Thanks, this is exactly what I was looking for. I tried experimenting with
different codecs myself, and GSM seems to be the only one that works...
neither iLBC or Speex went thru. I'm using XLite v1.x Asterisk 0.5.0,
wonder if it's a softphone's problem?
I have got X-Lite to work with
Cool. I know for a fact my lab setup has it commented out, so
with a small tweak I'll be doing real testing.
IANAL, but SmartNET likely won't cover the skinny license. Even
with Call Manager you have to buy a license for every phone you
deploy. The license varies from model to model, but
You can try AVM FRITZ with chan_capi from kapejod.
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de YO Internet
Information
Enviado el: lunes, 22 de septiembre de 2003 0:03
Para: [EMAIL PROTECTED]
Asunto: Re: [Asterisk-Users] ISDN BRI hardware
Hi,
Anybody have lots of experience with PCI ISDN cards and Asterisk? I'm
thinking of getting a BRI in my house to deliver more advanced signaling to
my PBX (yes, I'm a geek :-)) but I've never played with isdn4linux.
Is there any particular BRI card that works better with Asterisk than
-= On Sun, 21 Sep 2003 14:43:47 -0700, Mark Hagler [EMAIL PROTECTED] said:
Hi,
Anybody have lots of experience with PCI ISDN cards and Asterisk? I'm
thinking of getting a BRI in my house to deliver more advanced signaling to
my PBX (yes, I'm a geek :-)) but I've never played with
Hi,
Is there anyway to use xlite though a nat
I have a xlite - nat- asterisk.
* is on a public IP.
When I do this, I get an error on the asterisk server because it is trying
to use the dirty ip of the computer running xlite.
All of the settings in xlite seem to have no effect!
Hello,
I have 5 digium's g.729 codecs and succesfully
register with asterisk, I have incomming call from my cisco AS5300 to
Asterisk through IP. But Asterisk always use g711 ulaw instead of g.729. When I
disable all other codecs other than g.729 in both cisco and asterisk, calls get
the G.729 from digium are the G.729A
type.
Greetings,
Tj
-- Tjardick van der Kraan
Tel +32 4 34 40 522Fax +32 4 34 40 525GSM
+32 497 45 27 36
IAXtel: 1 700 344 0522FWD: 26322IPtel:
91331
Belgium
- Original Message -
From:
Chee
Foong
To: [EMAIL PROTECTED]
Hi,
I have a problem with asterisk-0.5.0 which I don't
understand. The monitor says when making a call:
*CLI -- Executing Dial(SIP/roger-c456,
Modem/ttyI0:BYEXTENSION|60|tTm) in new stack
-- Called ttyI0:1234567890
WARNING[196621]: File dsp.c, Line 1198 (ast_dsp_process): Unable to
IC, does that means they are not
compatible?.
Funny thing is, call make from asterisk to
AS5300is fine using codec G.729.
But call from AS5300 to asterisk result in
incompatible codec.
This is very strange.
Foong
- Original Message -
From:
Tjardick van der Kraan
To:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Monday 22 September 2003 04:02, Jeremy McNamara wrote:
You have to enable ring indications
exten = whatever,1,Dial,CAPI/22545070:b98013356|300|Tr
That doesn't work when you use H323 directly. As in
Dial(H323/ip$12.34.56.78|120|r) ... Works fine
Are you using SIP or H323? If SIP, what are the allow= and disallow=
lines in your sip.conf?
On Mon, 2003-09-22 at 03:08, Chee Foong wrote:
IC, does that means they are not compatible?.
Funny thing is, call make from asterisk to AS5300 is fine using codec
G.729.
But call from
I have found that mixing the Dial() format with | can cause problems.
Does Dial(H323/ip$12.34.56.78,120,r) work as expected?
On Mon, 2003-09-22 at 03:09, Tais M. Hansen wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Monday 22 September 2003 04:02, Jeremy McNamara wrote:
You have
Hi all,
when I try register my netergy SIP Phone with *, I can't do it
due to the next message:
1 headers, 2 lines
Reliably Transmitting:
NOTIFY sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK3438300a
From: asterisk sip:[EMAIL PROTECTED];tag=as34fa433f
To:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Monday 22 September 2003 10:16, Eric Wieling wrote:
I have found that mixing the Dial() format with | can cause problems.
