similar to: Time Based Goto Ifs Act Strange?

Displaying 20 results from an estimated 2000 matches similar to: "Time Based Goto Ifs Act Strange?"

2007 Jul 03
1
Configuring BLF or Asterisk presence/Hints feature
Hi all, I am working on asterisk 1.2.18 zaptel 1.2.17 Polycom 650 polycom 430 SIP version 2.0.3.0131 for IP 650 SIP version for IP430 2.0.3.0127 freepbx 2.2.1 I am trying to configure BLF using asterisk but failed. I would be thankfull if somebody help me. Regards FArooq ********************************************** 1 ********************************************** in my
2006 May 26
1
Not able to make any calls
Hi All, I have registered "abhijit" for SIP in asterisk Server. I am able to register my softphone (SJPhone) to the server using the name "abhijit". But whenever I try to make any calls I am gettinh the following error message:- *CLI> -- Registered SIP 'abhijit' at 172.20.28.85 port 5060 expires 120 May 26 07:34:52 NOTICE[2761]: pbx.c:1738 pbx_extension_helper:
2009 Jul 21
1
[PATCH node-image] Moved all temporary files into a single work directory to clean up.
All temporary files are kept in a single directory. At the end of the autotests that one directory is deleted. Signed-off-by: Darryl L. Pierce <dpierce at redhat.com> --- autotest.sh | 20 +++++++++++--------- 1 files changed, 11 insertions(+), 9 deletions(-) diff --git a/autotest.sh b/autotest.sh index c9f8a2d..d658cf3 100755 --- a/autotest.sh +++ b/autotest.sh @@ -40,6 +40,7 @@ # an
2007 Jun 25
1
Ring the second line when 1st line is busy
Hi, I ma using Asterisk 1.2.18 & FreePBX 2.2.1. I have assigned every users in office with Polycom with 2 extensions as below 555 8555 I have configured Follow-me to ring when the users doesn't picks the phone on line 1(555) after 10 seconds & then ring the line 2(8555). But this is not a standard telephony which I have been advised to change like below. If someone calls
2004 Jul 01
3
R: execute a context from cron
> I want to have call forwarding (from the POTS) > turned on at the close of work and turned off > automatically by *. I would have a look at GotoIfTime: http://www.voip-info.org/wiki-Asterisk+cmd+GotoIfTime That should be much easier than a cron job Regards -Manuel ___________________________________________________ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax
2010 Aug 26
1
Timecondition fallthrough on 2nd GSM Modem, First modem and ZAP's are all fine
Hello, we have an asterisk (1.4.21.2) with ZAP and mISDN channels, the mISDN are 2 incoming GSM Modems, each with 2 simcards. No, the mISDN line one and two are fine, but when I get a call on line 3 something with the time is wrong. Timeconditions fall through to off-hours even if the time of the call is clearly inside business hours, here a log excerpt: [Aug 26 11:04:36] VERBOSE[3112]
2005 Jul 26
2
Stumped on vMail problem, any ideas?
Hello all, I think I have most of my AAH 1.3 setup running (Asterisk 1.0.9), but somehow something is not quite right with my vMail setup. I would have sworn this was all working, but maybe I was just dreaming. Anyway here is what is happening, say I am on extension 200 and I want to call to extension 201. If extension 201 is no connected, then it rolls right into vMail with the message the
2007 May 02
6
allowing call every 15mins
Hello all, I have a set up that answer my customer. and its working well, however, the number of call to technical dept is what i want to reduce. I want all call to get to voice prompt except that that enter when minutes is 15, 30, 45, 60(in multiples of 15 minutes). how can i achieve this and what application can i use to get this done. I will be glad, if someone can give me a hint on this.
2009 Jul 21
2
[PATCH node-image] Adds a preserve option for autotest VMs.
If the -p option is provided, then no VMs are destroyed. Instead they, and their related networks, are left intact. Signed-off-by: Darryl L. Pierce <dpierce at redhat.com> --- autotest.sh | 11 ++++++++++- 1 files changed, 10 insertions(+), 1 deletions(-) diff --git a/autotest.sh b/autotest.sh index c9f8a2d..b72ec98 100755 --- a/autotest.sh +++ b/autotest.sh @@ -219,6 +219,9 @@
2010 Mar 26
3
[PATCH node] Update autobuild and autotest scripts for new build structure
Autobuild has to be updated to call make in the recipe directory and move the resulting iso to the main build directory. Importing the existing autotest.sh script from ovirt-node-image Signed-off-by: Mike Burns <mburns at redhat.com> --- autobuild.sh | 7 + autotest.sh | 764 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 771 insertions(+), 0 deletions(-)
2007 Jul 05
1
Need Help in Asterisk BLF/Presence/Hints
Hi all, I am working on asterisk 1.2.18 zaptel 1.2.17 Polycom 650 polycom 430 SIP version 2.0.3.0131 for IP 650 SIP version for IP430 2.0.3.0127 freepbx 2.2.1 I am trying to configure BLF using asterisk but failed. I would be thankfull if somebody help me. Regards FArooq ********************************************** 1 ********************************************** in my
2009 Dec 15
1
[PATCH] The autotest timeout is now a command line configurable option.
