On Monday 25 April 2005 3:05 pm, Daniel Salama wrote:
I'd like to create a dial rule that when someone tries to dial a
particular number, the same number is dialed, except that prefixed with
some additional digit(s). How can this be specified on extensions.conf?
exten =
: [Asterisk-Users] Dial Plan - How to prepend a digit
I'd like to create a dial rule that when someone tries to dial a
particular number, the same number is dialed, except that prefixed with
some additional digit(s). How can this be specified on extensions.conf?
Thanks,
Daniel
Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, April 25, 2005 4:05 PM
Subject: [Asterisk-Users] Dial Plan - How to prepend a digit
I'd like to create a dial rule that when someone tries to dial a
particular number, the same number is dialed, except that prefixed
Do you mean like this?
This is an example
Here I have 3 trunks (2 SIP and 1 IAX)
To dial out in rix, i'll have to dial a 0 before areaprefix, which
here always begins with another 0 (040xx)
I.e to dial that area on trunk rix I'll have to dial 0040xxx = (00Z.)
EXTEN:1 will remove one
Title: Dial While on IVR
While the call is going into the IVR how can I Dial an extension and get immediately connected interrupting the IVR?
Robson
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Hi all
Could someone please care to share an example of the Dial W option
usage. I cannot seem to find any reference to it usage. I know you use
*1 in features.conf to start the monitor, but from there I am lost.
Master
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I forgot the command to have asterisk dial and hangup from the console.
Thanks everyone
-
Dan Levine
CYTEXONE | Your Technology Specialists
t: 877.CYTEXONE x 810
l: 212.477.0990 x 810
e: [EMAIL PROTECTED]
http://www.cytexone.com
___
On Monday 18 April 2005 5:56 pm, Dan Levine wrote:
I forgot the command to have asterisk dial and hangup from the console.
dial
hangup
(try 'help' from the CLI)
-josiah
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Dan Levine wrote:
I forgot the command to have asterisk dial and hangup from the console.
exactly as you said!
(help dial help hangup will show you the syntax)
bye
Ronald
Thanks everyone
-
Dan Levine
CYTEXONE | Your Technology Specialists
t: 877.CYTEXONE x 810
l: 212.477.0990 x 810
Sent: Monday, April 18, 2005 6:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] DIAL FROM CONSOLE
Dan Levine wrote:
I forgot the command to have asterisk dial and hangup from the console.
exactly as you said!
(help dial help hangup
Wiplinger
Sent: Monday, April 18, 2005 6:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] DIAL FROM CONSOLE
Dan Levine wrote:
I forgot the command to have asterisk dial and hangup from the console.
exactly as you said!
(help dial help hangup
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Dial Macro Arguments
the feature you are talking about is still not commited to stable. at
the moment it is only availabe in CVS HEAD. You can try to download
the patch and apply it, however I did
I have just
inherited a Asterisk box which is configured as follows.
10 internal
Sip Phones
3Pots
Lines
1 voip
provider (SIP)
Call come in
over the pots lines however Outbound goes out thru the VOIP
provider.
However
looking at the configs I cannot figure out what controls how call
-users@lists.digium.comSubject: [Asterisk-Users] dial
plan
I have just
inherited a Asterisk box which is configured as follows.
10 internal
Sip Phones
3Pots
Lines
1 voip
provider (SIP)
Call come in
over the pots lines however Outbound goes out thru the VOIP
provider.
However
looking
On April 14, 2005 05:48 pm, Michael Di Martino wrote:
Call come in over the pots lines however Outbound goes out thru the VOIP
provider.
However looking at the configs I cannot figure out what controls how
call are sent out.
In other words where in the config files does it determine that all
Hello all! I posted a message a while back about a problem I was having
in December. I was unable to send arguments to the macro in the dial
command. I was told back then to use ^ as the delimiter between the macro
name and the arguments and that I had to upgrade to a newer version of
Asterisk.
the feature you are talking about is still not commited to stable. at
the moment it is only availabe in CVS HEAD. You can try to download
the patch and apply it, however I did not succeed in applying it to
1.0.7 so I had to use HEAD.
On 4/14/05, Shaun Tierney [EMAIL PROTECTED] wrote:
Hello all!
J. Arnaud wrote:
Hi,
I am using the dial out feature
(/var/spool/asterisk/outgoing) but when I look in
CDRs,
calls that reached a all circuits are busy now,
please call later are considered as ANSWERED.
