I've read a lot of conflicting information on this around the web, and
wanted to see if I could get some thoughts for any of you..
What's the proper (or best, etc) build order for install Asterisk and
it's needed libraries. Most often I see 1. Zaptel / Dahdi 2. libpri
3. Asterisk. However, I've
!
--
-Jonathan
On Thu, Apr 23, 2009 at 12:18 PM, Jimmy Ezell wrote:
> Dan thank you, yes that seems to help. It looks like the bridging is
> happening now and I see the light come on in the second FXO port, but then I
> get a busy signal after that and the call still does not
On Wed, Mar 18, 2009 at 4:18 PM, Andrew Furey wrote:
> On 19/03/2009, Jonathan Thurman wrote:
>> Also, is there a way to retain deleted messages for a length of time
>> before they are purged? We currently have that "feature" on our
>> production VM serve
of time
before they are purged? We currently have that "feature" on our
production VM server that I am trying to replicate. Thanks!
-Jonathan
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To
parameter,
a file name. But the attempts I have tried seem unsucessful. I have tried
dialing out and then calling SendFAX and calling SendFAX before the dial.
No success.
Can someone please provide me with an extensions.conf example of how to use
SendFAX?
Thank you.
Jonathan Augenstine
parameter,
a file name. But the attempts I have tried seem unsucessful. I have tried
dialing out and then calling SendFAX and calling SendFAX before the dial.
No success.
Can someone please provide me with an extensions.conf example of how to use
SendFAX?
Thank you.
Jonathan Augenstine
ON
level 1: uniqueid=1235508550.71744
Jonathan Bailey
Marshall County, Iowa
1 E Main St, Marshalltown, IA 50158
- Original Message -----
From: "Jonathan C. Bailey"
To: asterisk-users@lists.digium.com
Sent: Wednesday, February 25, 2009 8:39:42 AM GMT -06:00 US/Canada Central
Subject: [a
nnounce,n,Answer
exten => parkedannounce,n,Wait(1)
exten => parkedannounce,n,Hangup
[parkreturn]
exten => _,1,Noop(Returning Parked Call)
exten => _,n,SIPAddHeader(Alert-Info: info=<${AASTRA_PARKRINGBACK}>)
exten => _,n,Set(CALLERID(name)=FrPark:${CALLERID(name)
Matt,
Asterisk version == 1.4.22
dtmfmode == info
calls are bridged through Asterisk (canreinvite=no)
Jonathan
On Sun, Dec 28, 2008 at 3:23 PM, Matt Florell wrote:
> On 12/28/08, jonathan augenstine wrote:
> > I am trying to resolve an issue and I believe it is my configuratio
configuration issue? Or do I need to handle this on the dial plan level?
Jonathan
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Have you checked out OpenSBC (www.voip-info.org/wiki/view/*OpenSBC)?*
On Fri, Dec 12, 2008 at 6:19 PM, Steve Edwards wrote:
> One of the above is frequently used to front-end Asterisk.
>
> I used OpenSER to front-end a farm of Asterisk servers and was very happy
> with it. The ability to take a b
FYI >> I was informed by A. Minnesale that app_confcall was originally
developed for Asterisk 1.2. He stated that there would probably be a
significant amount of work to update it to Asterisk 1.6.
Jonathan
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I am trying to build app_confcall and it is failing. Are there known build
issues with this module. I am running Asterisk 1.6.0-beta9.
Jonathan
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To
Is it possible to create extensions in the voicemail.conf remotely by using
the manager interface. I cannot seem to find any documents or examples
describing that capability.
Jonathan
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We're using D-Link DES-3028P switches (24 10/100 + 4 gbit). They also have the
DES-3052P which is a 48 port version of the switch. We're paying ~$500, I think
for the 24 port version from Graybar.
-Jon
- Original Message -
From: "David Gibbons" <[EMAIL PROTECTED]>
To: "Asterisk Users M
On Aug 26, 2008, at 5:34 AM, Chris Mason (Lists) wrote:
> Jonathan Disher wrote:
>> He has two buildings (the office,
>> and the shop proper), separated by about 3-400 yards.
