Displaying 20 results from an estimated 2000 matches similar to: "GotoIf Problems"
2019 Dec 11
0
[PATCH 2/2] tests: fix podcheck tests
Pass to the various podcheck invocations the paths with POD files that
are included.
Followup of commit 46e59e9535c2fcd1c188464b5249a249f22af1a0.
---
cat/test-docs.sh | 13 +++++++++----
customize/test-virt-customize-docs.sh | 1 +
diff/test-virt-diff-docs.sh | 1 +
edit/test-virt-edit-docs.sh | 3 ++-
fish/test-docs.sh |
2003 Jun 24
1
Problems with # and extensions.
I get the following message when I dial 74#. Does anyone have any ideas
on what might be going on? If I don't require numbers to be terminated
with # everything works as expected, but you have to wait for the digit
timeout, of course.
MESSEGE:
DEBUG[1150520624]: File pbx.c, Line 1683 (ast_pbx_run): Oooh, got
something to jump out with ('#')!
-- Invalid extension '#' in
2004 Jan 02
1
Asterisk Gotoif / last called
Hi guys
Ive been trying to get this to work for ages now, basicaly im trying to do if ${woteva} = "" (nothing), or its none existenant then do label 1, else label 2. for my last called function, so it will play a different message if theres no last call in the system or it was anonymous.
ive tried
exten => 1000,1,GotoIf($[${last-call${CALLERIDNUM}} = ""]?4:5)
and heaps of
2007 Jun 07
3
getting at ${CALLERIDNUM}
Hi all --
I'm having awesome fun with Asterisk & voicepulse connect together.
So cool.
I'm trying to have the caller id read back to me. Do I need to do
something to have this sent across in the sip.conf? Or is there
something I need to do somewhere to enable the reading of this data?
Thank you!
Matt
Here is my extensions.conf
exten => _XX.,1,Answer()
exten
2006 Jan 18
1
bug in Authenticate application ?
I'm Japanese. Sorry,English is not so understood,Please let me question by
items.
In Asterisk-1.2.1 and 1.2.2,I cannnot understand the operation of
Authenticate application's 'j' option.
exten => 123,1,Answer()
exten => 123,2,Authenticate(789,j)
exten => 123,3,Playback(pin-number-accepted)
exten => 123,4,SayDigits(111)
exten => 123,103,SayDigits(999)
In this
2003 Nov 04
1
Demo Weather Report AGI v2.0
Some of you may know me as ManxPower from #Asterisk at irc.freenode.,net
I've posted my demp weather report Asterisk AGI script at
http://www.fnords.org/~eric/asterisk/downloads/
I have no affiliation with Cepstral.
Below is the README:
Contact: Eric Wieling <eric@fnords.org>
If you want a demo of this AGI script you may call via IAXtel
1-700-923-3645 extension 2101. Option 23 is
2003 Oct 31
1
Making list of IAX providers
I want to have a list of companies providing services via IAX on my
Asterisk web page. If you know of a company that does this or run a
company that does this please e-mail me at eric@fnords.org with 1) web
site, 2) contact info and 3) services provided. I don't want to put
pricing info on the page.
Thank you.
--Eric
--
Sample configs, scripts, more : http://www.fnords.org/~eric/asterisk/
2003 May 14
20
Call forwarding
Yo,
Inspired by the example in the tips & tricks-section of
"http://www.junghanns.net/asterisk/", I built a more elaborate
call divert-feature. This one validates if the extension a call-forward
is to be set to is actually valid for the current context and
additionally saves this context into the DB and always uses it to
originate the divert from, as you can't expect the
2003 Nov 05
2
Ping AGI Demo
I have a ALPHA version of my new ping AGI demo available.
Access via:
IAXTel 1-700-923-3645
or
Dial(IAX2/guest@ext.fnords.org)
When asked for an extension, enter 2101. This will bring you to the
System Services menu. The Cepstral version of the ping is option 28,
the Festival version of the ping is option 32.
Please report problems and/or issues directly to me. I'm trying to get
2007 Aug 23
0
How to get callee extension in applicationmap(features.conf)
hello,
I use trixbox.I had define a feature code testfeature:
[applicationmap]
#include features_applicationmap_additional.conf
testfeature => *3,callee,Macro,vote
[featuremap]
blindxfer => ## ; Blind Transfer
disconnect => ** ; Disconnect Call
automon => *1 ; One Touch Record
atxfer => *2 ; Attended Xfer
testfeature => *3
here is my macro-vote:
[macro-vote]
exten
2003 Oct 21
0
Iitter Buffer Settings
I'm trying to come up with good jitterbuffer related settings for my
Asterisk boxes.