Is it the expected behavior? It there a way to change
that?
If you have analog calls are
The Tiki has a sample of screening: the Dial command can specify a
macro that should talk to the answering channel to help Asterisk
decide whether to bridge the two channels or do something else.
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd+dial
On Thu, 31 Mar 2005 09:16:21
Phil, thanks - I went down that road and wasted an evening before realizing
that it wasn't in the stable version..
I'll probably do a rebuild with cvs-head and see if that introduces any
other problems.
Thanks - Joe
The Tiki has a sample of screening: the Dial command can specify a
macro
My extensions are going to dial out to multiple locations, where machines
may answer the phone instead of the called party. As such, I would like
asterisk to prompt the called party to provide acknowledgement by dialing a
digit before asterisk connects the call.
I saw that there is a bug entry
On Thu, 31 Mar 2005, Joe Presto wrote:
My extensions are going to dial out to multiple locations, where machines
may answer the phone instead of the called party. As such, I would like
asterisk to prompt the called party to provide acknowledgement by dialing a
digit before asterisk connects
be my only option.
Joe
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Peter Svensson
Sent: Thursday, March 31, 2005 1:48 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] dial cmd - called
On Thu, 31 Mar 2005, Joe Presto wrote:
Peter, thanks. This would be a less than optimal solution for me, as I
wouldn't be able to pass the caller id of the orig caller (which I could do
via IAX), nor would I be able to announce the caller ID after the call so I
could prescreen whether to
Hello,
I just setup the Asterisk to integrate with Panasonic legacy PBX. Config as followings,
PSTN -- PanasonicPBX--TDM400P(FXO)--AsteriskPC-- Internet
* is for AA / Voicemail and VOIP in/out
Currently the AA / Voicemail function for incoming PSTN callsare working well.
My problem is
Hi,
I've managed to get my asterisk server up and running with a single POTS
line and a polycom IP500.
It will happily answer the phone line, tranfer calls, voicemail, etc.
The problem comes when I pick up the polycom phone and want to place an
outside call.
If I dial 913237773456 it just
Is it possible to initiate/receive calls from a url (that is without
having to install and configure a PC soft phone) using asterisk?
If yes, may I please get some sites, pointers, HOWTOs on how its done?
You can also try the Flash Operator Panel, http://www.asternic.org. It
supports
Hello,
Is it possible to initiate/receive calls from a url (that is without
having to install and configure a PC soft phone) using asterisk?
If yes, may I please get some sites, pointers, HOWTOs on how its done?
Thanks,
Julius.
___
Asterisk-Users
Julius Kidubuka wrote:
Is it possible to initiate/receive calls from a url (that is without
having to install and configure a PC soft phone) using asterisk?
If yes, may I please get some sites, pointers, HOWTOs on how its done?
I think you need asterisk call manager, that can initiate calls for
. März 2005 22:31
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Dial from a URL - Possible?
Julius Kidubuka wrote:
Is it possible to initiate/receive calls from a url (that
is without
having to install and configure a PC
Julius Kidubuka wrote:
Is it possible to initiate/receive calls from a url (that
is without
having to install and configure a PC soft phone) using asterisk? If
yes, may I please get some sites, pointers, HOWTOs on how its done?
I do this from my webbased CRM/groupware app.
I
On Wed, 16 Mar 2005 19:58:55 -0800, Luki [EMAIL PROTECTED] wrote:
Anyway, if anyone ever needs this info, they can Google it now :-).
Might be a good thing for the wiki too. ;)
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Hi everyone,
I'm wondering I would accomplish the following: I want to dial several
SIP extensions simultaneously, HOWEVER, for different times (say ext
10 for 15 sec and ext 11 for 30 sec), and potentially with different
headers (such as ALERT_INFO) and codecs for each extension. Obviously
: Wednesday, March 16, 2005 5:23 PM
Subject: [Asterisk-Users] Dial multiple extensions,but different
variables/timeouts
Hi everyone,
I'm wondering I would accomplish the following: I want to dial several
SIP extensions simultaneously, HOWEVER, for different times (say ext
10 for 15 sec and ext 11
try the local channel.
the local channel allows you to have:
[default]
exten = 123,1,Dial(${DEVICE1},30,tr)
exten = 124,1,Dial(${DEVICE2},45,tr)
exten = 125,1,Dial(Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED])
that will go thru the dial plan of 123 and 124. However, when I tested
it for what I
-
From: Luki [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, March 16, 2005 5:23 PM
Subject: [Asterisk-Users] Dial multiple extensions,but different
variables/timeouts
Hi everyone,
I'm wondering I would accomplish the following: I want to dial several
SIP extensions
Huh, that sounds interesting. I never knew what the local channels
were for. I will give it a try. At least I know where to start now...
thanks C F.
