HI Matt
I'm unsure about Fedore, but hav several instances of * running with
different avm cards running find (usb, fritz and c2) on debian stable
(woody).
The tricky part is to find the correct verion of the avm drivers to load
(since they are compiled for suse.
Does capiinfo show anything?
I am pretty sure that I had used meetme in the past (many months ago)
with great results. Small number of users, mixed connections, IAX2
and SIP.
For the past month or so, meetme has been a real pain due to very
large latency. I can take 2 phones on the local lan and still get many
seconds of lat
On Tue, 19 Oct 2004, Michael Loftis wrote:
> We figured it out. Well I did. You pretty much have to use
> pridialplan=unknown in zapata.conf it looks like, with the others libpri
> seems to try to get stupid with the actual digits sent/coded to the remote
> switch.
Also, your telco may int
Hi
I am just wondering if chan_mISDN is a worthwhile alternative to
zaphfc which I am having issues with. I have 2 hfc-s modem cards
in my asterisk box.
Any comments or advice will be appreciated.
Thanks
Clive
On 19 Oct 2004 at 11:16, Brian West wrote:
> Well the error does give you some clue
Brent Franks wrote:
> Hello,
>
> Our client currently has two X100P's running in an HP box that has
> been running for almost a year now with no problems. They have found
> however that two phone lines are not enough and are bringing in a
> third phone line. I wouldn't expect this line to be used
I'm using 3 X100Ps with no problems in an old IBM machine.
-Tim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brent
Franks
Sent: Tuesday, October 19, 2004 11:31 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Sipura or X100P Option
Hello,
Our clien
I was just trying to find out if the fact that the driver doesn't load
is because it's
not plugged into the phoneline, thats all.
On 19-okt-04, at 23:00, Benjamin on Asterisk Mailing Lists wrote:
On Tue, 19 Oct 2004 21:55:57 +0200, Alex van Es <[EMAIL PROTECTED]>
wrote:
I just got my X100P card
Hello,
Our client currently has two X100P's running in an HP box that has been
running for almost a year now with no problems. They have found however
that two phone lines are not enough and are bringing in a third phone
line. I wouldn't expect this line to be used very often as there are
only t
Current CVS has some realtime changes for voicemail & sip (pull from
database, no reload required!) can we expect to see more fo the same?
Could the dial plan eventually be databased? Or Is this even possible
due to the complexity of it. (its obviously possible, but would it just
confuse things m
el Flynn wrote:
> Hi all,
>
> Just wanted to see what you guys have to say about the setup we're
> planning to install - 16 incoming POTS lines, 50 extensions. As it is,
> I've got two options:
>
> 1) Lots of ATAs
> 1 x * server
> 4 x TDM04B for 16 incoming lines (can't do fractional E1 - client's
Hi all,
Just wanted to see what you guys have to say about the setup we're
planning to install - 16 incoming POTS lines, 50 extensions. As it is,
I've got two options:
1) Lots of ATAs
1 x * server
4 x TDM04B for 16 incoming lines (can't do fractional E1 - client's
requirements)
25 x Sipura SPA-
In my case, I was running two X100P's. Not exactly the TDM40, but
should be the same concept. The driver for that card is slightly
different, in the fact that it uses the wcfxs kernel module (even on
FXO interfaces), rather than the wcfxo module of the X100P. However, I
doubt it makes a difference,
Brian West wrote:
Yes but I'm just saying that if you want it get a checkout it from now as in
"THIS POINT IN TIME" otherwise you're gonna have fun in the next few weeks
once the major changes start going in.
Oooh /me is getting excited about the new changes...any hints of things
to come?
--
Chee
Pavel, have you resolved the CCM issue?
I have the same problem, I can place calls from Asterisk to CCM but not the
other way, same zero tcpdump when going CCM -> Asterisk? Think it is
something to do with the CCM Media Termination Point but all shows OK.
Reams of CCM logs don't really say what i
Thank you Marconi.
