Displaying 20 results from an estimated 6000 matches similar to: "vmoutcall]"
2006 Oct 30
1
dealing with blind transfers to invalid extensions
Running Asterisk 1.2.8 kernel 2.6.13.4-1.
Everything in my dialplan seems to be working well except for one
problem.
When calls are blind transferred to an invalid extension I would like
the call to go to the operator on ext 1000?
What is the best way to do this? Thanks in advance
Here's a snippet of my extensions.conf
[default]
exten=>_10XX,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
2006 Feb 19
3
Loops and Variables
I have the following in my dialplan, counts the number of loops and when
it hits greater then 5, exit. It works, but errors initially with,
"syntax error, unexpected TOK_PLUS, expecting TOK_MINUS or TOK_LP or
tolken; Input: +1".
Could somebody tell me why?
Thanks:
; ****************************************
; Setup a varriable to count the number of
; times the message has been
2005 Sep 15
3
${DIALSTATUS} problems
Hi.
I'm dialling two numbers - one that's unobtainable, one that's busy.
${DIALSTATUS} is coming back ANSWER each time right before the channels hang
up.
Am using the following dialplan macro to dial out.
[macro-advdial]
exten => s,1,Dial(${ARG1},20,g) ; Ring the interface, 20 seconds maximum
exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status
2006 Mar 15
2
Do Not Disturb?
I looked on the voip-info wiki and found sparse and conflicting information
on how to do this with Asterisk...
My incoming lines are all on Zaptel. Is there a simple why to implement a
'*363" (do not disturb) toggle via the dialplan?
It would be nice to be able to pick up an extension, dial *363, and have all
calls sent to voicemail without ringing the extensions. Doing it again would
2005 Mar 03
3
Why ${EXTEN} variable changes after Goto ?
Hi,
I'm trying to implement dynamic routing of incoming calls to local extension
if previous outgoing call was unanswered.
But after I do Goto to s-NOANSWER, variable ${EXTEN} changes to
's-NOANSWER'. I guess this is normal, but I don't understand why ? How to
workaround on this one ?
Thanks in advance,
regards,
Rob.
[outbound-capi-ISDN]
exten => _0.,1,NoOp(Calling ISDN
2010 Apr 29
2
Code in extensions.conf to leave a voice mail in another PBX ?!
Hi Guys,
i spent some time to figure this out (since i love how dialplan is written)
but i decided to ask for your help guys.
i have two asterisk servers one is 1.2 the other is 1.4, from 1.4 (pbx1) to
1.2 (pbx2) i can leave a voice mail without any pb, but from pbx2 to pbx1 it
just hang up.
in pbx2 extensions.conf:
i am using: exten => 8029,1,Dial(IAX2/pbx1/${EXTEN},20,tTWwr)
in pbx1, i
2007 Jul 04
1
Dialout Macro and transfer call in progress
Dear All,
I can not transfer call in progress. What's wrong with my macro? I think tT flags is enough right?
[macro-stdexten]
exten => s,1,Set(temp=${DB(CFU/${ARG1})}) ; Get CFU key
exten => s,2,Set(DNDStatus=${DB(DND/${ARG1})}) ; Get DND key
exten => s,3,GotoIf($["${temp}" = ""]?5) ; If not existing, goto priority 5
exten =>
2007 Nov 29
2
Using existing extensions.conf macros, and co-habitation
I'm trying to set up my extensions.conf file using some of the existing
macros like stdexten, etc. while at the same time having two logically
separate virtual PBX's (with no "default" context) and two trunks coming
into separate contexts, i.e. one for residence and one for my at-home
business.
I noticed, however, that macro-stdexten depends on the "default" context:
2007 Jul 13
3
Macro: s-NOANSWER, _s-.
Hi List;
I have this example for Macro and I am not able to
understand some line, if any one can help me plz :)-
[macro-voicemail]
exten => s,1,Dial(${ARG1},20)
exten => s,2,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,Voicemail(u${MACRO_EXTEN})
exten => s-NOANSWER,2,Goto(incoming,s,1)
exten => s-BUSY,1,Voicemail(b${MACRO_EXTEN})
exten => s-BUSY,2,Goto(incoming,s,1)
exten
2005 Jan 18
1
Asterisk and IAX softphone (firefly) problem/question
Quick question from a newbie,
I have asterisk configured to dial IAX extensions (which works). When
dialing from one IAX extension (using Firefly) to another IAX extension
(also using Firefly), the Firefly client rings on the receiving end and
gives the option of accepting or denying the call. However, when I dial in
to Asterix using a VoicePulse number and dial the same extension Firefly
2009 Jul 20
1
callforward with asterisk-gui.problem with stdexten
Hello, i am trying to enable call forwarding on asterisk 1.6 with
asterisk-gui
If i set my stdexten as follows (with the lines i marked) everything seems
like working.