> Your inter-building distance exceeds ethernet over copper limits, you
> will need a fiber link.
On Aug 26, 2008, at 2:27 AM, Gordon Henderson wrote:
> Do you have some sort of IP connectivity between the sites? 400
> yards is a
> too long for copper cat5, but can be done with fibre, wireless or
> free-space optics... (which I don't personally recommend!)
The current plan is wireless bridge
I am looking to replace the phone system at my father's shop with an
Asterisk box and some Cisco phones, but one piece of the
implementation is tripping me up. He has two buildings (the office,
and the shop proper), separated by about 3-400 yards. Currently with
the ancient Meridian syste
tpdir: Asterisk channel driver that emulates a radio transmitter
and receiver and sends the audio to rtpDir using UDP over IP digitally
Please write me/the forum and tell me/us about your experiences with
radio systems and asterisk.
Thanks in advance.
Regards,
Jonath
and 5 seconds) and lost packets do not show up with a rtp rtcp
stats...
This is weird. Any help you can offer would be appreciated. We spent 6
hours on phone with Digium support yesterday and could not locate an
issue within asterisk itself.
-Jonathan
On Wed, Aug 13, 2008 at 9:08 PM, Steve T
elp or point me to someone that can.
-Jonathan
[EMAIL PROTECTED]
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asterisk-users mailing l
elp or point me to someone that can.
-Jonathan
[EMAIL PROTECTED]
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Everyone-
We're looking at using some Citel gateways to serve one of our sites (40
extensions, Toshiba phones). I've found that people seem to like the product
from demos, but I was wondering how many have some of the gateways in
production and if they seem to do the job for the long run.
-Jo
I can't seem to find anything via Google, and haven't seen this before.. What
does a channel listed like Zap/0:27-1 mean? I can't figure out what the colon
signifies. I seem to see channel numbers like these just before the T1 card in
my Comdial switch craps itself.
-Jon
__
esday, April 29, 2008 3:21:30 AM GMT -06:00 US/Canada Central
Subject: Re: [asterisk-users] OT: Polycom 3.0
How do they get away with that?
On Mon, Apr 28, 2008 at 7:23 PM, Jonathan C. Bailey
<[EMAIL PROTECTED]> wrote:
> Try the RPM from Trixbox. If you need something to open the
Try the RPM from Trixbox. If you need something to open the file on Windows,
7zip works fine..
http://yum.trixbox.org/centos/5/RPMS/repodata/repoview/firmware-polycom-0-3.0.1-2.html
-Jon
- Original Message -
From: "Darrick Hartman (lists)" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing
We've been using D-Link DES-3028P and DES-3052P switches. They can supply full
power to EACH port unlike the Linksys switches we've tried. They're also rock
solid from our experience.
-Jon
- Original Message -
From: "Hilary Miller" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List -
My guess is that you don't have any spans set up, or Asterisk doesn't have
zaptel support... Is chan_zap.so loaded?
-Jon
- Original Message -
From: "Jerry Geis" <[EMAIL PROTECTED]>
To: asterisk-users@lists.digium.com
Sent: Sunday, April 13, 2008 1:27:56 PM GMT -06:00 US/Canada Central
Su
We use Nagios for network monitoring. We've got a check_pri script that should
be fairly universal. It will return "critical" for any alarm. Feel free to use
the script as you see fit. YMMV - may skin cats, etc (you know the disclaimer
drill)...
#! /usr/bin/python
# Checks PRI status - retur
We used it in our installation and had some issues. We were passing fax and
modem calls through via the second port as a TDM bridged call. For some reason,
the timing was off even though we explicitly set the timing in the redfone.conf
file. We replaced it with a Sangoma A102d and haven't been h
That's surprising.. When I looked at pricing, the Snom 370 was about $50 more
expensive than a 57i for us (the 57i was $205). Also, configuration wasn't too
bad on the Aastra, but that may just be me.
