I have started to receive a lot of positive response
for the [EMAIL PROTECTED] project. For those of you
unfamiliar with this project the goal of [EMAIL PROTECTED]
is to make a full featured version of Asterisk very
easy to install.
We have created a 1 step .iso that installs RHEL
(RedHat
are in the UK, I can recommend sendmytxt.
An answer to your question: Yes, this can be done without GSM modem
connected.
Rene Kluwen
Chimit
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of B
Sent: donderdag 10 augustus 2006 19:21
To: asterisk-users
Hi,
Im looking to implement SMS callback,
Can anyone recommend any software which handles this?
Can this be done without having a gsm modem connected? (i.e. for SMS
received by GET/POST from short codes, or programatically, etc)
Thanks for sharing!
Hello list,
Was wondering if anyone knows how to get DTMF to work on voipjet..
Tried,
dtmf=rfc2833
dtmfmode=rfc2833
doesn't seem to work...
Any clues?
Cheers!
___
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asterisk-users mailing list
To
Hi all!
Ext A, B and C are SIP phones.
Ext A receives a call from Ext B. Ext A wants to transfer the call to Ext
C. Ext A puts the first call on hold, dials Ext C, then simply hangs up as
soon as the call to Ext C starts *ringing*. In other words, Ext B wants to
be sure Ext C is ringing (i.e
hi.
(B
(BI got these compile errors while install asterisk.
(Breadline and openssl are compiled using gnu source, and kernel version is 2.4.17.
(B
(BCompile errors message is follows.
(BSomeone cleared this problem?
(BPlease, help!
(B
(BRegards.
(B
(B
This error probably indicates something significant. What
(B sort of system are you compiling on? It's quite unusual for
(B "pwd" to be missing on any unix-like system. So it must not
(B be in your search path. It's almost always /bin/pwd, and if
(B you don't have /bin an
Thanks for reply.
(B
(BOf course, I had already read ML like follows.
(BThis case errors are almost same, perhaps.
(BBut, tell the truth, I can't understand What I have to do?
(BI had recompile some version automake source, and tried to recompile asterisk.
(BThe result does'nt go well.
(B
Hi.
(B
(BAt last, I can compile asterisk.
(B
(BI had compiled low version of ncurses, glibc readline termcap and so on..
(BFinary, I coud compiled asterisk.
(B
(BThanks a lot!!
(B
(B
(B
(B Thanks for reply.
(B
(B Of course, I had already read ML like follows.
(B This case errors
Hi.
(B
(BI want to connect Cisco7960 phone using IOS CME to Aasterisk VoiceMail system.
(BBut, DTMF relay is not work well, so I am able to hear VoiceMail intro and I am not
(Bable to contorl Asterisk VoceMail system.
(BDid anyone connect CME to Asterisk VoiceMail system?
(B
(BAsterisk
(B--
$B>.ED??G7(B [EMAIL PROTECTED]
(B
(B
(B
(B___
(BAsterisk-Users mailing list
(B[EMAIL PROTECTED]
(Bhttp://lists.digium.com/mailman/listinfo/asterisk-users
(BTo UNSUBSCRIBE or update options visit:
(B http://lists.digium.com/mail
I sent a few days ago right config parms that I got from BV. Try it, works
on my *.
Eugene B.
- Original Message -
From: Jay Carter [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Saturday, March 12, 2005 5:42 PM
Subject: [Asterisk-Users] Broadvoice outgoing problems
Hello
Here you can find new settings I got from BV and now works on my *. make
sure you use correct section in dial command in extensions.conf.
Eugene.
;;=
Some modifications since the last time. Now that asterisk has the secret
in
Hi,
I am new to Asterisk and wanted to ask a question concerning PSTN trunking. Is there a way to have DID's sent over IP to a switch? I know if One switch has traditional PSTN like a PRI this can be done, but is there a service provider offering this so I dont have to buy any tradtional PSTN
(B
(B
(BI've been using asterisk-0.5.0.
(B
(BI've been reading about the externip param (it
(Blooks like it is only available in the lastest releases).
(B
(BCould someone tell me the version number (or tag)
(Bto check out of CVS so I can get this functionality?