--Luki
On Wed, 16 Mar 2005 20:01:32 -0500, C F [EMAIL PROTECTED] wrote:
try the local channel.
the local channel allows you to have:
OK, great... the local forking approach works great. Example:
[extensions]
exten = 10,1,Dial(Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED])
[test]
exten = 11,1,SetVar(_SIP_CODEC=g726)
exten = 11,2,SetVar(_ALERT_INFO=Bellcore-r6)
exten = 11,3,Dial(SIP/11,10)
exten = 12,1,SetVar(_SIP_CODEC=ulaw)
hello
i want to rout my calls to h.323. i have registered my
asterisk with GnuGatekeeper. but it is not routing my
call to h.323 channel. he is saying Internal channel
initialization failed. Bad binary?
can any one check my settings what is problem here
thanks in advance
kamran
hallo,
i trying to dial with a python script via the manager interface, it works
ok but i would like to send a soud file name as a variable to the dialplan,
so that i can call a number and send it a different soundfile i choose in
my pyton script.
the problem is, that the * dials correct and
Hi,
I just installed TDM22B on Asterisk, Dialing in and
out using regular anonog phones works, However, how do
I configure my sip softphones to dial out using the
Zap channels and enbling dial between my sip phones
and analog phone interanally.
Thanks
Could you do something with the h (Calling party Hangup)
eg
exten = h,1,DoSomething
On Sun, 6 Mar 2005 15:00:23 -0500, George Burt [EMAIL PROTECTED] wrote:
I am trying to run a macro at the beginning of call and after the call is
terminated.
exten =
Hello all!
I'd like my * to call out to somebody and offer the called party
a menu system. Everything should just be as if the called party
had initiated the call themselves.
This is my try:
exten = 100,1,Dial(Modem/g1:0555321)
exten = 100,2,Goto(mainmenu,s,1)
This doesn't really work, because
] On Behalf Of Raoul
Bönisch
Sent: Tuesday, March 08, 2005 9:13 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Dial() out and offer a menu system
Hello all!
I'd like my * to call out to somebody and offer the called party
a menu system. Everything should just be as if the called
I want Asterisk to do the following:
- call a voicemail system by dialing a number and
playing a DTMF tone
- record what is said by the called party and save
the recording to a file
- end the recording when a particular phrase is
said by the called party
- put that recording into an Asterisk
I am interested in using the M(x) option on the Dial command to run a macro
upon connection of a call.
I am using the lastest stable release. The wiki indicates that improvements
have been made for the 1.1 version (sending parameters delimited with ^).
Does M(x) work at all with the current
I am trying to run a macro at the beginning of call and after the call is
terminated.
exten = 33,1,Macro(makeOnJS,${EXTEN},${CALLERIDNUM},${DATETIME})
exten = 33,2,Dial(SIP/33,15,tg)
exten = 33,3,NoOp(makeOffJS*${EXTEN}*${CALLERIDNUM}*${DATETIME})
exten =
hi all
i am using CVS with Realtime mysql on backend. Dial
application is invoked again and again what is the
reason. I have tested it by printing some message to
debug. this application is invoked again and again
here is debug you can see lot of messages from
app_dial.c at the end. Any one tell
Hello list,
Is it possible to implement an application that satisfies the following
scenario using agi and php?
- user picks up phone
- he wants to redirect all his calls to the cellphone
- he dials *400 for example and all the calls addressed to him are
diverted
- he comes back to office next
Look at [EMAIL PROTECTED]
http://asteriskathome.sourceforge.net/
Vertical Service Codes are built in the dialplan:
*78/79 Do Not Disturb
*70/71 Call Waiting
*72/73 Call Forward This is what you are possibly looking for.
*90/91 Call Forward Busy
*69 Call Trace
Of course, you can build just
On Saturday February 26 2005 4:45 pm, John Millican wrote:
On Saturday February 26 2005 4:30 pm, Chris Ford wrote:
I tried to call you number to see what I would get and you have a verizon
Voice messaging service.
if you called the 6037862111 that is a voicemail number tyhat i was calling
see bottom
- Original Message -
From: John Millican [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, February 28, 2005 10:21 AM
Subject: Re: [Asterisk-Users] Dial out through Broadvoice
On Saturday February 26
On Monday February 28 2005 1:17 pm, Roger Hanson wrote:
see bottom
snip
Hello all,
When I call the Broadvoice number all is good.
