I do, like I do with my IAX2 softphone. It's just that I haven't
tookthe time to make a webpage that explains what it does and provide
alink to download it.I already send it to peoples on this list that
asked for it.Anybody want it, just email (privately, since this list is
already
Sorry about the previous post. Is this still available? The main
thing is I need a management tool I can use in commercial sales.
Regards,
Chris
[EMAIL PROTECTED]
Original Message
I do, like I do with my IAX2 softphone. It's just that I
Thanks.
Chris
- Original Message -
From: Time Bandit [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, March 25, 2005 12:59 PM
Subject: Re: [Asterisk-Users] Web based Asterisk management tool
Sorry
I have a pretty simple setup. I have a box running
[EMAIL PROTECTED] with an X100P card and 2 Grandstream BudgeTone-100 phones.
I thought it would be rather simple but I seem to be having a hell of a time
getting call waiting to work. I have come across some solutions in the
mailing list
We have 1 Asterisk box, 2 Grandstream B100 phones, and
2 businesses. We have business A using a PSTN line via a X100P card and
business B on a Broadvoice VOIP line. Both phones are for both companies.
What I want to know is if it is possible to some how
distinguish which lines the call
Is there a good place to get the command references for the Extensions.conf?
The Wiki and other documentation seem to be rather limited and don't explain
all the parameters.
Regards,
Chris___
Asterisk-Users mailing list
Can someone help on the marco? when I use this marc to dial out it connects
the call just fine, but never executes the voice-rec marco.
Chris
[macro-dialout-trunk-rec]
exten = s,1,GotoIf($[foo${ECID${CALLERIDNUM}} = foo]?4);check
for CID override for exten
exten = s,2,SetCallerID
I am trying to run a macro from the options on the dial command. I want
to play a sound on an active call and repeat it.I figured I needed to use a
marco, but the macro never activates. The call is placed and connected, but
I never see the macro run. I invoke the macro by
Oh my gosh! I've been staring so long at it that I didn't even see my
typo. I was not talking about *8.I am using the prefix of 8 instead of
9. Like 8401234.
Regards,
Chris
- Original Message -
From: Wilson Pickett [EMAIL PROTECTED]
To: Asterisk Users Mailing List
I need to make a time loop in the Extensions.conf. I want it to play a
file every 5 minutes on a call. If I can't use wait because it ignores all
audio. Anyone have any suggestions?
Regards,
Chris___
Asterisk-Users mailing list
answered and people talking on the
call.
Chris
- Original Message -
From: Race Vanderdecken [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Monday, April 11, 2005 11:28 AM
Subject: RE: [Asterisk-Users] timed Loop
.If an inmate makes a
phone call, I want it to play You are speaking on a recorded call with an
inmate at X facility every five minutes.
Regards,
Chris
- Original Message -
From: Dylan VanHerpen [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users
That won't work on outgoing calls, will it?
Regard,
Chris
- Original Message -
From: Eric Wieling aka ManxPower [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, April 11, 2005 2:46 PM
Subject: Re
I used the example in the Wiki for setting up to * servers with IAX. Both
servers are on my network.I'm trying to get the configuration working
before I put it on the remote site.
I got ServerB to Register with ServerA.If I try to make a call to a SIP
extension on ServerA
Will you send me your relevant sections in IAX_additional.conf and
extensions_additional.conf?
Chris
- Original Message -
From: Tim Litwiller [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, April 12, 2005 4
I would think so, but it doesn't work. It maybe a stupid question, but do
I need to create a route to the trunk? The instructions don't say anything
about it.
Chris
- Original Message -
From: Colin Anderson [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial
to the same context the
zap channels use. I am trying to dial a SIP extension.Any ideas?
Regards,
Chris
- Original Message -
From: Colin Anderson [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Tuesday, April 12, 2005 4
Im running [EMAIL PROTECTED] 0.8 so I dont know if
this pertains solely to the @home version or if you guys can help me but
When I get a voice mail it sends an email with a link to the voicemail
system. Problem is the IP address is wrong and I need to change it.
Christopher Dittrich,
disallow=
context=from-internal
canreinvite=yes
callgroup=
callerid=Test 200
allow=
Regards,
Chris
- Original Message -
From: Franz Knipp [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, April 14, 2005 11:06 AM
Subject: [Asterisk-Users] Siemens optiPoint 420 phone
I've read a lot on this board and in the WIKI. Is there no hope of a
X100P with SpanDSP accepting incoming fax? Everytime I try it fails to
train.
Is there something I have missed that could fix the problem?
Chris___
Asterisk-Users mailing
the server is
installed is also on broadband.