(B
(B(And, if its
Hi all,
I have been facing about the problem to know who is online with asterisk PBX.
However users wanted to see it right away, without launching any application.
As I could not find any solution with IP phones and users were really
complaining, I decided to write this little application that
Hi,
I have an SMS enabled fixed phone - Siemens S150. How can I send SMS
messages from my Asterisk to my fixed phone. I am trying as guided in
SMS command description, but all I can do is to send SMS from phone to
Asterisk, but not the other way.
Have anyone had the same problem?
Thanks,
Hoa
Hi,
I have an SMS enabled fixed phone - Siemens S150. How can I send SMS
messages from my Asterisk to my fixed phone. I am trying as guided in
SMS command description, but all I can do is to send SMS from phone to
Asterisk, but not the other way.
Have anyone had the same problem?
Thanks,
Hoa
My intention is to setup Asterisk to be a message center to receive
from and send SMS to fixed phones. Can it be possible? My fixed phone
can dial to Asterisk and send SMS to Asterisk, but I cannot setup the
other way: make Asterisk dial to fixed phone and send SMS to fixed
phone.
On Sat, 04 Dec
Hello,
I am trying to exchange SMS between a fixed phone and an Asterisk. The
intention is to make the Asterisk become a SMS Center, because we do
not have public SMS Center in our country.
I have two phone lines, one for Asterisk and one for the SMS enabled
fixned phone. I also config the fixed
Hello,
I have some issues with Asterisk, and I see them fixed in bug
database. It is also said the fix is updated in CVS. However, when I
checkout the most current release from CVS, I cannot find the
modification as described in the difference file in the bug database.
How can I apply the
Can you try to add some parameters to zapata.conf before defining channel 3,4
cidsignalling = dtmf
cidstart = polarity
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Hello,
I am trying to make my Asterisk to recognize CallerID from incoming
PSTN calls. I am using a TDM400. CallerID in my country's PSTN is
based on DTMF. Some information from my configuration files:
/etc/zaptel.conf:
loadzone=nl
defaultzone=nl
/etc/asterisk/zapata.conf
usecallerid=yes
hey people,
i've installed an asterisk version on my Linux Distribution.
Then i wanted to compile the packages for H323 support.
The openh323 and pwlib libraries compiled without any error. But at least i
got one in /~/asterisk-oh323-0.5-10/asterisk-driver/chan_oh323.c in Line
1128. Too many
Hello,
i have a Winmodem (Softwaremodem) i know, this is a problem under linux.
But asterisk loaded a Modem channel.
What i wanted to know is, can i use this channel to make a PSTN call??
If yes, how can i do that. Which *.conf files do i have to change??
Has someone experience with that??
Kindly
Hi folks,
I am using opengk to handle h323 calls.
* and my clients register at opengk successfully.
But everyone can register to my gk??
Is there a way to restrict the clients by using the authorisation of
h323.conf ??
Cheers,
Harald
___
Asterisk-Users
If you are moving from 1.2.x to 1.4.x then you may need to update a bit of
your dial plan. If not you just needs to install the new version of asterisk
and remove the modules from the old version and you should be good to go.
Also I personally back up all my config filed just in case.
-
Can it be that asterisk does not have permission to copy the file over ? Also
check your date settings on the server.
- Original Message -
From: Asterisk guy
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Sunday, July 22, 2007 5:29 PM
Subject:
are you trying to connect to localhost ? try 127.0.0.1 as well as the
machines IP and or host name.
- Original Message -
From: NicklasZ [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, July 23, 2007 11:40 AM
Subject: [asterisk-users] ODBC Connection failed
Hi,
im
try priorityjumping=yes in extensions.conf
- Original Message -
From: Michael J. Liberatore [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, July 23, 2007 11:21 PM
Subject: Re: [asterisk-users] Upgrade Procedure
I was able to get H.323 to work with G729 on 1.2.18. It works real well.
