similar to: Ring the second line when 1st line is busy

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

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:

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

Hi, I'm still in the process of debugging this, but I have a gotoif statement that looks like this: exten => 26,1,GotoIfTime(7:00-18:00|mon-fri|*|*?ext-queues,210,1) exten => 26,n,Goto(ext-local,${VM_PREFIX}127,1) I have others setup the same way that also seem to have the same 'issue'. The issue is that they work, but they seem to require (and I don't understand why) a

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

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

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 @@

Hello , I want to enable recording for a few extensions. In sip.conf it is defined as record_out=Always record_in=Always under the section of extension.but it doesn't work. Extensions are defined in the extension_additional.conf file like exten => 10,1,Macro(exten-vm,10,10) exten => 10,hint,SIP/10 exten => ${VM_PREFIX}10,1,Macro(vm,10,DIRECTDIAL) I can't be sure

Hi, I have Asterisk-1.2.18 install with FreePBX & more than 75 extnsion, daily I come accross an issue & try resolving them its either user learning curve or my ignorance. But, I dont know what to say regarding this issue. I have my Dial Plan for internal users to have a 3 Digit Extensions. So instance my Ext is 239 & someone dials the main #, its gets the

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(-)

Hello. I am running Asterisk v1.6 on Ubuntu 10.10 with FreePBX v2.8. When I call the voicemail for any of my extensions, the call just dies. On a softphone, I get no sound whatsoever; it just hangs up after a couple of seconds. On my handset attached to my SPA-3102, it get a sound like when you leave an analogue phone off the hook. I have three extensions setup and they all do the same thing.

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:

.... does anyone onlist have any 1st-hand experience (good or bad) w/ DirectDial.com for procuring computer components ? Thanks in advance :-). I am interested in them as a potential source for parts to build an Opteron box running Linux, probably CentOS, to bring this slightly on topic .... -- William A. Mahaffey III ---------------------------------------------------------------------

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

Hi, We have a PRI connection & when its was on test networks we had echo problems withoutside line. So I bought a TE212P card resolve the echo problem. Which did to an extent. Its using asterisk 1.2.18 & RHEL4-Update 4. But now when we are live, there is a terrible echo between 2 SIP calls. If I call the same extension from outside the voice is clear. I am not sure whats

I had MCGP working to a ADIT 600 fine with debain sarge stable / asterisk stable - wanted to try red hat and got the below message - then I re-installed debian and am still getting the same message below - any comments are greatly appreciated - I did play with the config files with no prevail - the Adit seems to be doing its job per tech support at CAC. I listed my conigs below I go off hook

Hi, I have a production asterisk-1.2.8 system with FreePBX & PRI Digium card. I am looking for a paging system to an external speaker. I can page to internal Polycom 501 VoIP. But, what hardware or system do I need to integrate with the asterisk to have this acheived. -- Deepak Linux your Life, Don't Window it [[]] { All for the best }

Hi, I have a class to add users to all the host servers. We are in the process to have a coexisting user which belongs in NIS & as well as /etc/passwd. We have NIS clients (yp running) on all host servers. So when running puppet is fails to add or modify user, bcos the user already exists in NIS. Eg: A user pcruise is an existing NIS user. When using useradd or

Hi, I am using file function to copy a directory & its content to few servers. I do it below way. I am not sure whether file uses sync mode or does a copy of every file over & over when something changes. Bcos when running the puppet daemon using puppetd -test it takes a while. Where can I find a list of function like file, user group etc & what args to pass file {

In this dialplan, the SIP user agent is a Sipura two line adapter with line 1 as SIP ID "1000" and line 2 as SIP ID "2000". Basically I have this set up so that 1000 and 2000 are "lines in hunting" on incoming extension "555". I want an incoming call to try to ring ext. 1000, if 1000 is busy, then ring 2000, if 2000 is also busy than ring Voicemail. Here