If anyone knows how to setup this scenario, I would much appreciate it and
it would get me on my way with the asterisk server.
Kindest Regards
Chris
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a NIC.
Lyle
- Original Message -
From: Chris [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, December 30, 2004 7:19 PM
Subject: [Asterisk-Users] A simple scenario
Does anyone know how to setup asterisk up so that 1 home user can dial
into
it make and receive calls
Thanks for your response Lyle I appreciate it. I have setup an Asterisk
server at my place of work, this is behind a firewall on a private IP
address range. I have set this system so that a monitored port is allowed
through.
From my home PC I want to connect to the server via an IP call using
would be much appreciated thanks everyone!
-Chris
--
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.300 / Virus Database: 265.6.9 - Release Date: 1/6/2005
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Default for IP 500 (prolly the other too, but not sure)
username: PlcmSpIp
password: PlcmSpIp
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joseph
Sent: Friday, January 07, 2005 9:18 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED] wrote:
Christopher,
Just as a total guess, check to be sure the PoE portion
of his cat5 cable are not either grounded, or touching
each other. Also, be sure he's not connected to a PoE
capeable switch since cisco is reverse polarity of the
PoE standard.
Niles
Nice work Andy.
Filled in some issues I didn't quite understand. Looking forward to more.
Chris
Tielman,
You can take a look at the quick and dirty guide I'm slowly putting
together if you like...
http://www.automated.it/guidetoasterisk.htm
I'd appreciate any feedback you have
I am trying to configure Asterisk to use a Polycom
SoundPoint IP 500 phone. Does anyone know where I can get the software and
configuration file for this phone. I spoke to Polycom support and they say
it's up to the SIP vendor to provide this.
I've got Asterisk up and running nicely using a
couple of different softphones. Audio quality is suffering a bit due to
the hardware that I am working with. So I tried to use a Polycom hardphone but
the politics is enough to give you a headache. Polycom seems to support
SIP only if you buy
hi all,
I'm trying to build an IVRs. anyone here can
spare a sample extensions.conf? or maybe
a link.
Thanks in advance!
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Does anyone have a location to download the latest Polycom firmware etc?
Other than the extranet site, because I am not a reseller, there fore I
have no login.
[minirant]
And shouldn't end users be granted access to this kind of thing anyway?
Geeze
[/minirant]
Thanks,
Chris Cherry
--
No virus
Wow if that phone is Poe, and 80/90 bucks. Thats a steal right
thereawesome.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ken
D'Ambrosio
Sent: Tuesday, December 14, 2004 5:14 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Sipura 841 delayed:
Are you able to get MOH working by setting up an extension in your dialplan?
-Chris
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in signalling/RFC conformance..
Thanks in advance,
Chris Bennett
[general]
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
domain=proxy.myhostname
disallow
missed something.
It *feels* like an Asterisk bug but maybe a SIP expert can spot the
problem in signalling/RFC conformance..
Thanks in advance,
Chris Bennett
[general]
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
missed something.
It *feels* like an Asterisk bug but maybe a SIP expert can spot the
problem in signalling/RFC conformance..
Thanks in advance,
Chris Bennett
[general]
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
hi,
i need instructions on how to configure asterisk as
a media server.
i need your help.
thanks in advance.
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hi,
im installing latest asterisk from cvs on solaris
9. but when i run make i got this error
/bin/sh: build_tools/make_version_h: cannot
executemake: *** [include/asterisk/version.h] Error 1
what i did was chmod 777 all files under
asterisk/build_tools/
and when i run make again i got
/make_version_h: not found
Some time in the future, on Wed, Jul 20, 2005 at 06:03:46PM +0800, chris
wrote:
hi,
im installing latest asterisk from cvs on solaris 9. but when i run make
i got this error
/bin/sh: build_tools/make_version_h: cannot execute
make: *** [include/asterisk/version.h
: not found
chris wrote:
bash-2.05# ls -l
total 30
drwxr-xr-x 2 root other512 Jul 19 12:33 CVS
-rw-r--r-- 1 root root 405 Jul 19 12:33 make_build_h
-rw-r--r-- 1 root root 983 Jul 19 12:33 make_defaults_h
-rwxrwxr-x 1 root root 181 Jul 19
Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, July 21, 2005 3:26 AM
Subject: Re: [Asterisk-Users] /bin/sh: build_tools/make_version_h: not found
On Wednesday 20 Jul 2005 18:31, chris wrote:
hi kevin,
i tried removing the enitre asterisk directory
which script file was affected?
thnks.