- Original Message -
From: Cesc Santa [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, July 24, 2007 1:54 AM
Subject: [asterisk-users]
you can set in sip.conf the moh to an invalid class or you can create an mp3
that is blank
- Original Message -
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, August 15, 2007 3:11 PM
Subject: [asterisk-users] Disable MoH for certain phones
Hi,
Is it
Hi,
I am trying to run an Asterisk (1.4.10) server on Ubuntu Linux Fiesty Fawn. The
server is behind NAT. I am testing SIP with the X-Lite client from xten. The
client is also behind NAT.
I realize that this is amongst the worst configurations, but I have been made
to believe that it can
B wrote:
Hi,
I realize that this is amongst the worst configurations, but I have been
made to believe that it can work... eventually. However, currently SIP
call set up seems to go fine, but no media is transferred in either
direction. For example, the following is output
Does anyone know if it is possible to change the packetization time in Asterisk
? I was told by a client of mine that adjusting this with using G729 can
greatly lower the amount of bandwidth used.
___
--Bandwidth and Colocation Provided by
Thank you very much for your prompt replies. Perhaps I will consider moving to
a 1.2 version of Asterisk.
Date: Sun, 19 Aug 2007 12:08:36 +0100
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk and Client NAT
On Sun, 19 Aug 2007, G B
- Original Message -
From: Dan Austin [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, August 19, 2007 7:58 PM
Subject: Re: [asterisk-users] Change Packetization Time
Dovid wrote:
Does anyone know if it
Hi List,
I have a client who has a TDM400P with 4 FXO. He has a problem them when some
one calls, then hangs up it takes a good 10-15 seconds or more of the card to
realize that the line was hung up on. The phones keep reigning After a bit it
hangs up on the line. Also there has been some
snip
Until 1.4 improves, I'm staying with 1.2
/snip
Ditto
___
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
What is happening ? Please email us the SIP Debug. Also with paging most phones
require a special SIP header for the phone to know that it has to pick up right
away.
- Original Message -
From: Stuart J. Newman
To: asterisk-users@lists.digium.com
Sent: Monday, August 13, 2007
- Original Message -
From: Jody Gugelhupf [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, August 27, 2007 3:55 PM
Subject: [asterisk-users] voip provider settings problem, please help
hi ppl, i'm using asterisk 1.2 because i'm making use of voiceone, but
before i
B or anything like that. I want to use realtime for
queues,voicemail, sippeers and extensions. The only issue that I have
come up with so far is that a common voicemail table would cause each
box to try and send out mwi indicators since it appears each * box pulls
all of the voicemail boxes
snip
I am running an SC1435 with two dual core Opteron 2212, four gigs of RAM
and a couple 250gig SATA drives. Totally VoIP so I cannot comment on
cards or interrupts, but so far it has been flawless.
I would like to see how many G729/ULAW conversions it could handle. How
would I go about
Ditto. Would you complain if some one gave you a free flight that it wasn't
first class ? Asterisk is free Stop the moaning
Enough The Digium/Aseterisk bashing seems to be at an all time high
recently. You seem to be involved in a lot of it. Russell has given most of
snip
Digium has done this, for me, as well.
However, in either case, I have reservations about letting others wack
away at my machines, especially if one cannot see what they are doing.
No so much not trusting them, but not learning a thing along the way.
When I voiced that concern to the Digium
into the plane and hijack it, so the airline releases nothing
more than a security upgrade for it. Now, however, we find that after
flying for 2 hours, the wheels fall off the plane.
On 8/30/07, Dovid B [EMAIL PROTECTED] wrote:
Ditto. Would you complain if some one gave you a free flight
- Original Message -
From: Jay R. Ashworth [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, August 30, 2007 3:57 PM
Subject: Re: [asterisk-users] where is 1.4.12?
On Wed, Aug 29, 2007 at 09:48:18PM -0400, Matt wrote:
I guess that's my point. I realize asterisk
- Original Message -
From: Jared Smith [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, August 30, 2007 5:41 PM
Subject: Re: [asterisk-users] where is 1.4.12?
On Thu, 2007-08-30 at 08:02 -0500, Eric
Are you using 1.4.X on one and 1.2.X on another ?