Does Dial(H323/ip$12.34.56.78,120,r) work as expected?
Doesn't change anything.
Here's a better explanation of the
Hello,
I am using H.323 with chan_h323.
Here is my config in h323.conf:
allow=g729
if I set allow=ulaw, G7.11 alway get used. Therefore I disallow it. I want
to use G.729. G.711 is too heavy for my network
Any with AS5300 manage to get the digium's g.729 working
Foong
- Original Message
Also, can the BRI interface cards participate in
conference, etc., since they aren't a Zaptel
interface?
I haven't used conferencing but I believe you can
load the ztdummy emulator to get it working..
I am successfully using the zaptelrtc from
http://www.junghanns.net/asterisk/ It uses the
Hi Lele,
that did the work.
now my telco is happy too.
many thanks
Konrad
Lele Forzani wrote:
We connected an * box with an e100p to an E1/PRI from a telco here in Italy.
After we had it working perfectly the telco told us that, despite the circuit
appeared to work fine, and we could place
add a disallow=all above the allow=g729 line.
On Mon, 2003-09-22 at 04:28, Chee Foong wrote:
Hello,
I am using H.323 with chan_h323.
Here is my config in h323.conf:
allow=g729
if I set allow=ulaw, G7.11 alway get used. Therefore I disallow it. I want
to use G.729. G.711 is too heavy for
hello,
I have tried that but get disconnected once asterisk answer the call.
Got the following error
1:02.899 H225 Answer:813ae50 h323.cxx(4167) H323
CreateLogicalChannel - unknown data type
Guess it's the difference btw g.729 on AS5300 and g.729 on asterisk.
Cisco AS5300 has
I doubt that it's a codec problem. Maybe chan_h323 doesnt' support
G729. JerJer would know.
On Mon, 2003-09-22 at 04:55, Chee Foong wrote:
hello,
I have tried that but get disconnected once asterisk answer the call.
Got the following error
1:02.899 H225 Answer:813ae50
Hello,
Camparing chan_h323 config with chan_oh323 config,
In the codec section chan_oh323 allow me to specify frame value.
Is there a equivalent in chan_h323? Or if not, what
is the default frame value if I use G.729(digium).
Foong
Hi,
I am running Redhat, I loaded the mysql and mysql-devel RPM's and then recompiled *..
I thought it would be that simple but it looks like I have missed something becasue it
doesn't look like the module has been complied..
What did I leave out?
--
I had a similar problem a while ago,
The g729 negotiation with chan_h323 might cause problems sometimes with
compatibility between g729a and g729b.
While g729a and b are perfectly compatible, the as5300 might have problems
recognizing g729b as g729.
(I had to allow g729a,b and ab on my hardware
Hello,
Is there a asterisk developer guide/source code doc
or something like that?
I want to see if I can implement the admin menu
function for meetme.
Foong
-- Original Message --
From: Bill Schultz [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
Date: Fri, 19 Sep 2003 17:28:18 -0800
I'm considering using asterisk to replace an existing PBX in a 40 room hotel and
would appreciate any comments, corrections or
Title: Mensaje
Hi,
I
would like to configure a stage for SIP phones. This stage would be the
next:
two
netergy SIP phones connected to Asterisk through chan_sip.
one
X100P or AVM FRITZ to outside lines.
I
think that sip.conf would be the next:
;;
SIP Configuration for
-- Original Message --
From: Steve Totaro [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
Date: Sat, 20 Sep 2003 12:29:54 -0700
i am just curious how many * systems are in the real world with more than one user.
do you run a certain version? you dont
Hi,
I am running Redhat, I loaded the mysql and mysql-devel RPM's and then recompiled *..
I thought it would be that simple but it looks like I have missed something becasue
it doesn't look like the module has been complied..
What did I leave out?
Ok been doing some testing and I get
Does the X100P card work in PCI-X (3.3v) slots or will it in the future.
Thanks
Chad
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The zlib compression library. zlib-devel or so and zlib would probably be the
packages you are looking for.
That did it.. Thanks..