- Original Message -
From: chris [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, July 21, 2005 8:50 AM
Subject: Re: [Asterisk-Users] /bin/sh: build_tools/make_version_h
:
include/asterisk/strings.h:232: parse error before `va_list'
include/asterisk/strings.h:232: warning: function declaration isn't a
prototype
make: *** [term.o] Error 1
bash-2.05#
any ideas on how i can fix this?
thnks in advance.
chris.
- Original Message -
From: Tzafrir Cohen [EMAIL
:
include/asterisk/strings.h:232: parse error before `va_list'
include/asterisk/strings.h:232: warning: function declaration isn't a
prototype
make: *** [term.o] Error 1
bash-2.05#
any ideas on how i can fix this?
thnks in advance.
chris.
___
Asterisk
- Original Message -
From: chris [EMAIL PROTECTED]
To: Frank Tarczynski [EMAIL PROTECTED]; [EMAIL PROTECTED]
Sent: Tuesday, July 26, 2005 9:54 AM
Subject: Re: [Asterisk-Users] /bin/sh: build_tools/make_version_h: not found
hi Tzafrir
any idea how we can fix this error?
include
hello,
i just fixed my serctl problem. and when i tried
serctl add usernamepasswrod
email_add
i got thie error,
awk: bailing out near line 1awk: syntax error
near line 1awk: bailing out near line 1HA1 calculation
failed
thnks
___
hello,
i got this error when i run make after downloading
asteirsk from cvs.
gcc -pipe -Wall -Wstrict-prototypes
-Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include
-Iinclude/solaris-compat -I/usr/local/ssl/include -D_REENTRANT
-D_GNU_SOURCE -O6 -Wcast-align
-DSOLARIS
hi
i gues the error is in this line
include/asterisk/strings.h:232: parse error before
`va_list'
can anyone help me please. how can i fix
this?
much thnks.
chris.
- Original Message -
From:
chris
To: asterisk-users@lists.digium.com
Sent: Tuesday, July 26
thnks.
chris.
- Original Message -
From: Steve Drach [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, August 06, 2005 12:04 AM
Subject: Re: [Asterisk-Users] function declaration isn't a prototype
In file included
, however, i have a
new error.
else \
mv include/asterisk/version.h.tmp include/asterisk/version.h ; \
fi
rm -f include/asterisk/version.h.tmp
make[1]: `ast_expr.a' is up to date.
make[1]: Leaving directory `/export/home/fst/chris/cvs/asterisk'
gcc -g -o asterisk io.o sched.o logger.o
hi dave,
any suggestions on myencoutrered problem below?
thnks so much.
chris
- Original Message -
From: chris [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, August 08, 2005 4:51 PM
Subject: Re: [Asterisk-Users
hello,
can anyone help me? im gettitng this error when i
tried runnin make on solaris 9
rm -f include/asterisk/version.h.tmpmake[1]:
`ast_expr.a' is up to date.make[1]: Leaving directory
`/export/home/fst/chris/cvs/asterisk'gcc -g -o asterisk io.o
sched.o logger.o frame.o loader.o
, it is installed by default in
/usr/sfw/lib
Ondrej
Rollin Weeks wrote:
Chris,
The problem is that your compiler can't find a library called
libcrypt.so.0.9.7. This library is apparently needed by
libssl.so. These are both runtime, shared libraries. The
result is that you end up
Command line:
crle -c /var/ld/ld.config -s /usr/local/ssl
bash-2.05#
then i tried make again, but error still exist, 'undefined reference'
pls advise. i really need to make asterisk working on solaris.
thanks so much for the replies. :)
chris
- Original Message -
From: Derek Whitten
hello,i was able to install openssl by hand successfuly, then i
tied make.. theerror is till there,any more
ideas?thnks
- Original Message -From: "Derek Whitten" [EMAIL PROTECTED]To: "chris" [EMAIL PROTECTED]Sent:
Saturday, August 13, 2005 12:29 AMSubject: Re: [A
Has anyone used a hardphone HP300 with asterisk?
Regards,
Chris
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Is there any documentation on how to setup the ASTCC?I've got it working,
but I don't quite understand what the web interface is referring to.
Chris
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You need this before wcfxs
/sbin/modprobe zaptel
Regards,
Chris
- Original Message -
From: Remco Barende [EMAIL PROTECTED]
To: Asterisk Users List asterisk-users@lists.digium.com
Sent: Saturday, April 23, 2005 12:55 PM
Subject: [Asterisk-Users] ztcfg doesn't do anything from /etc
*sigh*
I always get an error if I don't.