I'll try it soon.
tatsuya
Where to get it:
http://www.carcara.lncc.br/marconi/mr_busydetect.patch
tatsuya
Hi,
The other admin must've thought it was a temp file. I did put the file
back there, so it must be accessible now.
___
Asterisk
Anything after these versions:
zaptel.c version 1.95 (known working)
chan_zap.c version 1.357 (known working)
with a tor2 card... causes kernel panic...
Can anyone else confirm this? I honestly think it's a combo issue with the
new zap reload and that zaptel change. But I have spent hours try
Is this mike oconnor as in the Australian mick oconnor
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike
O'Connor
Sent: Wednesday, 20 October 2004 11:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] chan_h3
> "Matthew" == Matthew Boehm <[EMAIL PROTECTED]> writes:
Matthew> There is no way to convert existing files to g729?
The reference codec has a cli to do that. It converts from raw
16-bit signed linear files (sox filetype sw) to g729 files that
should work with *'s format_g729.
I beleive it
Check your ping times throughout the day. A ping crossing through between 2
diferent fast ethernet switches, if you ever see pings above 10ms you might
have some concern, but if you don't (and you most likely will not) you should
be 100% as far as network bandwidth is concerned.
Try checking ou
We figured it out. Well I did. You pretty much have to use
pridialplan=unknown in zapata.conf it looks like, with the others libpri
seems to try to get stupid with the actual digits sent/coded to the remote
switch.
___
Asterisk-Users mailing list
[
Hi All
Is there a better mailing list where I should ask these questions ?
Thanks
Mike O'Connor wrote:
Hi all
I spent a few hours trying to information on asterisk, h323 and sip
support for codecs with 20ms packetisation, and have not been able to
find anything relivatant.
Our supplier of call t
Ok.. total brain fart.. sorry..
lol
:)
On Oct 19, 2004, at 5:55 PM, Matthew Boehm wrote:
2 PAP2NA's with 2 ports each = 4 lines
Matthew
- Original Message -
From: "Darren Sessions" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[EMAIL PROTECTED]>
Sent: Tu
> I was wondering if anyone has any side-by-side comparisons of capacity on
> Mac hardware vs. Intel/AMD hardware, doesn't have to be anything official,
> just some people with real-world asterisk on PPC experience.
I have run a bunch of SIP streams straight through (reinvite=no) a
PIII 500MHz bas
Kanuri, Seshu (Company IT) wrote:
/SNIP/
Have you looked into that open-source implementation of G729? There
was something on the WIKI about 3 >different implementations of it. One
being where you paid license per channel fees, one that was
free/open source, and another I can't remember
Hi Guillermo,
Guillermo Freige wrote:
It's possible to use the "Transfer" function in UniCall MFC/R2 lines?.
The command seems to do nothing when called from a R2 call, but it
works fine from a SIP phone. Transfer and 3waycalling options are set
to "Yes" in unicall.conf. I've tried the "hook" co
Hi all,
Does anyone know if its possible to get an ATA 286 to make the
actual phone to ring instead of just the ATA ringing?
Cheers,
Dee
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[EMAIL PROTECTED]
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T
I am running the 7920, and I got the phone to login and
register (I think) and the Asterisk server see the button presses. However I
get:
Device SEPxxx sent a keypress, but there is no
active channel.
If I press forward it sees the Call forward function. At any
rate I get th
Yes but I'm just saying that if you want it get a checkout it from now as in
"THIS POINT IN TIME" otherwise you're gonna have fun in the next few weeks
once the major changes start going in.
bkw
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On
The separate context is to enforce 'proper' extensions sorting, take a look
at the wiki.
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To UNSUBSCRIBE or update options visit:
http://lists.dig
I'm having trouble getting outbound calls out, immediate release on local
calls from the partner switch, long distance calls seize the trunk but
according to the switch eng. at the CO they're only seeing 10 digits, not
11 (IE theres no 1 ) and the 7 digit dialing doesn't show up at all on
their
How to get NA version from Linksys.