When I try to call out through DISA on my broadvoice line i get
the
following:
Executing Dial(SIP/147.135.0.129-0815bc60,
SIP/[EMAIL
Am i not providing some helpfull info? If not tell me
what i am missing and i will get it. I am sure I have missed somethins but i
do not know what/ I greatly apreciate all the help so far.
John Millican
The service might just be down. I was up and working just fine and a
few hours ago
- Original Message -
From: Gabriel Gunderson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
Sent: Monday, February 28, 2005 4:49 PM
Subject: Re: [Asterisk-Users] Dial out through Broadvoice
Am i
: [Asterisk-Users] Dial out through Broadvoice
Am i not providing some helpfull info? If not tell me
what i am missing and i will get it. I am sure I have missed
somethins but i
do not know what/ I greatly apreciate all the help so far.
John Millican
The service might just be down. I
snip
So, what exactly is happening again? You can rx calls but not tx calls
over Broadvoice? Correct?
Can you rx calls over any other VoIP provider or PSTN?
Could you post your current configs again?
I was unable to tx could rx all day no problem i was getting an error:
Hello all,
When I call the Broadvoice number all is good.
When I try to call out through DISA on my broadvoice line i get the following:
Executing Dial(SIP/147.135.0.129-0815bc60,
SIP/[EMAIL PROTECTED]|30) in new stack
-- Called [EMAIL PROTECTED]
-- Got SIP response 480 Temporarily Not
Hello all,
When I call the Broadvoice number all is good.
When I try to call out through DISA on my broadvoice line i get the
following:
Executing Dial(SIP/147.135.0.129-0815bc60,
SIP/[EMAIL PROTECTED]|30) in new stack
-- Called [EMAIL PROTECTED]
-- Got SIP response 480
-Users] Dial out through Broadvoice
Hello all,
When I call the Broadvoice number all is good.
When I try to call out through DISA on my broadvoice line i get the
following:
Executing Dial(SIP/147.135.0.129-0815bc60,
SIP/[EMAIL PROTECTED]|30) in new stack
-- Called [EMAIL PROTECTED]
-- Got SIP
On Saturday February 26 2005 4:30 pm, Chris Ford wrote:
I tried to call you number to see what I would get and you have a verizon
Voice messaging service.
if you called the 6037862111 that is a voicemail number tyhat i was calling to
test knowing it would not be busy and would not bother
Hi ALL;
I saw several examples of "Dial" app with the
format:
Dial(Local/..)
Anybody knows what the "Local" technology
means?
Regards
Mohamamd
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On Wed, 16 Feb 2005 03:40:38 +0330, mohammad [EMAIL PROTECTED] wrote:
I saw several examples of Dial app with the format:
Dial(Local/..)
Anybody knows what the Local technology means?
Did you try the WiKi? Or Google?
http://www.google.com/search?q=asterisk+local
--
Peter
: February 16, 2005 12:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Dial (Local/.)
On Wed, 16 Feb 2005 03:40:38 +0330, mohammad [EMAIL PROTECTED] wrote:
I saw several examples of Dial app with the format:
Dial(Local
Are you paying me? Did I ask you to do this? Did you get permission
from all 10,000 to do this?
On Wed, 16 Feb 2005 13:40:41 -, Bill Seddon
[EMAIL PROTECTED] wrote:
Mondial Software Limited
020 7043 2795
www.mondialsoftware.com
Click here to view our presentation of Cash Controller
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Can the Dial() command tell the difference between busy and congestion?
At the moment it seems to be treating them both the same on my server. I
want to route the calls out via a SIP gateway unless that is congested, in
which case dial out through
Hi!
Can the Dial() command tell the difference between busy and congestion?
At the moment it seems to be treating them both the same on my server.
With bristuff 0.1.0 and later a patch to Dial() is included as follows:
app_dial modification (jumps to +201 if channel is unavailable)
Apart
Philipp von Klitzing wrote:
Hi!
Can the Dial() command tell the difference between busy and congestion?
At the moment it seems to be treating them both the same on my server.
Yes, but not on analog ports.