Regards,
Chris
- Original Message -
From: Eric Wieling aka ManxPower [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, April 23, 2005 7:15 PM
Subject: Re
I had to manually add the lines to the make file in apps/If you read
the patch file there is only like 4 lines you have to add.
Regards,
Chris
- Original Message -
From: Sahil Gupta [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, May 03, 2005 9:02 PM
a little confusing as well, but it is simple now that I
understand it.
Chris
- Original Message -
From: mr. barker [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Thursday, May 05, 2005 4:43 AM
Subject: [Asterisk-Users
When I receive a fax the tiff is completely black although the fax says it
was successful.
Any Ideas? I have libtiff-3.5.7-22.el3 installed with SpanDSP 0.0.2pre6 (Have
tried pre17)
Chris___
Asterisk-Users mailing list
=xxx.xxx.xxx.xxx
[boxa-user]
type=user
secret=mypassword
host=xxx.xxx.xxx.xxx
context=from-internal
Regards,
Chris
- Original Message -
From: mr. barker [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Thursday, May 05, 2005 1:58
I haven't gotten to keys yet.
The documentation out there doesn't seem to be very good.
Chris
- Original Message -
From: Tim Pushor [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, May 05, 2005 4:06 PM
or IAX tries to communicate. I had trouble with SIP over the internet.
Sip would see the internal address of the Router's WAN. (static NAT) I
haven't figured a way around it.
Chris
- Original Message -
From: mr. barker [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non
I will try it tomorrow. After I got it to work I didn't pay much
attention to the other configurations.
Chris
- Original Message -
From: Tim Pushor [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday
I tried it that way, but it just rings and eventually says all circuits are
busy.
Chris
- Original Message -
From: Tim Pushor [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, May 05, 2005 4:06 PM
Subject
I have.The receiving box doesn't show anything, but tcpdump shows
bidirectional traffic.
I can only think that it is an authentication problem.When you generate the
key do you need the password? Does the name of the key have to match the user
name?
Chris
- Original Message
us the relavent portions of the iax.conf and dialplan, as
well as details as to where you put your keys, both private and public.
Chris wrote:
I have.The receiving box doesn't show anything, but tcpdump shows
bidirectional traffic.
I can only think that it is an authentication
it
is the only thing I have changed.
When I make a call it rings about 10 times and then says All circuits are
busy
Chris
- Original Message -
From: Tim Pushor [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, May 06
.
I added the secrets back and removed the RSA specific config and it started
working again with no problems. It has to be something specific to
encryption traffic for authentication.I have another test server on the
local network I can test with.
Chris
- Original Message
I could only get mine to work
with a Digium card.
Chris
- Original Message -
From:
Ma Zhiyong
To: Asterisk-Users@lists.digium.com
Sent: Sunday, May 08, 2005 11:26 PM
Subject: [Asterisk-Users] spandsp
Hi,
I installed spandsp
No timing?Do you have the wcfxo, wcfxs, or ztdummy loaded?
- Original Message -
From: Daniel Salama [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, May 11, 2005 1:09 PM
Subject: [Asterisk-Users]
Edit the Makefile for the zaptel drivers. You will see two commented
lines that say ztdummy. Uncomment them and rebuild.
Once you install the rebuild, do a modprobe ztdummy and you should be good to
go. BTW, you do need an active USB for ztdummy to load.
Chris
- Original Message
]
[serverb]
type=peer
trunk=yes
username=servera
secret=mypassword
host=216.226.232.143
- Original Message -
From: Chris Mason (Lists) [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Wednesday, May 11, 2005 5:01 PM
I forgot because I haven't moved to a 2.6 kernel.
Chris
- Original Message -
From: BJ Weschke [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, May 11, 2005 6:56 PM
Subject: Re: [Asterisk-Users] Problem
It sounds like you don't have USB support compiled in the kernel.
Chris
- Original Message -
From: Daniel Salama [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, May 11, 2005 11:55 PM
Subject: Re: [Asterisk
You will have to add the 4 or so lines to the make file manually.
Chris
- Original Message -
From: Mark Ratering [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, May 24, 2005 7:51 AM
Subject: [Asterisk-Users] spandsp issue
I'm trying to compile spandsp
Does anyone know of a E911 interface I can get?
Regards,
Chris
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Has anyone experienced problems with Vontage and Asterisk. I'm using
Asterisk (Current Stable) and Sipura-841 phones.While talking on an
outbound call the transmission seems to fade out until the other person can't
hear me but I can hear them.