BTW, it also looks like the Snom has support for an electronic headset "lifter"
on some GN Net
Dominik,
apart from the good responses, please get rid of the 't' in the options of
dial or you will be allowing the called party to transfer the call while you
are paying.
Regards,
Jonathan GF
On Dec 26, 2007 3:32 PM, Dominik Zalewski <[EMAIL PROTECTED]> wrote:
> Dear
ian J. Menendez
patch for AnswerOnPolarity... If your provider allows it also, you won't
haver further issues, just tune the echo with fxotune.
Disable echotraining. Not needed. Disable fax detection. Will work, let's
see ;)
Regards,
Jonathan GF
On Jan 4, 2008 11:18 PM, Miguel A F
I'm having (I think) timing issues in relation to bridged T1-T1 calls via
dynamic spans. Fax calls are intermittently working, but voice is fine. My box
has a Sangoma A400 inside it as the primary Zaptel timing source. My T1 PRIs
that are hooked to the box come in via a foneBRIDGE2 (dynamic TDMo
Did you look at logger.conf?
From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of bilal ghayyad [EMAIL
PROTECTED]
Sent: Sunday, September 16, 2007 5:21 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] stop log/debug messages into /var
ACK
Mike Hammett wrote:
>
>
>
> -
> Mike Hammett
> Intelligent Computing Solutions
> http://www.ics-il.com
>
>
>
>
> ___
> --Bandwidth and Colocation Provided by http:
Hey Robert,
you can't imagine how much i appreciate your post, which is most a
tutorial than a post :)
Really, many thanks for your thoughts. Take for sure i will try to
implement the options you showed me here in asap.
Thank you again!
Best regards,
Jonathan GF
On 9/4/07, Robert L
Atis,
thanks for the quick post. I tried, probably wrong, to make a "simple"
macro for all local switching, but i realized it became hard to
mantain and can divert to errors in the future.
I think i will go towards your proposal. Thanks for the input :)
Jonathan GF
On 9/3/07, At
to obtain the same result but being easier to configure??
Thanks in advance.
Best regards,
Jonathan GF
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Guillermo,
el username deberia de ser igual al nombre del canal, es decir,
si [pbx1]
entonces username = pbx1
Saludos,
Jonathan GF
On 9/3/07, Guillermo Rodriguez <[EMAIL PROTECTED]> wrote:
> Hola Alex,
>
> He puesto el username y aun asi me da el fallo de autentificacion :-(
e
2 => 2,The Studio,,,saycid=yes|delete=no|tz=europe
All configuration (sip, extensions, voicemail, etc...) is available @
http://www.surestorm.com/asterisk/ for those that want to help.
Thanks in advance.
Best regards,
Jonathan GF
___
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On 29/08/07, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
> On Tue, Aug 28, 2007 at 11:42:09PM +0100, Jonathan Hunter wrote:
>
> > A kernel call trace of when my system crashes, along with copies of
> > various config files, is here: http://pastebin.ca/674091 but I have
>
system, and I don't believe it's a hardware
issue as it has always worked fine under previous versions of zaptel.
My hardware is a TDM400 with an (unused) X100P as well - see below for
output when my machine boots up.
Any suggestions gratefully received - where do I start?!
Thanks,
Jonat
that hardware for a very
small office and the further administrator do not understand so much about
Asterisk, although they can handle unix/linux boxes.
Any help would be really appreciated.
Thank in advance for you help.
Regards,
Jonathan GF
___
--Band
Thank you all for your post, i've found them quite interesting and will give
work for some time :)
Thanks again.
Cheers,
Jonathan GF
On 8/20/07, Eric Chamberlain <[EMAIL PROTECTED]> wrote:
>
> Using the phone itself as a GSM-SIP gateway is not possible with the
> native V
interest please let me know.
Probably Mitcheloc is right too, there are a lot of manners to achieve this
and the problem is mine that i don't know how to search what i want. Anyway,
thank you for your inputs. Any others will be welcomed, for sure.
Regards,
Jonathan GF
On 8/20/07, mitchel
med.
Thanks in advance.
Jonathan GF
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Working for two different VOIP providers I have had no success getting
this from upstream providers such as Level3, Time-Warner and Verizon.