We are ISP and VoIP provider and would like to sell these box'es to
our local customers.
I have mailed sales at linksys but no reply.
Could somebody tell me how to get approved to buy NA version of these boxes.
Thank you in advance!
Bartek
- Original Message
On Wed, 20 Oct 2004 00:53:46 +0200, adriavidal @ gmail. com
<[EMAIL PROTECTED]> wrote:
> I'm using a blue G3 with YDL and by now h323 support is broken for PPC,
> hope Digium get on soon.
AFAIK, Digium have nothing to do with H323. It's a third party add on
module. In fact, there are two different
Hi
I have managed to get SER up and running fine, and got my UA (Xlite and
IPphone) talking to each other, all authenticating nicely from the
DBand then I started to look at asterisk, I figured I may as well
split the proxy (ser) and do the call features like voicemail and all
using asterisk.
On Tue, 19 Oct 2004 15:57:54 -0600, Michael Loftis <[EMAIL PROTECTED]> wrote:
> include => bogons
>
> [bogons]
>
> exten => _.,1,what_to_do_for_fat_fingers
Thanks a lot. That does the trick. I tried the _. match before but not
in a separate context and somehow I created an endless loop that was
The NA's allow you to switch providers. There was no hacking/cracking
involved. Plug it in, get a DHCP address, web page, asked for proxy server,
userid, password, *boom* registered with asterisk.
Matthew
- Original Message -
From: "Deon Rodden" <[EMAIL PROTECTED]>
To: "'Asterisk Users Ma
On Tue, 19 Oct 2004, Steve Kann wrote:
> Stefan de Konink wrote:
> > Isn't this an opportunity for Digium to offer encoded G729 files for a
> > fixed price directly encoded from the original wav files?
>
> I think this is an opportunity for people to use unencumbered codecs..
>
> If even just the a
I'm using a blue G3 with YDL and by now h323 support is broken for PPC,
hope Digium get on soon. For the joy of PPC users.
Adrià Vidal
On 19 Oct 2004, at 20:34, Brian McSpadden wrote:
I can't say that I have yet had extensive experience with it yet, but
what I have seen is promising. I, like you
On Tue, 19 Oct 2004 17:58:00 -0400, Steve Kann <[EMAIL PROTECTED]> wrote:
> This explicitly repeats the invalid number back to them; you could
> prefix it with a message saying "the number you dialed" and postfix with
> "is invalid, blah blah"..
>
> exten => i,1,SayDigits(${INVALID_EXTEN})
> exten
Gonzalo Servat wrote:
What is this huge OH MY GOD difference between the two? (apart from the
-NA). I've googled and can't seem to find any site that lists the
difference(s).
The non-NA version is for Vonage service only, and comes out of the box
locked.
___
Should have googled with "site:digium.com"
PAP2-NA = generic version, configure yourself
PAP2= VONAGE-only version, possibly locked, definitely subsidized
> -Original Message-
> From: Gonzalo Servat [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, October 19, 2004 5:23 PM
> To: [EMAIL PROT
Yes, it's true! Connect the card to a phone line and the Red Alert
disappears. I don't think it draws power from the phone line, but it gives a
red alert if the phone line is not there. I have experienced it myself!
Yiannis.
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL
Hello Guys
Im trying to get Asterisk with my AVM fritz Card (ISDN) to
work. ( fedora core 1 )
I did found a easy how to.. it was posted from someone here
on this Mailing List
Im referring to http://lists.digium.com/pipermail/asterisk-users/2004-June/052118.html
MY PROBLEM: I c
On Tue, 2004-10-19 at 16:56 -0500, Matthew Boehm wrote:
> No. The link you gave is for a PAP2 NOT a PAP2NA. There is a HUGE OH MY GOD
> difference between the two model numbers.
What is this huge OH MY GOD difference between the two? (apart from the
-NA). I've googled and can't seem to find any s
Deon Rodden wrote:
When do you think the last stable CVS will be available before "lots of
stuff" begins to change? I want to find the best possible Asterisk and stick
with it, for some time, maybe until 2.0; If I get CVS right now, what if
tomorrow or the day after he comes out with a better CVS.