--
__
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I was just wondering if there was a way that you could
have two calls on one line and switch between the two
without initiating a threeway conversation?
I would imagine that Flash is the way to do this, but
when I Flash twice, a 3-way call is initiated. If I
turn threeway off, then I can't
It does support it but you have to uncomment -DWANT-G729 in h323/Makefile
On Mon, 22 Sep 2003, Eric Wieling wrote:
I doubt that it's a codec problem. Maybe chan_h323 doesnt' support
G729. JerJer would know.
On Mon, 2003-09-22 at 04:55, Chee Foong wrote:
hello,
I have tried that but
On Saturday, September 20, 2003 10:37 PM, Mark Spencer
[SMTP:[EMAIL PROTECTED] wrote:
I was surprised to see that it's 240 volts (peak-to-peak)! Egad..
no
wonder it shocks fingertips.
20Hz (50ms cycle), 2 second long clean sine waveform. I was just
surprised to see twice as much
Dan Austin wrote:
Cool. I know for a fact my lab setup has it commented out, so
with a small tweak I'll be doing real testing.
IANAL, but SmartNET likely won't cover the skinny license. Even
with Call Manager you have to buy a license for every phone you
deploy. The license varies from model
Hi,
Where or wgat is the correct way to set the accountcode= setting when using
chan_capi?
Do I do it in capi.conf or extensions.conf with a setvar?
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Eric Wieling wrote:
I doubt that it's a codec problem. Maybe chan_h323 doesnt' support
G729. JerJer would know.
I babysit systems that terminate hundreds of thousands of G.729 based
H.323 calls per day using chan_h323 and As5300.
Jeremy McNamara
Hi all,
i have big problems to make a h323 call over the gatekeeper from my
provider.
The provider demanded following account data:
H323 ID:XXX-XXX-XX-X
DetinationNumer: XXX
I have configured the oh323.conf following:
gatekeeper=XX.XX.XXX.XXX
alias=XXX-XXX-XX-X
Isx the alias
Its fairly simple.. meetme isn't that big you can find where the hooks are
its commented in the code.
bkw
On Mon, 22 Sep 2003, Chee Foong wrote:
Hello,
Is there a asterisk developer guide/source code doc or something like that?
I want to see if I can implement the admin menu function for
I was just wondering if there was a way that you could
have two calls on one line and switch between the two
without initiating a threeway conversation? I would
imagine that Flash is the way to do this, but when I
Flash twice, a 3-way call is initiated.
It's a tricky one isn't it.. There were
zoa wrote:
I had a similar problem a while ago,
The g729 negotiation with chan_h323 might cause problems sometimes
with compatibility between g729a and g729b.
While g729a and b are perfectly compatible, the as5300 might have
problems recognizing g729b as g729.
(I had to allow g729a,b and ab on
On Sunday 21 September 2003 11:30 pm, Uriel Carrasquilla wrote:
I like it.
I am thinking of putting this query in a C++ but I am a bit concern
on 1) scalability
2) delays in setting up the calls
shoud I be concerned?
The query:
mysql select sum(billsec) from cdr where calldate
'2003-09-01
Hello everyone,
I have posted once a message that I had problem
with Asterisk, ATA and X-Lite.
The problem was: When I called from ATA to X-Lie it
did not want to work. The connectuion apper in astersik but I could not hear
anything.
Right now I updated asterisk from CVS and I still
have
Hi All,
I need an example of sip.conf connection with ICH
My connection don´t works
Thanks!
Miklos
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I need an example of sip.conf connection with ICH
/etc/asterisk/sip.conf:
[iconnect]
type=friend
secret=1234
username=12345678
host=sipauth.deltathree.com
dtmfmode=inband
/etc/asterisk/extension.conf
exten = _61XX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],45)
Hope this helps. If not, what
hello,
got anybody succesfully setup asterisk with three avm fritz pci cards -
using the howto described in
http://www.quiss.org/caiviar/Two-Fritzcards-HOWTO
i already have asterisk working with 2 cards, by when i add third card
and compile driver ( see capiinit debug below ) asterisk freeze
If you are using Microsoft Outlook and you are reading this message you
need to make 500% sure you are not propagating virii. I posted our
support (at) nufone d0t net addy on this mailing list last night and
have never posted it in an unprotected fashion like that anywhere else.
