'll have to do some work to make sure it all functions
properly, but it
Hi,
I don't believe this is the forum to discuss how much you have to pay for
T1 lines. We currently pay $900 per month per PRI line, and we only got
that discounted rate because we have 31 total PRI lines.
How far away from your main NOC are you located? Maybe you should look at a
wireless
You are correct, the SIP image doesn't support push, like he SCCP image
does.
Lee Goodman
- Original Message -
From: Jared Smith [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, September 09, 2003 6:54 PM
Subject: Re: [Asterisk-Users] Pushing data to a 7960
On Tue, 2003-09-09 at
I have ADSI working on my Aastra (Vista/Nortel) 350 phone and everything is
working fine.
However, I want the asterisk.adsi to load into the 'self-load' slot but
can't figure out what the correct FDN for doing this is. Does anyone know
the right FDN for the SL slot on these phones?
Also, does
Hello, I posted a question a few days ago and as part of a discussion, someone
mentioned asking the list for consulting help...so I am. I originally type something
out and sent it to the admin portion of this list by mistake. This is more hopefully
correct attempt.
If you are interested in
One is a bicolor LED which indicates:
off - span not configured / driver not loaded
green - OK
red (flashing) - RED Alarm
yellow - Yellow Alarm
The second is an orange LED which indicates a loopback (local or remote)
is up for testing purposes.
Mark
On Wed, 10 Sep 2003, Chee Foong wrote:
On the grandstreams if I recall the docs are incorrect on how the transfer
feature works. Transfer + EXT + Transfer
bkw
On Tue, 9 Sep 2003, Hielke Christian Braun wrote:
Hello,
hope somebody can help. I have setup a queue which maps to some
Budgetone SIP phones. When a call is answered,
Dude,
NuFone so totally ROCKS... I have yet to have any issues. The
last issue I had wasn't even related to NuFone.. but this stupid Nachi
worm nailing our routers and causing packets to be dropped. Other than
that the call quality is excellent. Customers can't tell the diffrence.
bkw
On Wed, 10 Sep 2003, Jim Mercer wrote:
i've not done an autoconf before, and i suspect it will require not a small
amount of tweaking.
i suspect the BSD patches will head in the direction of a number of those
tweaks.
Some of the patches that get applied in the normal /usr/ports tree under
Hi,
Is it possible to use asterisk with Free World Dialup (FWD) ?
Did someone manage to make it work? how?
Best,
-P
--
__
Sign-up for your own personalized E-mail at Mail.com
http://www.mail.com/?sr=signup
CareerBuilder.com has over
Yes I did.
To see, dial #23195 via FWD. You will be greeted with the test asterisk
message.
I used samples from John Todd's web site.
I have currently setup to outbound calls, not inbound yet.
Awesome!
http://www.loligo.com/asterisk/example-configs.2003-04-24/
Frank...
-Original
Hi there,
I´ve been out for some months now, haven´t been checking the list at all.
Does anyone know if the problem with the Quicknet Linejack (FXO) card dial out to PSTN
with asterisk was solved?
Is anybody working on it?
Cheers,
-Z
--
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Zara Trousk
Sent: Wednesday, September 10, 2003 10:40 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Free World Dialup (FWD).
Hi,
Is it
Hi
We are trying to get better sound quality out of the prompts on our Asterisk
system. We had some new ones made by thevoice.digium.com and they are in WAV
format instead of the default GSM format on the Asterisk server. The problem
is, when you dial in to the server using the FXO card (X100P)
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I've tested the S100U unit in two different computers both with
different USB subsystems. The one it worked the best in used the
usb-uhci USB driver. The other system uses the usb-ohci USB driver. To
make sure it was not the installation, I used the
Eric Wieling wrote:
That would be reinvite= and canreinvite= in the user entry for each SIP
endpoint. Asterisk will allow the endpoints to talk directly to each
other if both those settings are = yes (the default, I think) AND both
endpoints use the same protocol (SIP) AND the same codec.
I
Hello All,
I am a newbie looking to learn about Asterisk. I'm new to IVR and all
that goes with it. I would like to know if it is possible to grab the
number of an incoming call, have Asterisk, or third party software
return the call with an automated voice message allowing the original
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Jason A. Pattie wrote:
| It's even worse on the usb-ohci machine. Most of the time, the S100U
| doesn't even give me dialtone.