I've updated the firmware on the 841
Try this
http://www.voip-info.org/wiki/view/Asterisk+Teliax
- Original Message -
From:
Rob
Fugina
To: asterisk-users@lists.digium.com
Sent: Monday, October 17, 2005 5:14
PM
Subject: [Asterisk-Users] Teliax IAX
problems -- Asterisk doesn't see answer
Do you actualy send faxes through them?
Regards,
Chris
- Original Message -
From: Chris Mason (Lists) [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, October 28, 2005 7:27 PM
Subject: Re: [Asterisk-Users
I've been using Teliax.com.
Chris
- Original Message -
From: Jason Brashear [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, November 03, 2005 10:03 AM
Subject: [Asterisk-Users] Looking por a provider to work with asterisk
I know about broadvoice.com
If you use a SIP video phone with ASterisk. Does the monitor function
record video and audio?
Regards,
Chris___
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It's not too difficult. I know several Asian companies who would gladly
develop the card if you send them a sample and pay a development fee.
Chris
- Original Message -
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, November 16, 2005 3:38 PM
Subject
.
The digium cards cost $300+ a useful alternative that cost $100 +/-
would be nice.
I have generally only bought cable connectors from the Asian
companies.However one of them produces a nice DVR unit for vehicles they
claim they developed from scratch.
Chris
- Original
hello,
i change my OS from solaris 9 to solaris 10, tried
running make to install asterisk but i'm getting the error below:
make -C editline libedit.amake[1]: Entering
directory `/export/home/fst/ice/cvs/asterisk/editline'/bin/sh makelist -h
common.c common.h/bin/sh makelist -h emacs.c
i can fix this.
thnks
- Original Message -
From: Frank Tarczynski [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, August 29, 2005 3:08 AM
Subject: RE: [Asterisk-Users] error compiling on solaris 10
Message: 11
Date: Sun, 28 Aug 2005 11:46:29 +0800
From: chris
Is there a problem at Teliax? I'm looking for a VoIP provider and when I call
them they never answer the phone and the voice mail says it's full.
Chris___
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Asterisk-Users mailing list
: Monday, August 29, 2005 12:18 PM
Subject: Re: [Asterisk-Users] teliax
\I concur. They seem to be always busy.
Chris wrote:
Is there a problem at Teliax? I'm looking for a VoIP provider and when I
call them they never answer the phone and the voice mail says it's full.
Chris
Their online support says off line and goes to email.I have emailed
them several times and still haven't got answers to my questions.Everytime
I get a response from them I have to repeat my question and then I never hear
the answer.
Regards,
Chris
- Original Message
Their plans look good, but it just feels like I am being ignored. Some
guy named David emailed me off the Asterisk-biz list from Teliax with his
direct number.I'll give that a try.
Regards,
Chris
- Original Message -
From: Darrick Hartman [EMAIL PROTECTED]
To: Asterisk
hello,
any advise? :)
thnks
- Original Message -
From: chris [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, August 29, 2005 3:26 PM
Subject: Re: [Asterisk-Users] error compiling on solaris 10
hi frank,
i
No, I don't have service with them.
I am thinking about getting service from them and I had some specific
questions about porting telephone numbers and clear up some things about their
packages.
Regards,
Chris
- Original Message -
From: Rick Baranowski [EMAIL PROTECTED
: Tuesday, August 30, 2005 9:43 AM
Subject: Re: [Asterisk-Users] error compiling on solaris 10
Message: 26
Date: Mon, 29 Aug 2005 15:26:31 +0800
From: chris [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] error compiling on solaris 10
To: Asterisk Users Mailing List - Non-Commercial
Generally I have used Intel Chipsets on ASUS motherboards. I've always
used Kingston RAM. I've used Intel P4 CPU on S478 and LGA775.
The Asus boards almost always have NIC and sometimes on board VGA. I've not
had any problems with the hardware.
Regards,
Chris
- Original
I've had lots of luck with the Intel/Asus and I am the part supplier.
Chris
- Original Message -
From: Matt Florell [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, September 08, 2005 9:58 AM
Subject: Re
I'm not writing a printer driver so I probably couldn't use the idea.
I've always disabled CUPS.
Regards,
Chris
- Original Message -
From: Tzafrir Cohen [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, September 09, 2005 1:04 AM
Subject: Re: [Asterisk-Users
It seems that using AstFax would mean that you would have to have a
dedicated email server for faxing.
AstFax expects the number in the email address.So all emails would have to
be piped to the program.
Which maybe fine in some circumstances.
Am I wrong?
regards,
the htpasswd utility to add the user and password and then you will have to
modify the httpd.conf file
Chris
- Original Message -
From: Zeeshan Zakaria [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Friday
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