--On Sunday, October 17, 2004 10:31 -0400 Brian Kurkowski
<[EMAIL PROTECTED]> wrote:
Michael,
I usually read and don't do much posting, but I had to on this.
Sorry things getting badly buried lately recent reply brought this
thread back to my attention and I realised I'd missed this post.
On Tue, 2004-10-19 at 16:23 -0500, [EMAIL PROTECTED] wrote:
> http://www.amazon.com/exec/obidos/tg/detail/-/B0002V8KX6/
> ref=ase_interactiveda81-20/104-3724093-2519924?v=glance&s=electronics
--
That doesn't appear to be a link to a NA, my understanding was they
weren't going to be available the
Hi all,
I have the following net:
sip-phone --> * --> portaSIP
phone is x-lite on win and Asterisk is on Debian. The sip provider has
portaSIP, I don't know any more.
The recording has succeeded, 'sip show peers' and 'sip show registry' are OK,
while I have a problem on the INVITE:
Oct 19 23:28
Hello, all. I'm thinking of installing Asterisk at my company, and
someone asked if our current network might not have some issues. I
don't think so, but I'd like a vote of confidence to be sure. Our
network has a gigabit switch, with 6 100 Mbit switches plugged into it,
and roughly 150 ethernet
Stefan de Konink wrote:
Isn't this an opportunity for Digium to offer encoded G729 files for a
fixed price directly encoded from the original wav files?
I think this is an opportunity for people to use unencumbered codecs..
If even just the asterisk community got together to put half their G729
If I had the choice of buying Sipura SPA-2000's or Linksys's PAP2-NA's, I
should go with LinkSys?
I thought linksys didn't support it working with any other provider other
than vonage? Changing it would void their warranty or tech support or
whatever.
You got the PAP2-NA's for ~$50? That's a bet
Benjamin on Asterisk Mailing Lists wrote:
This is a question I am almost too embarassed to ask but here we go ...
Is it not possible to use the i extension to trap attempts of users
misdialling numbers otherwise not in the dialplan/context?
I have seen this in so many examples and I always thought
No. The link you gave is for a PAP2 NOT a PAP2NA. There is a HUGE OH MY GOD
difference between the two model numbers.
Matthew
- Original Message -
From: <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[EMAIL PROTECTED]>; "Brandon Patterson"
<[EMAIL PROT
Ummm... It used to be a while back there were 2 different CVS servers or
directories or something, 1 for Head and 1 for Stable. But some time ago,
only one version of CVS showed. I assumed they temporarily merged the 2,
every new release was just a new stable release.
I'm now on the download sect
[default]
include => extensions
include => bogons
[extensions]
exten => 44X,...
exten => 45X,...
[bogons]
exten => _.,1,what_to_do_for_fat_fingers
any context that bogons is included in will have DigitTimeout long times
after finishing dialing. SIP users won't notice this, but those in an IVR
or
2 PAP2NA's with 2 ports each = 4 lines
Matthew
- Original Message -
From: "Darren Sessions" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[EMAIL PROTECTED]>
Sent: Tuesday, October 19, 2004 4:08 PM
Subject: Re: [Asterisk-Users] Wonderful Success with PA
- Original Message -
From: "Brian Kurkowski" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <[EMAIL PROTECTED]>
Sent: Sunday, October 17, 2004 10:31 AM
Subject: RE: [Asterisk-Users] Alternatives to the T100Ps?
| Michael,
|
| I usually read and don'
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Kristian Kielhofner
> Sent: 19 October 2004 22:12
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] X100P red alert
>
> Alex van Es wrote:
>
> > Hi all,
When do you think the last stable CVS will be available before "lots of
stuff" begins to change? I want to find the best possible Asterisk and stick
with it, for some time, maybe until 2.0; If I get CVS right now, what if
tomorrow or the day after he comes out with a better CVS.