So far today
Hi,
Im new to PBX, and now IP-PBX. I found out
about asterisk and need to find see if it is suitable. Firstly I live in
Australia (any ACA rules I need to know before purchasing
hardware?)
Before I purchase hardware, can asterisk do the
following,
- 3 businesses want me to setup a small
[EMAIL PROTECTED] needs to fix their spam filter. Please stop using it or
learn to configure it.
+ 1 Sep 22 AntiSpam UOL (6828) RE:Re: [Asterisk-Users] MS Outlook
Thanks,
Brian
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On Mon, 2003-09-22 at 13:42, Brian West wrote:
I second that... I have received a load of virii from people on this
list..
Received: from torch.junct.com (sootbox.junct.com [65.168.64.10])
by www.bkw.org (8.11.6/8.11.6) with ESMTP id h8MIcEJ06998
for [EMAIL PROTECTED]; Mon,
Has anyone browsed through the source code and
made a list of menu option for VoiceMailMain2?
Or know of some user documentation hiding
in Internet land some place? If not there well
be soon. Ho hum.
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[EMAIL
Any news on this regard?
If this is not implemented yet, what alternatives do we have? A channel
bank?
- Original Message -
From: Paulo Mannheimer [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, September 11, 2003 10:23 AM
Subject: RE: [Asterisk-Users] Is there any MFC-R2
My company has approx. 500 Cisco CP-7960G IP Phones that are coming out of
service. They were deployed for about 6 months. These include the AC power
adapter and station license. We also have some other related equipment. If
someone is reading this and is interested, shoot me an email
[EMAIL
Suddenly after recompiling my 2.4.22 kernel I can no longer load
chan_zap:
Sep 22 21:25:08 ERROR[16384]: File chan_zap.c, Line 5145 (mkintf): Unable to get span
status: Inappropriate ioctl for device
Sep 22 21:25:08 ERROR[16384]: File chan_zap.c, Line 6638 (load_module): Unable to
register
James Sizemore wrote:
Has anyone browsed through the source code and
made a list of menu option for VoiceMailMain2?
Or know of some user documentation hiding
in Internet land some place? If not there well
be soon. Ho hum.
Start here
http://www.voip-info.org/wiki-Asterisk+cmd+VoiceMailMain2
Also
cvs update the zaptel source and make clean install it.
Jeremy McNamara
Louis-David Mitterrand wrote:
Suddenly after recompiling my 2.4.22 kernel I can no longer load
chan_zap:
Sep 22 21:25:08 ERROR[16384]: File chan_zap.c, Line 5145 (mkintf): Unable to get span
status: Inappropriate ioctl for
I've asked earlier on the mailinglist without getting any answer:
- What's the differences between voicemail and voicemail2 ?
From my initial look I thought it was that voicemail2 allowed contexts as
well as mailboxes.. so you could do virtual hosting or multicompany
working.. Company A has
Does any-e-one know if the 4 port FX0 cards will
be shipping anytime soon?
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I'm using app_festival to speak some text to callers. I'm having two
problems with this. The first is with IAX calls (I've not tried others)
the first few seconds of the speech is garbled. The second problem I'm
having is the the volume of the speech IS VERY LOUD. I tried putting
the following
On Mon, Sep 22, 2003 at 03:41:43PM -0400, Jeremy McNamara wrote:
cvs update the zaptel source and make clean install it.
That did it, thanks a lot.
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Trying to read and understand bits and pieces of chan_sip.c I've found these I would like someone to clarify:
* srvlookup=yes|no
* pedantic
* canreinvite=update|yes --update seems new
Being curious, especially for srvlookup functionality...
/O
___
On Mon, Sep 22, 2003 at 03:25:00PM -0400, Sales wrote:
My company has approx. 500 Cisco CP-7960G IP Phones that are coming out of
service. They were deployed for about 6 months. These include the AC power
adapter and station license. We also have some other related equipment. If
someone is
Actually, MS Outlook by default blocks all executables. I'm not sure why
there is so much negativity around the Outlook client. Perhaps we
should all go back to the cave and use Pine.