Sorry, forgot to mention that the S100U device is detected and setup
correctly on this machine and no error messages are
On Wed, 2003-09-10 at 10:42, Olle E. Johansson wrote:
Eric Wieling wrote:
That would be reinvite= and canreinvite= in the user entry for each SIP
endpoint. Asterisk will allow the endpoints to talk directly to each
other if both those settings are = yes (the default, I think) AND both
S100U's are very flaky.. I am on my third one...
The first one lasted about a month and then started clicking and freaking which got
worse and worse to the point where there was just noise.. and the noise was all
comming from the S100U..
The second one started by dropping calls all the time
On Wed, 2003-09-10 at 10:51, Olle E. Johansson wrote:
Lubomir Christov wrote:
today I found this security report regarding Asterisk SIP Security.
http://www.securiteam.com/securitynews/5LP0720B5G.html
Important information. Why a silent patch and no information to the mailing list?
Hey guys,
I have looked around the archives and any other documentation I could
find, and cant figure out how or if * will use the ringmaster service. I
did see some info about ringing cadence and distinctive ring, but I it
looked to me like it was regarding * generating different ring cadence,
On Wednesday 10 September 2003 07:53 am, Jim Mercer wrote:
any excentricities the SCO-linux people add to the code can usually
be ifdef'd back to normality. 8^)
Try the one in acl.c, referencing /proc/net/route. To do the
analogous on FreeBSD, you have to parse kernel internal structures.
If somebody's interested...
Cisco confirmed that current SIP images up to 5.3 cannot
register any lines other than line 1 with backup proxy.
I've submitted a feature request.
Michael.
On Friday 05 September 2003 10:35 am, Michael Ulitskiy wrote:
Well, on the other hand Release Notes for
well depending on the hardware you are using and where you are
using it at, in some cases there is. look in zapata.conf search
'callprogress'.
Chee Foong wrote:
Yes you are right, Sorry my mistake.
So, is there a way to detect busy, answer, or no answer call?
Foong
- Original
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Hash: SHA1
Thanks WipeOut. I agree. I really hoped that wasn't the case, though.
WipeOut . wrote:
| S100U's are very flaky.. I am on my third one...
|
| The first one lasted about a month and then started clicking and
freaking which got worse and worse to the
On Wednesday 10 September 2003 05:02 am, Alastair Maw wrote:
denon wrote:
With regards to Asterisk on FBSD, I for one would love to see it
happen. I prefer FreeBSD over Linux in almost every case.
However, personally I have a few concerns:
Namely, the primary developer is a Linux nut
In /etc/asterisk/zapata.conf:
busydetect=no
callprogress=no
On Wed, 2003-09-10 at 02:44, Surajee Ratnayake wrote:
hi,
I hav a softPBX setup. Our set up has 2 servers, one is connected to
an ISDN PRI E1 coming from PSTN central office and the other server is
connected to another E1 which
Thanks for the info. Pretty much what I figured. I would be VERY interested
in hearing the prompts over a Digital T1 line instead of my analog port. If
you could make a phone number available, I would appreciate it very much
Thanks
Lee Goodman
[EMAIL PROTECTED]
- Original Message -
Steven Critchfield wrote:
I've added a security page to the Wiki:
http://www.voip-info.org/tiki-index.php?page=Asterisk+security
Maybe there should also be a link for best practices with respect to
dial plan layout.
I guess since this is my second comment on the wiki, I should log in and
On Wednesday 10 September 2003 10:51 am, Olle E. Johansson wrote:
Lubomir Christov wrote:
today I found this security report regarding Asterisk SIP
Security.
http://www.securiteam.com/securitynews/5LP0720B5G.html
Important information. Why a silent patch and no information to
the
I agree with you about warning that some SIP phones, especially the Cisco
phones, do not handle this well.
Also, should can reinvite=yes in the example be canreinvite=yes without
the space or will it work either way?
It might also be useful to note in the documentation that these settings
make
Thank you guys a million!
I´ll try this weekend and I let you know.
Any suggestions on other free VoIP providers like FWD and IAXTEL ?