I wonder if Mark
> > Hooked a 4-line vtech phone up to 2 PAP2-NAs and basically had created
> > a 4
> > line ATA for $100.
2 ATA's w/ 2 Ports each I think.
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To UNSUBS
where did you get them from?
Gary
- Original Message -
From: "Matthew Boehm" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, October 19, 2004 4:37 PM
Subject: [Asterisk-Users] Wonderful Success with PAP2-NA
Finally got "authorized" to purchase some PAP2-NA's from Linksys's.
W
On Oct 19, 2004, at 4:17 PM, Brandon Patterson wrote:
Brian approved providers get them from Linksys (volume),
Tech Data, DH Dist and Mirco D.
WHERE DID YOU GET THE PAP2-NA?!??!!?
.o---o.
Brian Fertig
Network Engineer
Planet Telecom, Inc.
Tampa,
Hello,
I'm new to Asterisk, I'll be freeing up some hardware to play with it
next week. Would it be possible to eat up 3 PRIs coming from the
phone company (Lucent 5ESS) into one Asterisk box, ship the traffic
over IP to another Asterisk box and back out as a PRI to a legacy
switch? The le
This is a question I am almost too embarassed to ask but here we go ...
Is it not possible to use the i extension to trap attempts of users
misdialling numbers otherwise not in the dialplan/context?
I have seen this in so many examples and I always thought "Oh, this
will come in handy one day" bu
>
> Message: 7
> Date: Tue, 19 Oct 2004 15:37:01 -0500
> From: "Matthew Boehm" <[EMAIL PROTECTED]>
> Subject: [Asterisk-Users] Wonderful Success with PAP2-NA
> To: <[EMAIL PROTECTED]>
> Message-ID: <[EMAIL PROTECTED]>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Finally got "authorized
Mike Taht wrote:
Lately I've been working in relative isolation (e.g. at home) and I
find I like the idea of sharing a virtual audio room with my fellow
programmers. (We all share thoughts currently via irc).
So I'm listening in a asterisk conference room right now (using
SJphone, everyone mic
mattf wrote:
With other projects maybe, but with Asterisk it's hardly the wild side. This
project has consistently had the most stable CVS versions of any project
I've ever used code from.
This will change. Heck, it already has. None of the 3rd party Asterisk
modules that I know of will work wit
Greetings,
Drop the third line.
exten => 31xxx,3,Goto(31xxx,1)
If everything goes correctly with the multiple Dial then Dial will
complete and return and then go to the next step in the dial plan. Which
in your case is to fall to step 3, which sends asterisk back to step 1.
And you start
Brian approved providers get them from Linksys (volume),
Tech Data, DH Dist and Mirco D.
WHERE DID YOU GET THE PAP2-NA?!??!!?
.o---o.
Brian Fertig
Network Engineer
Planet Telecom, Inc.
Tampa, FL Office
813.864.3161x107 Office
813.864.3164 D
Alex van Es wrote:
Hi all,
I just got my X100P card installed and asterisk keeps on complaining
that it cannot create a zap channel. I read somewhere on the internet
that the zap card will not work when a phoneline is not plugged in,
cause is draws power from the phoneline. Is this correct? Of co
I totally agree... if you want DUNDi get cvs-head NOW and I mean NOW.. next
week lots of stuff starts to change and it will NOT be something you will
wanna run in production.
bkw
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Olle E
The card requires no power from the phone company... it is powered from
the PCI bus.
However, I'm not sure if the card can detect if the line is there or
not... I've never tried it. Plug a line into it and see what you get!
:)
On Tue, 2004-10-19 at 12:55, Alex van Es wrote:
> Hi all,
> I just g
The PAP2 is essentially a Sipura. Other than the different skin, a
couple cool L.E.Ds, and an updated web interface - they might as well
be the same box. Linksys's entire line of VoIP boxes are based on the
Sipura technology.