-Sean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
-- Original Message --
From: Louis-David Mitterrand [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
Date: Mon, 22 Sep 2003 22:28:40 +0200
On Mon, Sep 22, 2003 at 03:25:00PM -0400, Sales wrote:
My company has approx. 500 Cisco CP-7960G IP Phones that are
I am running CVS-09/11/03-14:03 on Redhat 9.0
Trying to get call waiting / call waiting callerid working.
The setup is:
X100P asterisk - MGCP -- analog phone.
What changes to I need to make to my mgcp.conf and extensions.conf file to
allow answering of Call waiting calls? And how do you answer
Are you selling the phones individually?
-Original Message-
From: Sales [mailto:[EMAIL PROTECTED]
Sent: Monday, September 22, 2003 3:25 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] re: Anyone looking for IP Phones?
My company has approx. 500 Cisco CP-7960G IP Phones that are
On Mon, 2003-09-22 at 14:30, Sean Heiney wrote:
Actually, MS Outlook by default blocks all executables. I'm not sure why
there is so much negativity around the Outlook client. Perhaps we
should all go back to the cave and use Pine.
-Sean
Well, that's Outlook 2002 (and maybe with a
What am I if I use mutt...besides virus free? ;)
On Mon, Sep 22, 2003 at 04:30:49PM -0400, Sean Heiney wrote:
Actually, MS Outlook by default blocks all executables. I'm not sure why
there is so much negativity around the Outlook client. Perhaps we
should all go back to the cave and use Pine.
On Mon, 2003-09-22 at 14:33, Ariel Batista wrote:
I am very interested in why your no longer using these phones. We have one for
testing and so far it's not working. Are they not working? Is the main problem
configuration? Any information will be very helpful.
I think you're
On Mon, Sep 22, 2003 at 04:33:44PM -0400, Ariel Batista wrote:
I am very interested in why your no longer using these phones. We have one for
testing and so far it's not working. Are they not working? Is the main problem
configuration? Any information will be very helpful.
There web
On Mon, 2003-09-22 at 15:30, Sean Heiney wrote:
Actually, MS Outlook by default blocks all executables. I'm not sure why
there is so much negativity around the Outlook client. Perhaps we
should all go back to the cave and use Pine.
I'll assume you don't understand the english words you just
I have looked both these URLs over , neither is a User level
description of the menu choices. I'm trudging through the
code now, the only thing I have found so far that is not listed
in the voice mail prompts is that you can press 0 if you
have a o extinction in the same contest. I was hoping
do you like outlook look feel ?
fdisk /dev/hda
install Linux (redhat/mandrake/slack/debian/gentoo)
install evolution - it rocks
Il lun, 2003-09-22 alle 22:30, Sean Heiney ha scritto:
Actually, MS Outlook by default blocks all executables. I'm not sure why
there is so much negativity around
Please help! When I try to place a call pickup from a cisco phone 7960
using *8 the call is picked up but the other phone continues ringing. Is
there any problem with call pickup in SIP.
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I have posted a link to the tarball of my rather simple AGI script that
allows a user to input a Zip Code (USA only) via DTMF and have the
current weather conditions spoken to them. This is the first release
and I'm sure it will have some bugs. It requires a few modules from
CPAN and the
Just to add another post to that thread...
I'm wondering why people don't use the 'REPLY' button
correctly... some (read many) just hit REPLY on
any list email to start a new thread...
resulting in a messed-up view in thread aware clients...
thread views are very good, since can give an overview
On Mon, 2003-09-22 at 15:42, Manuel Marn Garca wrote:
Please help! When I try to place a call pickup from a cisco phone 7960
using *8 the call is picked up but the other phone continues ringing. Is
there any problem with call pickup in SIP.
It's a known problem... I wish someone would
Does anybody have a reasonable solution for an Outlook MAPI plugin that
works with asterisk? At very least, I would like Asterisk to push incoming
call information to the computer, which should then open an Outlook form,
launch a web browser, etc. Beyond that, it would be cool to have Outlook
Here's one thats way out in left field... don't use call pickup! :P
Problem solved sorta!
bkw
On Mon, 22 Sep 2003, Jared Smith wrote:
On Mon, 2003-09-22 at 15:42, Manuel Marn Garca wrote:
Please help! When I try to place a call pickup from a cisco phone 7960
using *8 the call is picked
Hello -
The final schedule for the Asterisk birds-of-a-feather meeting (as
an adjunct to the VON conference) in Boston looks like this:
Tuesday, September 23rd at 8:15 at VinnyT's of Boston near the Hynes
Convention Center.