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Does anyone know why this is not working?
exten=s,1,Dial(SIP/192.168.1.8:,20);Connectto192.168.1.8onport,witha20sectimeout.
exten=s,1,Dial(SIP/[EMAIL PROTECTED]:9876,20,r);Connecttosip.comport9876,requestingextension8500.
I defined a sip peer
called sip-gateway. If I dial
On Thu, 10 Feb 2005 16:33:46 -0500, Gene Willingham
[EMAIL PROTECTED] wrote:
exten = s,1,Dial(SIP/192.168.1.8:,20); Connect to 192.168.1.8 on
port , with a 20 sec timeout.
exten = s,1,Dial(SIP/[EMAIL PROTECTED]:9876,20,r) ; Connect to sip.com
port 9876, requesting extension
For a while now in my call center I've been seeing calls that come in,
hit an agent who is DND, and then bounce to the next agent, but instead
of ringing for 20 seconds thing ring for about one ring and then go back
to hold then cycle back through making it difficult to pick up the call.
Then
Hello,
Here is the dial command:
exten = 790,2,Dial(SIP/[EMAIL PROTECTED]|60|M(screen^${CALLERIDNUM}))
Here is the macro
[macro-screen]
exten = s,1,Wait(0.2)
exten = s,2,say number ${ARG1}
exten = s,3,Read(ACCEPT|screen-accept|1)
exten = s,4,GotoIf($[${ACCEPT} = 1 ] ?7:6)
exten =
I want to set up a feature or extension where you can enter your number and
have asterisk call you back. useful for overseas and some cell phone
packages. I started with privacy manager but when I issue hangup() it seems
the context also terminates. I'm guessing AGI may need to get involved
The Dial command can be made to make an announcement to the called party
before channel is bridged.
Is it possible to make that announcement a Festival command in some way.
--
Howard.
LANNet Computing Associates;
Your Linux people http://www.lannetlinux.com
Hi,
Case1:
-
-- extensions.conf
exten = 1023,1,Voicemail(101)
exten = 1023/101,1,MeetMe(200)
Case2:
-
- extensions table (using realtime extensions)
++-+--++--+-+
| id | context | exten|priority| app | appdata |
Well because I had sooo may problems with chan_agent.c I wrote this. I'm
releasing it under LGPL but if you use it or anything please let me know.
It'd be interesting if anyone finds this more useful than just a pile of
junk.
I've included a (working) example extensions file. SIP phones are
Attached is the example extensions.conf
extensions.conf
Description: Binary data
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Oh i forgot to mention
I have found a limitationcalls going through the queue system can NOT
be parked properly. More precisely with my stdexten macro and/or the agent
logic stuff the calls can NOT be rang-back to the original extension. They
end up (in my example) in from-sip,s,1
Has anyone been able to get the Dial Macro Patch applied to the
current CVS stable?
http://search.ebay.com/x100p_W0QQfkrZ1QQfromZR8
I know that this is in the CVS-HEAD, but I need the CVS-stable so that
I can utilize app_suppervaletparking
Thanks in advance,
Brian
Wow, I hate bad cut and pastes. This should have been:
http://bugs.digium.com/bug_view_page.php?bug_id=0002905
(I guess you all know what I was looking at before :)
On Thu, 13 Jan 2005 at 10:18 Brian S. Adelson ([EMAIL PROTECTED]) wrote:
Has anyone been able to get the Dial Macro Patch
Scheda wrote:
If anyone knows of a linux applicable IAX softphone,
I'd be more than willing to give it a shot, but I haven't found one so
far.
Have you tried iaxcomm?
http://iaxclient.sourceforge.net/iaxcomm/
--
Cheers,
Matt Riddell
___
Hey, I'm having some errors whenever I dial out and I can't dial in at
all. I'm using NuFone as my provider just so you know.
Jan 11 17:39:46 WARNING[1771]: chan_oss.c:413 soundcard_setinput:
Unable to re-open DSP device: No such device
Jan 11 17:39:46 WARNING[1771]: chan_oss.c:572 oss_write:
Scheda wrote:
Hey, I'm having some errors whenever I dial out and I can't dial in at
all. I'm using NuFone as my provider just so you know.
Jan 11 17:39:46 WARNING[1771]: chan_oss.c:413 soundcard_setinput:
Unable to re-open DSP device: No such device
Jan 11 17:39:46 WARNING[1771]: chan_oss.c:572
Well, I can't find a softphone thus far for linux that works with IAX.