Our experience has been that the Sipura rules supreme in features for
Deon Rodden wrote:
For the SIP client. I just can't imagine using a SIP client over a
connection that has 250+ ms response times. If I make it go online to the
802.11 networks, I can use the SIP Client with ULaw and get high quality SIP
calls at any hotspot. I wouldn't do this for every hot spot, b
-Original Message-
From: Olle E. Johansson [mailto:[EMAIL PROTECTED]
[snip]
> If you want to walk on the wild side, run CVS head...
> Please don't encourage people to use it in production
> environments, even if it from time to time seems to work well.
With other projects maybe, but
On Tue, 19 Oct 2004 21:55:57 +0200, Alex van Es <[EMAIL PROTECTED]> wrote:
>
> I just got my X100P card installed and asterisk keeps on complaining
> that it cannot create a zap channel. I read somewhere on the internet
> that the zap card will not work when a phoneline is not plugged in,
> cause i
On Tue, 19 Oct 2004, Deon Rodden wrote:
> Which codec should be used for 250ms response times? G729A?
What's the big deal about 250msec? My SA to US links are usually more
like 500msec RTT. It doesn't affect the protocols at all.
Steve
___
Ast
WHERE DID YOU GET THE PAP2-NA?!??!!?
.o---o.
Brian Fertig
Network Engineer
Planet Telecom, Inc.
Tampa, FL Office
813.864.3161x107 Office
813.864.3164 Direct
813.817.9961 Cellular
813.881.9762 Fax
Web: www.planet-telecom.com
email: [EMAIL PROT
What's the difference with a Sipura SPA-2000? Did they even change the
Sipura logos on their web pages? :)
E.
On Oct 19, 2004, at 4:37 PM, Matthew Boehm wrote:
Finally got "authorized" to purchase some PAP2-NA's from Linksys's.
Works like a charm with Asterisk. Web configuration has TONS of
opti
On Tue, 19 Oct 2004 13:30:20 -0400, Ferguson, Michael
<[EMAIL PROTECTED]> wrote:
> I just realised that I neglected to mention that the remote GS100 phone
> is sitting behind a firewall also.
Double NAT ?! Boy, you are really asking for trouble.
It's either tunneling or FWD then.
Sign up for two
Ryan Courtnage wrote:
http://www.dundi.com
Yet, another great idea!! Thanks Mark!!
I wish it was in v1.0, but I guess I'll have to update to head.
I wish it were in v1.0 as well. Would creating a patch for 1.0 be pretty
simple, or do the code changes run deep?
It will eventuall get into a release
Finally got "authorized" to purchase some PAP2-NA's from Linksys's.
Works like a charm with Asterisk. Web configuration has TONS of options and
looks nice.
Able to put line1 and line2 on seperate asterisk servers.
Hooked a 4-line vtech phone up to 2 PAP2-NAs and basically had created a 4
line AT
SS,
3600
* 1000 * 3 = 1080 or 10,800,000
You
have 108 or 1,080,000
My max record time
is 3 hours (108ms) and I am using a command similar to the following:
my $x = $AGI->record_file($wavfile, 'wav', '0123456789', 108, 1);
1,08
Thanks!
I of course had tried that a long time ago, but it did not work immediately.
The proper syntax is
exten => s,1,Dial(blah,55,r)
where 55 is my timeout value, in seconds.
However, thanks very much, it works now, your mail prodded me to go at this
again.
On Tue, 19 Oct 2004, Michael Loftis w
Your Own ISP .com wrote:
> I have it set the same for each phone within the sip.conf file if
> this is where you meant.
>
> FYI, I am using Grandstream 101 phones on both ends.
>
> Should it be set to yes for these phones?
If you want media streams to by pass * then set it to "yes". Otherwise
s
> http://www.dundi.com
>
> Yet, another great idea!! Thanks Mark!!
>
> I wish it was in v1.0, but I guess I'll have to update to head.
I wish it were in v1.0 as well. Would creating a patch for 1.0 be pretty
simple, or do the code changes run deep?