Cheers,
-Z
- Original Message -
From: Leif Madsen [EMAIL PROTECTED]
Date: Wed, 10 Sep 2003 11:15:11 -0400
To: [EMAIL PROTECTED]
Subject: RE:
Hello Everyone,
About a month ago, someone put a question to the list about which ADSI
spec to purchase from Telcordia. I looked in the archives, and it
appears that this question was never answered, so I'll put it to the
list in a slightly different manner: Do I need to purchase the
Telcordia
hi,
This question is about DBPut and DBGet,
Can i put about 1000 keys in a single family, (only once for the lifetime)
for ex.
exten = _X.,5,DBput(family/key1=${val})
...
exten = _X.,5,DBput(family/key1000=${val})
like above and if i later retrieve it, randomely, with inbound calls,
Thank you guys a million!
I´ll try this weekend and I let you know.
Any suggestions on other free VoIP providers like FWD and IAXTEL ?
Dont forget those are solutions that assume the * box has NO NAT in front
OR
a statefull / smart NAT , some magic redirection rules
-BEGIN PGP SIGNED
The demo 1235 extension that Asterisk ships with works fine and the
messages are sent to the address I set in voicemail.conf. I guess that
means that my configuration is working perfectly so far.
But when I set up another extension with a voicemailbox, no mail is sent
when a message is left,
Is there any interest in starting an ADSI list somewhere so people can help
each other out? I'm trying to get started with ADSI programming as well,
and can't seem to find any ADSI information anywhere.
If anyone is interested in starting an ADSI discussion list, contact me off
list ([EMAIL
On Wednesday 10 September 2003 12:12 pm, Jean-Marc V. Liotier wrote:
The demo 1235 extension that Asterisk ships with works fine and the
messages are sent to the address I set in voicemail.conf. I guess
that means that my configuration is working perfectly so far.
But when I set up another
I found the Black Dolphin web site and downloaded their (windows) ADSI
script IDE ($499, free but crippled demo), but found that the files that
it generates are binaries that will only work with another piece of
software that they sell for $299.
Did you try to ast_stream file on those audio file
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Jean-Marc V. Liotier
Sent: Wednesday, September 10, 2003 1:13 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Voicemail notification email with
no attachment despite attach=yes
The demo
I'd definitely be interested in talking and learning more about ADSI, not
sure if I'd prefer a mailing list or some kind of bulletin board/forum type
thing? I can set up a forum pretty easily and don't mind hosting it - anyone
else got any preferences?
I'm assuming that both circuits to the * box are E1/PRI, so those settings
wouldn't make a difference.
To the OP, you may want to run pri (intense) debug on the spans to see
what's going on.
If you are running RBS to the Nortel box, then the busydetect and
callprogress may be the ticket. You
in voicemail.conf :
1234 = 4242,Test mailbox,[EMAIL PROTECTED]
6004 = 4242;Other test mailbox,[EMAIL PROTECTED]
How about the second line having a semi-colon between password and
description, while the first line has a comma?
___
Asterisk-Users
If like me you run * on a VPN (or multihomed) gateway and want to serve
remote SIP clients, make sure you have
bindaddr = 192.168.0.1 ; or whatever is your box's private IP
otherwise * might bind to its public IP and send it as return address in
the SIP call setup, which will (should) be
On Wed, 10 Sep 2003, Alastair Maw wrote:
I keep meaning to sort out autoconf/automake stuff for Asterisk. I
notice that Asterisk is GPLed, so there won't be any licensing issues.
I'm quite surprised no one else has got round to it before.
Anyway, I have no time at the moment for this, but
We are trying to implement area-code dialing in our asterisk PBX.
Basically: we will have a number of customers, who may be in different area
codes, that want to direct-dial each other's extensions. We want this to
work like a real centrex, in that seven-digit numbers should try (1)
local VoIP
You have to actually purchase the software in order to save the files
you create. They have an example file that comes with the package, so
I'll play around with it tonight.
-Original Message-
From: TC [mailto:[EMAIL PROTECTED]
Sent: Wednesday, September 10, 2003 12:30 PM
To: [EMAIL
Tilghman Lesher wrote:
On Wednesday 10 September 2003 10:51 am, Olle E. Johansson wrote:
Lubomir Christov wrote:
today I found this security report regarding Asterisk SIP
Security.
http://www.securiteam.com/securitynews/5LP0720B5G.html
Important information. Why a silent patch and no
It should work but you need to do Goto(extensions,666${EXTEN},1)
Martin
On Wed, 10 Sep 2003, Ernest W. Lessenger wrote:
We are trying to implement area-code dialing in our asterisk PBX.