-Jonathan
Baji Panchumarti wrote:
> On 8/5/07, Michael Joyner wrote:
>
>
>> ᎣᏏᏲ,
>>
>> I am looking for VOIP (SIP/IAX) pro
secret=secret
qualify=yes
trunk=yes
I have restarted both Asterisk with no luck, any clues to where should I
search for the soluton?
thanks,
Jonathan.
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Tzafrir Cohen wrote:
> On Thu, Jul 05, 2007 at 08:09:32AM +0400, Eugene Prokopiev wrote:
>
>> Hi,
>>
>> Is it possible to filter messages on asterisk console, which was started
>> with -, to see messages only for one extensions? By default there
>> are all messages for any extensions displ
If it is one of the ones I am familiar with it's only one ethernet
interface and it's literally a switch on a PCI card. The system sees one
interface and there are 4 ports out the back.
If this is the case it's not really "instead of a switch" so it will
work fine.
at can be wrong with my intallation?
J
Jonathan Unai Marquez wrote:
> Hi,
>
> I have two new Asterisk installations (1.4.4 and 1.4.5) and I have
> created rsa keys and they can now see each other as online peers:
>
> moe*CLI> iax2 show peers
> Name/Usernam
hat I cannot place calls from one Asterisk to
the other.
chan_iax2.c:7285 socket_process: I don't know how to authenticate moe to
192.168.2.201
thanks in advance,
Jonathan.
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a
am.got-name.com/?auth=:\&type=http\&number=${1}`
echo $out;
}
-Jonathan
Bill Michaelson wrote:
> Is it just me, or is the AGI interface at cnam.got-name.com failing
> for others? Anyone know how to contact them without sendi
ask...
Best wishes,
Jonathan Barratt
Openface Internet Inc.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Klaverstyn, David C
Sent: Thursday, June 21, 2007 8:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
rocess..
-Jonathan
Lutgring, Sam wrote:
Has anyone had any experience using a modem through the Asterisk
system? I have some technical support personnel that need to use a
computer modem to connect to a remote system for troubleshooting. Is
there a SIP compliant gateway that will support a
Good call Eric, thanks for the suggestion.
Voicemail user guide now temporarily available at:
http://pbx.openface.ca/vmug.htm
Yours,
Jonathan Barratt
Openface Internet Inc.
514-315-3652
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Lubow
Sent
l send the HTML version I have to you off-list.
For the hardphones, Polycom's user manuals are fairly thorough. Let me
know if you need those...
Good luck!
Jonathan Barratt
Openface Internet Inc.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ma
Which sounds like exactly what I described. Asterisk in Dom0...
-Jonathan
Adam Robins wrote:
We are running Asterisk on native CentOS. We then install VMWare on
CentOS with Windows 2003 in the VMWare partition for AD services. We
have 50+ users in a call center environment with no issues
I don't think you need to modify th Makefile at all. That might be why
you are having trouble finding details on that.
-Jonathan
Jonson Player wrote:
Hello,
I just want to put all my sip accounts in mysql and asterisk use it
from mysql. How can I do that, could you be more specific beca
Why would you want to do this?
If you wanted to run multiple systems together on an Asterisk server I
would run the Asterisk server on Dom0 and the other stuff on DomU systems.
-Jonathan
James Harper wrote:
I did it back in the xen 2.x days with a BRI adapter (Traverse NetJet).
It worked
the phone
vlan.
-Jonathan
Douglas Garstang wrote:
I have a scenario here with IP phones, on a private 192.168 network
connecting to an Asterisk box, also on the same 192.168 private
network. We’d like to have the Asterisk box also be able to send
traffic to the public IP space. For this, we wou
Looks to me like you are saving it to the Asterisk DB.
http://www.asteriskguru.com/tutorials/dbget_function.html
-Jonathan
Nitesh Divecha wrote:
Hello All,
I was wondering where does Asterisk stores the blacklist numbers?