- Original Message -
From: Jerry Geis [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, August 02, 2007 3:34 PM
Subject: [asterisk-users] problem with rfc2833
I have the following:
pri box incoming/outgoing on box
How about sending a SipHeader to the second box and then on the second box look
for the header. If the header does not exist then ring the extension normally.
If the header is there than send back congestion (basically have a gotoif
before it hits the Exten = Foo,1,Voicemail)
- Original
You may want to consider upgrading your version of asterisk. Next you can try
using SER + Asterisk + Heartbeat.
- Original Message -
From: Khaled Chehab
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Cc: [EMAIL PROTECTED]
Sent: Tuesday, August 21, 2007 3:05 PM
snip
question2: it's possible read registration data from astdb from python/php
(or it is possible write sip registrations to mysql/sqlite? i do not
want realtime because of NAT issues)
/snip
Marek,
What NAT issues can realtime create that there won't be in static ?
- Original Message -
From: Steve Totaro [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, August 28, 2007 11:51 AM
Subject: Re: [asterisk-users] OT: DELL Platforms
Dovid B wrote:
snip
I am running an SC1435
- Original Message -
From: Eric ManxPower Wieling [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
Sent: Wednesday, August 22, 2007 4:08 PM
Subject: Re: [asterisk-users] Polycom behind NAT won't register to
- Original Message -
From: Adrian Marsh [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, August 30, 2007 2:34 PM
Subject: [asterisk-users] How to handle + prefix
Hi,
How can I have A*k convert a call from
It seems that the other end is having an issue authenticating you. I have
seen lots of switches act up with asterisk if you don't tweak you settings in
sip.conf just right. Do a SIP debug and have a look at te INVITE request.
- Original Message -
From: Ondrej Polívka
To:
- Original Message -
From: Steve Prior [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, August 22, 2007 7:13 PM
Subject: Re: [asterisk-users] phone as control interface (was 99 bottles of
beer)
Steve
Dan,
I sent this particular's traffic through a 1.4.X box to our 1.2.X box. Worked
like a charm.
Thanks for the help.
Dovid
- Original Message -
From: Dan Austin
To: Dovid B
Sent: Monday, August 20, 2007 9:22 AM
Subject: RE: [asterisk-users] Change Packetization Time
snip
How then does a users phone dials this. I have never seen a phone with +
on the keypad, nor have I ever seen dail plan logic in a phone that
could correctly handle the variable length issue of international
numbers in order to do a rewrite and send the + in front.
/snip
Then you must
- Original Message -
From: Steve Totaro [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, August 31, 2007 10:43 PM
Subject: Re: [asterisk-users] OT: DELL Platforms
Dovid B wrote:
- Original Message
, 2007 9:14 AM
Subject: Re: [asterisk-users] OT: DELL Platforms
On 9/1/07, Dovid B [EMAIL PROTECTED] wrote:
Why work with two separate devices when you can have one ? And yes the DC is
staffed 24/7 but do you want to call them every time you need a new CD/DVD
inserted in to the box
Sorry, I just realized that my lspci output was not verbose enough. Here is the
update:
# sudo lspci -vv
00:00.0 Host bridge: VIA Technologies, Inc. VT8374 P4X400 Host Controller/AGP
Bridge (rev 03)
Subsystem: Elitegroup Computer Systems Unknown device 0a82
Control: I/O- Mem+
Date: Fri, 7 Sep 2007 07:13:38 -0400
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] New Installed X100p
G B wrote:
Hi,
I just installed an X100p knockoff (OpenVox a100lp). I have seen my
errors in the mailing list but without resolution
limit at which moderator
approval is required. Thanks for the help.
Date: Fri, 7 Sep 2007 10:26:11 +0300
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] New Installed X100p
On Fri, Sep 07, 2007 at 12:02:29AM -0700, G B wrote:
Hi,
I just
I apologize, it seems my cutting and pasting is off. The final (and correct
output) of lspci -vv:
00:00.0 Host bridge: VIA Technologies, Inc. VT8374 P4X400 Host Controller/AGP
Bridge (rev 03)
Subsystem: Elitegroup Computer Systems Unknown device 0a82
Control: I/O- Mem+
: VIA Technologies, Inc. VT8235 ISA Bridge
Subsystem: Elitegroup Computer Systems Unknown device 0a82
Flags: bus master, stepping, medium devsel, latency 0
Capabilities: access denied
00:11.1 IDE interface: VIA Technologies, Inc.