I ran 4 pings for about 2 days from my main Asterisk server to remote
Asterisk servers. During that time there were some large file uploads
which caused the max rtt to be quite large.
Here are the results:
pkts loss min avg max mdev
132013 %0 70.36 78.13 1967.37 36.04
132013 %0 98.95 120.46 2419.24 111.26
2003 Oct 22
1
Inbound IAXTel failing?
Is anyone else having trouble receiving IAXTel calls? I don't know if
it's my config that's broken or IAXTel that broken. Several people have
given me their IAXTel numbers and calls to them all fail. I can call
FWD numbers via IAXTel just fine.
--Eric
--
Sample configs, scripts, more : http://www.fnords.org/~eric/asterisk/
BTEL Consulting 504-899-1387 or 850-484-4545 or
2003 Oct 25
0
Asterisk External Resources Page
I've submitted http://bugs.digium.com/bug_view_page.php?bug_id=0000434
requesting that Digium put up a page with links with external Asterisk
related resources. If you have a web site with Asterisk related
information, patches, samples, documentation, etc, please add a bugnote
to the above URL. There is a lot of good information out there, but
time and time again I hear complaints that
2009 Apr 17
1
how to call forward on 1.6
Hello,
I want to enable call forwarding for asterisk 1.6.0.6
I couldnt seen any config or option on gui or extensions.conf about it.
I found some dialing plans to enable it on web as follows:
[apps]
; Unconditional Call Forward
exten => _*21*X.,1,DBput(CFIM/${CALLERIDNUM}=${EXTEN:4})
exten => _*21*X.,2,Hangup
exten => #21#,1,DBdel(CFIM/${CALLERIDNUM})
exten => #21#,2,Hangup
;
2005 May 23
1
SendDTMF into a conference room
I have been trying to figure a way to SendDTMF into a MeetMe room using
the Manager API.
I can't redirect everyone into another context and then bring them back
because that would mess up my logic.
I am trying to use local channels and the originate Action to accomplish
this.
Exten: 3441115
Priority: 1
ActionID: actid-00000001
Context: senddtmftones
Action: Originate
Channel:
2005 Jan 15
1
SayDigits -- ToneDigits??
I have a user who wants to receive an ANI spitback in DTMF. Right now,
the "SayDigits(${CALLERIDNUM})" command works fine with voice. But I'd
like to end up doing both. Something along the lines of:
exten => 34,1,Answer
exten => 34,2,Wait(1)
exten => 34,3,Playback(vm-extension)
exten => 34,4,SayDigits(${CALLERIDNUM})
exten => 34,5,Wait(2)
exten =>
2003 Sep 19
1
Aastra 390 w/ADSI - Doesn't automagically use "Asterisk PBX" script
I have an Aastra 390 ADSI phone. It's not locked.
I can call ADSIProg without a problem and it programs my phone. Calling
Voicemail2 also programs my phone.
However, in order for the VMail option to appear on the screen I have to
go into the Services menu, pick Asterisk PBX and pick Select.
Then the VMail softbutton appears on the screen, but any time I make a
call it goes back to the
2006 Nov 08
0
Warning: "Channel does not have a CDR" when doing ForkCDR
Gang,
I'm having this error pop up when I do a ForkCDR, and I'm not sure how
to get around it. Here are a few log lines:
Nov 8 10:37:08 VERBOSE[28079] logger.c: -- Executing
ForkCDR("Zap/49-1", "") in new stack
Nov 8 10:37:08 WARNING[28079] app_forkcdr.c: Channel does not have a
CDR
The scenario occurs like this:
I use a .call file to generate a call on
2003 Nov 10
1
Menu's & Sub-Menu's
Hi all,
I am trying to get a Menu system to work, and having probs with the internal
extensions from the prompts.
Below is the extensions.conf section.
[mainmenu]
;
;"main menu" context with submenu
;
include => default
exten => s,1,Answer
exten => s,2,Background(hello)
exten => s,3,Background(thank_you)
exten => s,4,Background(if_you_know_extension)
exten =>
2010 Mar 03
0
Is this a bug?
Hi List,
I'm working on making one of my applications multi-lingual and
find that I have this problem. The SayDigits and SayNumber functions in
1.4.26.2 recognize but don't process the CHANNEL(language) function. Here's
a snippet to verify.
exten => 317,1,Answer
exten => 317,n,playback(tt-monkeysintro)
exten => 317,n,Set(CHANNEL(language)=es)
exten =>