Basically: we will have a number of customers, who may be in different area
codes, that want to
On Wednesday 10 September 2003 01:04 pm, Olle E. Johansson wrote:
Tilghman Lesher wrote:
On Wednesday 10 September 2003 10:51 am, Olle E. Johansson wrote:
Lubomir Christov wrote:
today I found this security report regarding Asterisk SIP
Security.
I took the liberty of adding Leif's FWD Asterisk configuration to the WIKI, so for an
- yet incomplete- overview on how to connect Asterisk and FWD please go to
http://tinyurl.com/mwe0
And, please add info or mail me configurations that works for connecting to FWD with
the asterisk server
A new RFC was published today, RFC 3601:
Abstract:
This memo describes the full set of notations needed to represent a
text string in a Dial Sequence. A Dial Sequence is normally composed
of Dual Tone Multi Frequency (DTMF) elements, plus separators and
additional actions (such as wait
On Wed, 2003-09-10 at 11:55, James Sharp wrote:
If I have a system with 1 machine to handle incoming H.323 calls and then
multiple machines to distribute them to T1 ports over TDMoE, where does
the codec translation take place? Does it take place in the master system
or does it take place in
bottom response = on
On Tue, 2003-09-09 at 12:41, Robert Boardman wrote:
I have an ISDN TA that has 2 POTS interfases (FXS), can these be used
with asterisk?
Thanks in advance
Robb
Yes, I have two such installations. Be advised there are some gotchas.
My TAs are older
Since no one yet has objected to this proposal, could you all move on
from discussing if it should happen, and toward doing something. In the
long run it shouldn't be that big of a deal to make the few changes
necessary. Also if someone jumps in and gets the autoconf started, I'm
sure Mark will be
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
WipeOut . wrote:
| S100U's are very flaky.. I am on my third one...
|
| The first one lasted about a month and then started clicking and
freaking which got worse and worse to the point where there was just
noise.. and the noise was all comming from the
Also it wasn't a proven exploit. They said it could allow an attacker to
obtain remote and unauthenticated access. And if pigs could fly I
would be a rich man!
bkw
Read the security vulnerability. It referenced CVS as of a certain
date. If you aren't keeping up with CVS changes, why are
On Wed, 2003-09-10 at 13:16, Tilghman Lesher wrote:
On Wednesday 10 September 2003 01:04 pm, Olle E. Johansson wrote:
Tilghman Lesher wrote:
On Wednesday 10 September 2003 10:51 am, Olle E. Johansson wrote:
Lubomir Christov wrote:
today I found this security report regarding Asterisk SIP
On Wed, 2003-09-10 at 11:55, James Sharp wrote:
If I have a system with 1 machine to handle incoming H.323 calls and
then
multiple machines to distribute them to T1 ports over TDMoE, where does
the codec translation take place? Does it take place in the master
system
or does it take place
At 11:37 -0500 10/9/03, Tilghman Lesher wrote:
Probably because Mark doesn't have time to realize that somebody
is going to publish a temporary vulnerability that he fixes in 5
minutes. When someone points out a bug in my own programs, I'll
go fix it, but I don't usually then publish a
Could the bindaddr=x.x.x.x be a way to make * work through a NAT?
I have * and a few 7960 phones behind a NAT. I am trying to register with a
proxy on the outside of the NAT. Registration is ok, but the VIA field has
my inside NAT ip address (192. 168.0.7). So the proxy doesn't know how to
send a
On Wed, 10 Sep 2003, Fearghas McKay wrote:
It has certainly caused some fervent checking amongst users I know, and
since the last release was some months ago if the vulnerability was present
then there will be users who have had to move from taking a stable build to
building from CVS, which
At 13:16 -0500 10/9/03, Tilghman Lesher wrote:
Read the security vulnerability. It referenced CVS as of a certain
date. If you aren't keeping up with CVS changes, why are you running
CVS at all?
The security advisory merely says update using CVS to a date later than Aug 15.