I looked into the dialplan and it shows that it
*"Set(DB(blac
as send calls via analog to a PBX to support 12 lines you will need 24
ports. If you are only using 12 ports a channel bank may not prove to be
cost effective. If you use a channel bank then the hardware for system 1
and system 2 could be the same exact system.
-Jonathan
Jeremy Mann wrote:
I
Tzafrir Cohen wrote:
> It actually is (maintained, and a recent version of it is in
> stable/testing.
Hmm.. I think several years ago it wasn't... I guess I'm just living in
the past. Sorry about that!
--
Jon-o Addleman - http://www.redowl.ca
___
--Ban
I forgot to add that the built-in support for playing mp3s which
replaced, for some people, the mp123 program, requires asterisk-addons,
which also isn't packaged for debian! There are other possibilities
though. I think you could use mp321 plus sox to convert to the proper
sound format, for exampl
pedro noticioso wrote:
> hi there guys!
>
> how can I eliminate this message?
>
> [May 11 11:00:46] WARNING[7039]: res_musiconhold.c:506
> monmp3thread: Unable to spawn mp3player
> [May 11 11:09:06] WARNING[7039]: res_musiconhold.c:424
> spawn_mp3: Found no files in
> '/var/lib/asterisk/mohmp3'
Jon-o Addleman wrote:
> I'm using the ices command to stream a conference to an icecast server.
> This is working nicely, for the most part, but the volume is very low.
> The streamed ogg vorbis audio is much quieter than what I hear in a
SIP
> client, for example (on the same machine with the
pattern.
Just rearrange your extensions so there's no potential crossover between
your wildcards and your explicit extensions and the problem should go
away. I don't know what reason you need the _XX5 entries for so I'll
leave it up to you to find the best arrangement...
Go
Hi Joel,
6 seconds sounds suspiciously like Asterisk's dialplan timeout value.
Perhaps you have a wildcard extension that it's waiting to match
against. Either post the relevant section of dial plan or send it to me
off-list, as you prefer, and we'll see what we can find...
werDNS and MYSQL.
-Jonathan
Justin Hamade wrote:
I have run into the exact same situation and have the same question.
I did it in the dial plan manually due to time contraints but if DUNDi
or ENUM or something else is better suited I would love to know.
Also the guides and tutorial that I fou
asked us how he could have been calling the same person at
the same time as he was already calling them... Quite embarrassing. Not
sure if issue has been fixed in subsequent release or not...
Best,
Jonathan
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED
ds are
set to ulaw.
Codec: g711
Asterisk: 1.2.17
Hardware: TE412p
OS: CentOS
Kernel: 2.6.16.18
I've been investigating this issue for weeks and I'm totally out of
ideas, so any help or suggestions anyone could provide would be grea
omer doing this and there is one place they can not fax and it's
because the fax machine at the other end will not negotiate a speed
lower than 14400 baud.
-Jonathan
Matt wrote:
Very much so... we actually have a fax machine up here in our NOC
running on g711u attached to an ATA, works
ont.
# gcc --version
2.95.3
Is "__builtin_expect" part of gcc, then, rather than an external
library? (i.e. would I need to upgrade gcc in this instance)
Thank you!
Jonathan
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asterisk-
e machine is quite old, so it is possible I need to upgrade/add
something - but what?
# uname -a
Linux myserver 2.4.25 #5 Wed Jan 26 18:20:35 GMT 2005 i686 unknown
# cat /etc/slackware-version
7.0.0
If anyone can point me in the right direction, I'd appreciate it!
Cheers,
Jonathan
--
"If
|
< Return the dial status--
Someone have an idea?
Regards.
--
Jonathan Alberto Rivera Gomez
http://linuxuanl.org
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I would be willing to mirror it also….
_
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Wednesday, March 14, 2007 9:39 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] What happend to voi
another set of "real" extensions for the phones and when a user extension is
dialed it rings the phone extension.
-Jonathan
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Patrick
> Sent: Monday, March 12, 2007
n => s,n,hangup()
exten => ACC-4,1,playback(${SOUNDS}welcome-4)
exten => ACC-4,n,park(704)
exten => ACC-4,n,hangup
But the h extension is never called?
ideas?