I only have one computer running so far. But in a few weeks I will be
able to get another box to set asterisk up on and then I can use
windows as well. If anyone knows of a linux applicable IAX softphone,
I'd be more than
I have more FXO ports on TDM400's than I have PSTN lines available for
testing. When all the lines were used up (the FXO ports are all in zap
group 2), and I did a Dial(Zap/G2/${phone},5) it thinks the Dial succeeded
even though there is neither line voltage nor dial tone. Can at least the
I have more FXO ports on TDM400's than I have PSTN lines available for
testing. When all the lines were used up (the FXO ports are all in zap
group 2), and I did a Dial(Zap/G2/${phone},5) it thinks the Dial succeeded
even though there is neither line voltage nor dial tone. Can at least the
lack
http://www.voip-info.org/wiki-Asterisk+config+extensions.conf+sorting
Erik
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Ian Chilton
Verzonden: dinsdag 14 december 2004 11:33
Aan: [EMAIL PROTECTED]
Onderwerp: [Asterisk-Users] Dial Plan Problems
Hi,
I
On Tue, 2004-12-14 at 21:32, Ian Chilton wrote:
Hi,
I am having a few dial plan problems which I wondered if anyone would be
able to help with.
Firstly, I wanted to send 0800 calls through 1 sip provider and other
08xx calls through another. I have this:
exten =
Dear group members,
Somewhere in this representation:
http://graphics.cs.uni-sb.de/VCORE/Publications/mark_spencer/mark.smil it is
mentioned that one can cal an Mp3 file. How is this implemented? When this
Mp3 is playing, is it then still possible to receive a call?
Thanks,
Willy
-Original
Hi,
I am having a few dial plan problems which I wondered if anyone would be
able to help with.
Firstly, I wanted to send 0800 calls through 1 sip provider and other
08xx calls through another. I have this:
exten = _0800.,1,Dial(SIP/[EMAIL PROTECTED],30)
exten = _0800.,2,Congestion
exten
Is there anywhere where I can download this? And any other presentations?
Regards,
PaulH
-Original Message-
From: Satchid [mailto:[EMAIL PROTECTED]
Sent: Tuesday, 14 December 2004 9:58 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Dial
For some reason I cannot get the 'D' option to send dtmf after connect.
This doesn't work
exten = _XXX, 1, Dial(Zap/r3,10,d(300) )
This does:
exten = 300, 1, Dial(Zap/r3,10,M(to-300) )
[macro-to-300]
exten = s,1,SendDTMF(300)
Of course, what I really need to send is not 300, but $EXTEN
but since I
-Commercial Discussion
Onderwerp: [Asterisk-Users] Dial Plan Help
All,
I've got a problem here. We are using a Digium 4 T-1 board in our * server.
The T-1's are ISDN. The problem I'm having is that we have an ivr setup so
that when someone dials our DID it goes to the s extension and starts
playing
-Original Message-
From: Shaun Tierney [mailto:[EMAIL PROTECTED]
Sent: 02 December 2004 22:37
To: Asterisk Users
Subject: [Asterisk-Users] Dial Command M(x) Option
http://lists.digium.com/pipermail/asterisk-users/2004-October/
065540.html
I never did find a solution
http://bugs.digium.com/bug_view_page.php?bug_id=0002905
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- Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Dial Command M(x) Option
http://bugs.digium.com/bug_view_page.php?bug_id=0002905
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- Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Dial Command M(x) Option
What version of Asterisk should I be applying this patch to? The patch
command doesn't seem to be working. I think because the dates on the
files
in Asterisk 1.0.2 don't match the dates in the diff file. Any
All,
I've got a problem here. We are using a Digium 4 T-1 board in our * server.
The T-1's are ISDN. The problem I'm having is that we have an ivr setup so
that when someone dials our DID it goes to the s extension and starts
playing the ivr which is fine, but if someone dials an extension for
On Fri, 2004-12-03 at 15:52 -0500, [EMAIL PROTECTED] wrote:
All,
I've got a problem here. We are using a Digium 4 T-1 board in our * server.
The T-1's are ISDN. The problem I'm having is that we have an ivr setup so
that when someone dials our DID it goes to the s extension and starts
exten=200,Goto(office,102,1);forward to 102 in office context
exten=201,Goto(office,110,1);forward to 110 in office context
These are invalid -- no priority -- and hence dropped. Didn't you see the
errors while loading (it's easy to miss, there's plenty of stuff output).
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