But if i make any change on asterisk-gui and apply it.. it recreates the
macro-stdexten and deletes my configuration regarding to it.
So where should i add my call-forward configuration???
Where am i making a mistake??
2009 Jul 17
2
How do I create an IVR/Dial Group that works properly?
Hi all,
I am trying to understand how I can get a simple IVR scenario to work
properly (having already removed most of my hair...).
The basic requirement is as follows:
* Caller arrives at our main number
* Caller is greeted and then told they can enter an extension number, if
known, or wait and their call will be connected to an available rep.
* The IVR then dials a group of extensions (if
2009 Feb 26
2
Problems with Outbound Calls
Hi everyone!
I'm quite a newbie at this Asterisk stuff so please bear with me.
We've recently decided to start training in Asterisk via AsteriskNow!
Asterisk version is 1.4.18.1 through AsteriskNow! 1.02
The box we have is paired with a Digium TE110P and we've managed to get
it to the point where incoming calls via a single DID (from NTT Japan)
can be received and answered
2007 Oct 15
2
Voicemail issues in 1.4.11
Asterisk isn't playing my voicemail greetings even though they are defined. Below are the relevant configs(from show dialplan) as well as the level 3 verbose messages asterisk is giving. Also a listing of the directory.
Asterisk just plays the "The person at extension..." message, not the greetings I have recorded.
Thanks
--
asterisk*CLI> show dialplan macro-stdexten
[
2004 Jun 26
2
Newbie needs help
I've been banging my head on a brick wall for about an hour now trying
to understand why the following doesn't work (which is even provided as
an example in the distribution!).
The goal is to create a voicemail-only extension not associated with a
phone. I'd rather not have an extension dedicated to VoicemailMain(),
so I would like the user to be able to hit '*' during
2007 Sep 03
1
Dificult macro, please advise
Hi,
BRIEF RESUME:
Is there any other way to obtain the same result but being easier to
configure?? Thanks!
EXTENDED RESUME:
i've configured a, rather difficult, macro that even for me without
being documented is difficult. I ask for the help of the experts to
know if the functionality it apports can be achieved better in another
way.
What i'm trying is to enable call a channel (e.g.
2007 Jan 19
1
Red: Sip Phone CID
Here is what I have in my extensions.conf file now. Trustrcid and
sendrcid are turned to "yes" in the conf file.
I'm not fully sure how the SIPCalledRPID works though. The example I
found seems to try and provide the stuff automatically (id and name),
but does the SIPPEER stuff even exist?
I think this is probably the right track though. Any insight would be
much appreciated.
2009 Feb 12
4
Asterisk Queue and URL Calling
Dear All
I want to integrate sugarcrm and asterisk , so when customer call the call
center the agent or extension which answers the call , before pickup the
phone and talk to customer , view his/her information if it is available.
I do this as described below :
1-Setup login username for sugarcrm for each extension
2-Extension Users will login to the sugarcrm.
3-Develop php script to handle new
2004 Dec 20
1
Example config for SPA-1001
Hi,
Has anyone managed to create a setup with a Sipura SPA-1001 as a client?
Right now I can connect to the device by dialing the extension number
but when I try to connect from the phone handset to make an outbound
call it gives an unavailable tone.
I'm using Line 2 on the SPA-1001 to connect to the local asterisk
server, line 1 is used to connect to my VOIP provider until I can get
the
2006 Feb 07
1
MFC/R2 in Brazil
I don?t know if the last message was with content. So, I sent again. I have
installed a Digium card TE210P and unicall for use MFC/R2. I think that it?s
all right but I can?t make and receive calls. I?m using asterisk 2.1 with
the patch made by Jos? P. Leit?o and the follow libs:
libsupertone-0.0.2
libunicall-0.0.3
libmfcr2-0.0.3
zaptel 2.1
My number is 34318300. The Telco send me only 8300.