It does not
On Wednesday 10 September 2003 02:02 pm, Lee Goodman wrote:
Could the bindaddr=x.x.x.x be a way to make * work through a NAT?
I have * and a few 7960 phones behind a NAT. I am trying to
register with a proxy on the outside of the NAT. Registration is
ok, but the VIA field has my inside NAT ip
On Wed, 2003-09-10 at 13:51, James Sharp wrote:
On Wed, 2003-09-10 at 11:55, James Sharp wrote:
If I have a system with 1 machine to handle incoming H.323 calls and
then
multiple machines to distribute them to T1 ports over TDMoE, where does
the codec translation take place? Does it
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Olle E. Johansson
Sent: Wednesday, September 10, 2003 2:22 PM
To: [EMAIL PROTECTED]
Cc: Free World Dialup - The Future of Dialing
Subject: Re:
Lee Goodman wrote:
Could the bindaddr=x.x.x.x be a way to make * work through a NAT?
I have * and a few 7960 phones behind a NAT. I am trying to register with a
proxy on the outside of the NAT. Registration is ok, but the VIA field has
my inside NAT ip address (192. 168.0.7). So the proxy doesn't
On Tue, Sep 09, 2003 at 02:38:01PM -0500, Eric Wieling wrote:
That would be reinvite= and canreinvite= in the user entry for each SIP
endpoint. Asterisk will allow the endpoints to talk directly to each
other if both those settings are = yes (the default, I think) AND both
endpoints use the
Read the security vulnerability. It referenced CVS
as of a certain
date. If you aren't keeping up with CVS changes,
why are you running
CVS at all?
One would hope people are not using the latest CVS
checkup as their production system. Most sane people
do a bit better quality control and
On Wed, 2003-09-10 at 14:39, James Sharp wrote:
If the remote ends can do the codec, then yes. If they can't deal with
the incoming codec, then it will be done at your h323 end point. The
benefit of IAX2 trunking is to cut down on your ethernet load and to
make expanding easier. Not to
I'm pretty sure the info has been posted to the mailing list several
times and should be in the searchable archives.
On Wed, 2003-09-10 at 14:28, Peter Pauly wrote:
On Tue, Sep 09, 2003 at 02:38:01PM -0500, Eric Wieling wrote:
That would be reinvite= and canreinvite= in the user entry for each
Hi
I've problem, i cant get tone on a FXS ZAP channel my configuration are:
-- zaptel.conf --
fxoks=1
--zapata.conf --
[channels]
immediate=yes
context=bell
signalling=fxo_ks
channel=1
--extensions.conf --
[home]
exten = 500,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED])
[bell]
exten =
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Alvaro Ivan Parres Peredo
Sent: Wednesday, September 10, 2003 4:15 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] NO TONE ON ZAPATA FXS CHANNEL
Hi
I've problem, i cant get tone on a FXS
change immediate to no
At 03:15 PM 9/10/2003 -0500, you wrote:
Hi
I've problem, i cant get tone on a FXS ZAP channel my configuration are:
-- zaptel.conf --
fxoks=1
--zapata.conf --
[channels]
immediate=yes
context=bell
signalling=fxo_ks
channel=1
--extensions.conf --
[home]
exten =
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Jason A. Pattie wrote:
| I have another observation for you. Do your S100U's get warm? I've
| left this one plugged into the USB port and the red LED is lit on the
| unit even though the wcusb driver is not loaded at this time. I noticed
| that the
My S100U also gets quite warm. I haven't had any trouble with it though.
--Ernest
At 01:31 PM 9/10/2003, you wrote:
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Jason A. Pattie wrote:
| I have another observation for you. Do your S100U's get warm? I've
| left this one plugged into the USB
I have come to realize that I don't have to have a g729a license in
order to make use of an ATA-186 or 7460 connecting to another 7460. I
just need to allow the codec in sip.conf.
Now what ramification does that have when I dial out over one of my
analog line (connected to * by a channelbank and
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Ernest W. Lessenger wrote:
| My S100U also gets quite warm. I haven't had any trouble with it though.
Thanks. I'm just trying to figure out as many variables as I can before
sending it back.
| --Ernest
|
| At 01:31 PM 9/10/2003, you wrote:
|
|
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