By default it's 120 ms, but can be changed through command line arguments. Signed-off-by: Darryl L. Pierce <dpierce at redhat.com> --- autotest.sh | 16 ++++++++++------ 1 files changed, 10 insertions(+), 6 deletions(-) diff --git a/autotest.sh b/autotest.sh index c67931a..bcd9bd5 100755 --- a/autotest.sh +++ b/autotest.sh @@ -62,6 +62,7 @@ Usage: $ME [-n test_name] [LOGFILE] -i:
2009 May 19
1
[PATCH node-image] Fixing the autotest script.
The test_stateless_pxe_nohd test was broken. Fixed. Result code was not matching the success/failure state for the tests. Fixed. Signed-off-by: Darryl L. Pierce <dpierce at redhat.com> --- autotest.sh | 115 +++++++++++++++++++++++++++++++++------------------------- 1 files changed, 65 insertions(+), 50 deletions(-) diff --git a/autotest.sh b/autotest.sh index 12d3e30..e5e23a8 100755
2005 Aug 02
12
WHat does it take
How many times do you ask for help here before getting a respone? Every single thing I do No matter what I get busy extensions. I am willing to pay someone to help here. Anybody got a clue? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050802/d0d1326c/attachment.htm
2005 Aug 27
2
gotoiftime
Does anyone know if gotoiftime can take any subset of 7 days for the days of the week or only a contiguous range? I want to use gotoiftime to change dialplan behavior on Monday, wedneday and Friday -- Executing GotoIfTime("Zap/8-1", "09:00-20:00|MON WED FRI|?21") in new stack Aug 27 19:27:25 WARNING[2676]: pbx.c:3729 get_dow: Invalid day 'MON WED FRI',
2020 Mar 28
0
[klibc:update-dash] dash: var: Set IFS to fixed value at start time
Commit-ID: 6dc1db1bce863f0e0e0abda1b9a58d8cb22863ca Gitweb: http://git.kernel.org/?p=libs/klibc/klibc.git;a=commit;h=6dc1db1bce863f0e0e0abda1b9a58d8cb22863ca Author: Herbert Xu <herbert at gondor.apana.org.au> AuthorDate: Sat, 19 May 2018 02:39:43 +0800 Committer: Ben Hutchings <ben at decadent.org.uk> CommitDate: Sat, 28 Mar 2020 21:42:55 +0000 [klibc] dash: var: Set IFS to
2003 Dec 19
2
GotoIfTime help
Hey All, I need to forward an extension to an other depending on the current time but I could not get it done with GotoIfTime. What I'm trying to do is ring on the extension 1 if time is between 8:00AM and 2:00PM and on extension 2 if is between 2:01PM 11:00PM. exten => 111,1,GotoIfTime(8:00-14:00|*|*|1-12?333) exten => 111,2,Dial(${Person1}) exten => 111,3,Dial(Hangup) exten
2001 Dec 30
1
ifs estimator
Hy all, I have written a small library that provide a new distribution function estimator based on IFS (that are essentially fractals). I would be pleased if any of you can build hte library for Unix-type machines and Windows implementations with the included makefiles as a Mac library is already working. As I'm not able to test these platforms any modification to makefiles is welcome.
2006 Feb 16
1
Playing sound File using GotoifTime function
I want to play a sound file using GotoifTime function. 1) What should be the appropriate format of this type of sound file? 2) Is there any method to copy this file into the destination directory using the browser of a PC other than the asterisk PC (currently i am using cp to copy the file in /var/lib/asterisk/sounds on asterisk PC)??? Waiting for ur kind reply !!
2019 Jan 25
0
[klibc:update-dash] [EXPAND] Split unquoted $@/$* correctly when IFS is set but empty
Commit-ID: af24ffa8f0b9d90e29d6daf77e5349dd3ffe4aec Gitweb: http://git.kernel.org/?p=libs/klibc/klibc.git;a=commit;h=af24ffa8f0b9d90e29d6daf77e5349dd3ffe4aec Author: Herbert Xu <herbert at gondor.apana.org.au> AuthorDate: Wed, 8 Oct 2014 15:24:23 +0800 Committer: Ben Hutchings <ben at decadent.org.uk> CommitDate: Fri, 25 Jan 2019 02:57:21 +0000 [klibc] [EXPAND] Split