Businesses tend to get a warm-fuzzy from kno
On Tue, 19 Oct 2004 15:59:03 -0400, Kanuri, Seshu (Company IT)
<[EMAIL PROTECTED]> wrote:
>
> Remember that we discussed this at length over about 3000 posts. I again
> reiterate that it is called a "HACK" and I am Ok with what you are
> doing. Kevin and Others in this forum has made the licensing
On Tue, 19 Oct 2004, Anton Tinchev wrote:
> What signaling uses asterisk to comunicate with channel banks with the
> T1 or the E1 boards.
There are quite a few signalling protocols to choose from and quite a few
transport protocls as well.
> Is there any differences between T1 and E1 signalling
Brian West [EMAIL PROTECTED] top-posted:
> Head as of now is pretty damn stable.
>
Yes - I'm using CVS HEAD too, without any problems.
--
_/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/
_/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h
_/ _/_/ _/ _/ _/
I
suggest you report it as a bug
Umar
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Smith,
Simon JSent: 19 October 2004 03:02To:
[EMAIL PROTECTED]Subject: [Asterisk-Users] Anybody -
please help me with this
Hi
all,
I have
Remember that we discussed this at length over about 3000 posts. I again
reiterate that it is called a "HACK" and I am Ok with what you are
doing. Kevin and Others in this forum has made the licensing point very
clear.
All I am saying is let us call a spade a spade.
Seshu
eezeePhone
http://ipph
app_voicemail uses ast_streamfile or related. so no changes would be
necessary. if it finds a file already encoded in the correct type for the
channel it's about to play on it'll prefer to use that file instead of
attempting to transcode. when transcoding i believe it will also prefer to
tra
Hi all,
I just got my X100P card installed and asterisk keeps on complaining
that it cannot create a zap channel. I read somewhere on the internet
that the zap card will not work when a phoneline is not plugged in,
cause is draws power from the phoneline. Is this correct? Of course
eventually
I
>Message: 9
>Date: Tue, 19 Oct 2004 14:03:03 -0500
From: "Your Own ISP .com" <[EMAIL PROTECTED]>
>Subject: RE: [Asterisk-Users] How to ring internal extension?
>To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
><[EMAIL PROTECTED]>
>Message-ID: <[EMAIL PROTECTED]>
>Content-Typ
Again, this is all speculation, but I've never seen two definitions for a
user...maybe it doesn't know which to use, so it goes to general where the
context is incoming1. Try changing the username for one of the
sip.broadvoices...
>
> Message: 6
> Date: Tue, 19 Oct 2004 14:52:59 -0400
> From: "E
Lately I've been working in relative isolation (e.g. at home) and I find
I like the idea of sharing a virtual audio room with my fellow
programmers. (We all share thoughts currently via irc).
So I'm listening in a asterisk conference room right now (using SJphone,
everyone mic muted), and also
Head as of now is pretty damn stable.
bkw
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Robert Jackson
> Sent: Tuesday, October 19, 2004 1:56 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] DUNDi on Slashdot
>
> DUNDi made
I am trying to get oh323 to work with the lastest version of asterisk
anyone had any luck
anyone know if this version asterisk-oh323-0.6.3b works
with the latest asterisk
because it wont compile for me
chan_oh323.c:4697: warning: passing arg 4 of
`ast_channel_register' from incompatible pointe
Benjamin,
Thanks for your feedback.
-Original Message-
From: Benjamin on Asterisk Mailing Lists
[mailto:[EMAIL PROTECTED]
Sent: Tuesday, October 19, 2004 2:53 PM
To: Ferguson, Michael
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Almost there--R
On Tue, 19 Oct 2004 11:03:31 -0500, Matthew Boehm <[EMAIL PROTECTED]> wrote:
> There is no way to convert existing files to g729? The only reason we need
> the licenses is to access voicemail since they are in GSM. All our phones
> have g729 built in. But if you try and access VM, you get that "No
Ryan,
Thanks. That looks hopeful.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ryan
Courtnage
Sent: Tuesday, October 19, 2004 2:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Almost there--Remote connecti
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