VT82C586A/B/VT82C686/A/B/VT823x/A/C PIPC Bus Master IDE
...
This must be proof that I have purchased a real piece of @#$.
Thanks for all of your help.
Date: Sat, 8 Sep 2007 02:41:50 +0300
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] New Installed X100p
On Fri, Sep 07, 2007 at 08:45:11AM -0700, G B wrote
I just want to add that it is the best way to learn. Till today I thank
those on the list that told me to stay away from GUI's and learn the real
asterisk.
If you still can't figure out the difference I can help you out but it is
better if you learn on your own.
- Original Message -
There is a Biz list for a reason. Please look at the emails headers
Non-Commercial Discussion
- Original Message -
From: Claude Cunningham [EMAIL PROTECTED]
To: Commercial and Business-Oriented Asterisk Discussion
[EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial
Eric,
Try 5 polycoms behind the same NAT router. Let me know when you grab a drink
;)
- Original Message -
From: Eric ManxPower Wieling [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, September 12, 2007 2:43
- Original Message -
From: Jason Martin [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, September 06, 2007 4:58 PM
Subject: [asterisk-users] SIP Debugging to separate log file
Hello, I'm working with our SIP provider to nail down some call quality
issues
we're
Hi List,
I have a client that has an interesting request. He wants to have people call
in to a conference room and all be able to talk however they should not hear
each other. There should be admin access to for one user to call in and be able
to listen in to everyone that is talking (they may
: Wednesday, September 12, 2007 3:18 PM
Subject: Re: [asterisk-users] Linux-HA and Asterisk
How about 20+ on a Qwest DSL modem hitting our server? Works great.
On Sep 12, 2007, at 7:23 AM, Dovid B wrote:
Eric,
Try 5 polycoms behind the same NAT router. Let me know when you
grab a drink
Did you set externip= ?
- Original Message -
From: Christian [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, September 12, 2007 3:23 PM
Subject: [asterisk-users] Problems with Asterisk behind a firewall
Hi
CF,
I think the best idea is to wait till it comes out and run etheral. This
would be great for when I travel out of the US. (Yes I can use my network)
but why not use some else's ant not pay ;) Also I wonder if Hotel's out of
the US will start putting these up for their guests.
- Original
Hi,
I'm a complete newbie to Asterisk and have been reading through
documentation and sites for the last couple of hours trying to understand
what to do to start my learning curve with Asterisk, and am very confused.
For starters, what is the difference btwn the 1.2 and 1.4 branches of
Bob Pierce [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
On Thu, 2007-09-27 at 11:48 -0400, Eric B. wrote:
I'm a complete newbie to Asterisk and have been reading through
documentation and sites for the last couple of hours trying to
understand what to do to start my learning
Jim Canfield [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
Eric B. wrote:
site and got to chapter 4 or 5 and decided to take a break. Which is when
I
found AsteriskNow and TriBox and then started wondering if it was really
necessary / worthwhile to figure out all
done ?
Quoting Dovid B [EMAIL PROTECTED]:
Hi List,
I have a client that has an interesting request. He wants to have
people call in to a conference room and all be able to talk however
they should not hear each other. There should be admin access to for
one user to call in and be able
I have used nufone in the past and the quality is good.
- Original Message -
From: Alejandro Lengua
To: Commercial and Business-Oriented Asterisk Discussion ; Asterisk Users
Mailing List - Non-Commercial Discussion
Sent: Saturday, September 22, 2007 7:48 PM
Subject:
Have a look here:
http://www.voip-info.org/wiki-Asterisk+config+features.conf .Specifically at
applicationfaturemap.
- Original Message -
From: Atis Lezdins [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, October 08, 2007 2:10 PM
Subject: Re: [asterisk-users]
This is the first time that I am seeing this error. Can anyone help me with its
meaning ?
pbx.c:5939 pbx_builtin_serialize_variables: Data Buffer Size Exceeded!
Thanks.