--
==
Jonathan S.
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If it's using RBS then 56k is the right number.
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of JR Richardson
> Sent: Saturday, January 27, 2007 12:55 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] max tnt pri vo
Why don't you just give the secretary the boss' REAL extension and give a
different extension to the world that just rings the secretary?
-jonathan
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Ricardo Carv
A demonstration:
exten => _X.,1,Set(GROUP()=${CALLERID(num))
exten => _X.,n,Set(CDR(AccountCode)=${CALLERID(num))
exten => _X.,n,GotoIf($[${GROUP_COUNT(${CALLERID(num))} > 2]?103)
exten => _X.,n,Macro(trunk,${EXTEN},residential)
exten => _X.,n,Hangup
exten => _X.,1
I'm not sure if there is a more official method but Google has always
been my friend when searching the lists.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
Greene
Sent: Wednesday, December 27, 2006 12:05 PM
To: asterisk-users@lists.digiu
There is an index in the configuration file which I believe it will
obey. I'll try and find it later if you haven't found it by the time I
get to the office.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Wednesday, Decem
I don't use many of the features of astmanproxy but it does work. I use
it to capture events from several servers. Some of these are running the
1.4 beta releases.
-Jonahtan
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Tzafrir Co
0:3}
>
> --
> One day at a time, one second if that's what it takes
That works if the number is always NPA-NXX-. If you end up with
+1NPANXX or 1NPANXX then you don't have the right data.
-Jonathan
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${Number:-10:3} if I recall correctly would give you 3 characters
starting at the 10th from the end.
-Jonathan
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of John French
> Sent: Tuesday, December 19, 2006 10:35 AM
&g
I lines on "B" servers.
-Jonathan
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Pryakhin
Dimitry
Sent: Monday, December 18, 2006 8:23 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-use
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Eric "ManxPower" Wieling
> Sent: Thursday, December 14, 2006 4:30 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Hardware TDM Switching
>
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Eric "ManxPower" Wieling
> Sent: Thursday, December 14, 2006 4:30 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Hardware TDM Switching
; From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Daniel Gradecak
> Sent: Tuesday, December 12, 2006 1:06 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Asterisk Manager
>
> Hello Jonathan, thank you
sure that using the proxy, astmanproxy, you can achieve this
goal. It is recommended to use the proxy so that there is only one
connection to the server and all the other applications will connect to
the proxy.
-Jonathan
___
--Bandwidth and Colocatio
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Carla Schroder
> Sent: Monday, December 11, 2006 2:17 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] CLI History
>
> On Monday 11 December 2006 9:31 am, Douglas
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Douglas Garstang
> Sent: Monday, December 11, 2006 1:37 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [asterisk-users] CLI History
>
> > -Original M
k PBX
Max-Forwards: 70
Content-Length: 0
---
-- SIP/trainer2-08aa7af0 is busy
Any ideas/help would be highly appreciated.
Thanks,
Jonathan
--
Jonathan Palley | Idapted Inc.
[EMAIL PROTECTED]
___
--Bandwidth and Colocation provided by Easyne
I have noticed it too and do not use them anymore..
Jon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ira
Sent: Thursday, November 09, 2006 11:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Voxee lag probl
Good Morning,
I've recently gotten Asterisk installed and configured our IVR using
FreePBX. Things seem to be going well except a few of our inbound
callers are ending up in the wrong place when trying to connect to a
specific extension. The example I had this morning was someone trying to
call e
add
> features to it. I'm not the best at PHP but I can work my way around in it.
> I thought maybe freePBX allowed this with its users but I can't see where
> you can lock them down to only see information on a particular extension.
>
probably VoiceOne (http://www.voiceone.i
t; while asterisk is very capable
> of virtualhosting PBX-en on one instance.
destar (http://destar.berlios.de/) implements it, no?
and yes it is a great feature. :P
regards.
--
Jonathan Alberto Rivera Gomez
Grupo de Usuarios de GNU/Linux - UANL
http://linuxuanl.org
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