We'll try to get a corner booth in the downstairs room, and look
Hello,
Actually call from asterisk to AS5300 works fine with G.729. But not the
other way round.
I have tried enable all codecs, enable only g.729 on AS5300 but did not
manage to get it work
May I know what's you setting on both side Jeremy?
Thanks for the reply
Foong
- Original Message
Is there a recommended OS that Asterisk should be used with?
I have been trying to get Asterisk running on Red Hat 9.0 with little success.
Thanks!
Michael
Go back to your cave.
On your way, don't forget to patch sendmail (twice in the last 30 days),
OpenSSH, gtkhtml, and pam_smb. Just in the last month. Linux.
Security. Made for the Internet. Made for the cave.
Regards,
Sean
-Original Message-
From: [EMAIL PROTECTED]
On Mon, 2003-09-22 at 21:39, Michael A. Miller wrote:
Is there a recommended OS that Asterisk should be used with? I have
been trying to get Asterisk running on Red Hat 9.0 with little
success.
You hit 2 of my pet peeves at once. Fist, please understand that HTML
has no business in normal
Chee Foong wrote:
Hello,
Actually call from asterisk to AS5300 works fine with G.729. But not the
other way round.
I have tried enable all codecs, enable only g.729 on AS5300 but did not
manage to get it work
May I know what's you setting on both side Jeremy?
My systems only do termination:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael A.
Miller
Sent: Monday, September 22, 2003 10:40 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Recommended OS
Is there a recommended OS that Asterisk should be used with? I have been
trying to
Hi all.
I'm trying to get a simple configuration working so I can later expand it to
something more interesting.
I'm using kphone to call an extension on the * server. When I try to connect,
I get this error:
DEBUG[81926]: File chan_sip.c, Line 3562 (check_user): Setting NAT on RTP to 0
Yes, you are right. H.323 incoming call from the As5300 doesn't succeed.
outgoing call to AS5300 works fine like your system.
Foong
- Original Message -
From: Jeremy McNamara [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, September 23, 2003 11:14 AM
Subject: Re: [Asterisk-Users]
And we all certainly know that Windows is so secure. I am by no means a
Linux or Windows fanatic, they each have their strong spots.
And I find this thread a little off topic, totally not related to
Asterisk or VoIP/phone systems.
-Original Message-
From: [EMAIL PROTECTED]
On Monday 22 September 2003 11:25 pm, Andrew Joakimsen wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael A.
Miller
Sent: Monday, September 22, 2003 10:40 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Recommended OS
Is there a
Try
Nat = yes
Or
Nat = no
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Mike Diehl (Encrypted email
prefer
red)
Sent: Monday, September 22, 2003 11:38 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Can't get simple
Well, didn't I SAY it was a simple config? grin
Thanx, that worked.
Mike
On Monday 22 September 2003 09:49 pm, Andrew Joakimsen wrote:
Try
Nat = yes
Or
Nat = no
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Mike Diehl
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Michael A. Miller wrote:
| Is there a recommended OS that Asterisk should be used with? I have been
| trying to get Asterisk running on Red Hat 9.0 with little success.
I have successfully gotten * to compile on Redhat 9.0 and ClarkConnect
v1.2 (RH7.3
and I think use those cvs with rh7.3 and apt for RH is works well :-)
mack_jpn
Tilghman Lesher wrote:
On Monday 22 September 2003 22:37, Steve wrote:
On Monday 22 September 2003 11:25 pm, Andrew Joakimsen wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
On Tue, 23 Sep 2003, Michael A. Miller wrote:
Is there a recommended OS that Asterisk should be used with? I have been
trying to get Asterisk running on Red Hat 9.0 with little success.
At this time, Asterisk only seems to run with Various versions of Linux.
There are patches for other unix
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