Dovid___
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www.telephonydepot.com has good prices. Never needed their support so I
can't comment
www.voipsupply.com a bit more expensive than above. Great support
- Original Message -
From: bilal ghayyad [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, October 11, 2007 12:08
Try sip show channels from the CLI
- Original Message -
From: Scott Moseman [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, October 12, 2007 6:12 PM
Subject: [asterisk-users] Display channels and codecs
Is there an easy way to show all active channels AND the codecs
owfs and
other open source software to use them.
On Fri, 31 Aug 2007, Dovid B wrote:
I am new to the whole controlling devices in your home from asterisk.
Can
anyone give me a URL to devices that I can connect to my box that can
then
connect to the lights, security system, TV etc
Chanisavail does not work well for this. I would use priority jumping
(n+101).
- Original Message -
From: Alex Balashov [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion asterisk-users@lists.digium.com
Sent: Monday, October 15, 2007 6:47
None. Asterisk vanilla is the best IMHO.
- Original Message -
From: Anciso, Roy
To: asterisk-users@lists.digium.com
Sent: Monday, October 15, 2007 7:28 PM
Subject: [asterisk-users] What web GUI are people happy with?
Just wondering what web GUI people like for asterisk. I
with?
I use vi. Not sure if it has a web interface yet.
PaulH
On Tue, 2007-10-16 at 00:51 +0200, Dovid B wrote:
None. Asterisk vanilla is the best IMHO.
- Original Message -
From: Anciso, Roy
To: asterisk-users@lists.digium.com
Sent: Monday, October 15
- Original Message -
From: Mojo with Horan Company, LLC [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, October 16, 2007 1:20 AM
Subject: Re: [asterisk-users] PSTN failover
Dovid B wrote:
Chanisavail does
- Original Message -
From: Alex Balashov [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, October 16, 2007 1:12 AM
Subject: Re: [asterisk-users] PSTN failover
On Tue, 16 Oct 2007, Dovid B wrote:
Chanisavail
- Original Message -
From: John Faubion [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, October 16, 2007 4:11 AM
Subject: Re: [asterisk-users] phone as control interface (was99bottlesof
beer)
Does anyone
- Original Message -
From: John Faubion [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, October 16, 2007 9:46 AM
Subject: Re: [asterisk-users] phone as control interface(was99bottlesof
beer)
has anyone
What was the tech prefix ?
- Original Message -
From: Jon Weisman [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, October 16, 2007 8:27 PM
Subject: [asterisk-users] tech prefix
There used to be a prefix
- Original Message -
From: Atis Lezdins [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, October 17, 2007 5:18 PM
Subject: [asterisk-users] Play sound on hangup
Hi,
Does anybody have some ideas - how to play a sound file on channel, after
that
bridged
Kevin,
What kind of device are you using on the fridge ?
Dovid
- Original Message -
From: Kevin Withnall
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Thursday, October 18, 2007 12:04 AM
Subject: Re: [asterisk-users] Refrigerator Alarms
We use similar
Any chance that your dtmf is not set up correctly ?
- Original Message -
From: Michael Munger
To: asterisk-users@lists.digium.com
Sent: Tuesday, October 16, 2007 10:30 PM
Subject: [asterisk-users] Background not listening?
This ridiculously simple IVR is not listening to
While I am a fan of CentOS some pople just take it tooo far.
- Original Message -
From: Perssy Llamosas [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, October 18, 2007 9:54 PM
Subject: Re:
I know I can get free DID's with SIP, is anyone giving out free DID's with
IAX?
Thanks in advance,
In the words of the great jbot in the #asterisk channel on irc.freenode.net
Dovid ~ygwypf
jbot well, ygwypf is You Get What You Pay For. If the sole factor in your
decision to purchase a
- Original Message -
From: Don Pobanz [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, October 18, 2007 3:38 PM
Subject: [asterisk-users] Limit number of times a call can be forwarded
We have had a few
Hi List,
I am using a Plantronics CS50 head set with my Polycom 601. I use the button on
it to pick up calls. Is there any way to have the phone set up that if I pick
up with the button on the headset that it sends the call to the headset and
that I don't have to press the headset button on the
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