similar to: Passing Variables From Dial Macro To Parent Ruby

Hello, I've been looking for this on voip-info and this list threads, and I am guessing I am not looking right. What I need is the way to capture (and write to DB) the information on who 'picked' or 'received' the incoming call. Here is the example of .rb file that is called from extensions.conf: private def lokal

Hello everyone. I need a quick help on how to capture who answered the call with agi. Here is an example: -- Zap/32-1 is ringing -- Zap/33-1 is ringing -- Zap/34-1 is ringing -- Zap/35-1 is ringing -- SIP/operator1-e77f answered Zap/23-1 So how can I capture this value and write it to mysql? I already have this:

Here's something that should be easy for RUBY pro's. Here is a script: 1.times do r = $agi.exec('DIAL', SIP/voipuser&Zap/32&Zap/33&Zap/34&Zap/35) r = $agi.get_variable('DIALSTATUS') # $agi.set_variable(' WHOANSWERED

Hello, We have 2 grandstream GX 2000 phones behind NAT and Asterisk outside this natted network. We have the issue with calls to these SIP phones - no audio. It is probably the problem with port forwarding on router - but I am not sure how can I forward same sip ports (5004 to 5100) to two phones (nat addresses?)? Any help appreciated! Z. Zivanovic -------------- next part

I am trying this approach to see who picked the line: Here is what i am doing: EXEC DIAL SIP/ vaso &Zap/35||M(testing^30086) Macro: [macro-testing] exten => s,1,DumpChan() exten => s,2,AGI(whopicked.rb) exten => s,3,Hangup()

Hello, My questions are about Autodesk applications (AutoCAD 2008 & Civil 3D 2008). The main problem is that I can not install them to wine. When I insert CD, and on terminal type: Code: wine setup.exe I'm getting this messages: Code: zarko at zarko-pc:/media/cdrom$ wine setup.exe fixme:process:IsWow64Process (0xffffffff 0x33feb4) stub! zarko at zarko-pc:/media/cdrom$

I have a dialplan that looks like this: [dorecord] exten => _*99XX,1,Verbose(2,Doing custom record) exten => _*99XX,n,Answer() exten => _*99XX,n,Verbose(2,Doing custom record - before wait) exten => _*99XX,n,Wait(0.5) exten => _*99XX,n,Verbose(2,Doing custom record - before record) exten => _*99XX,n,Record(/tmp/prompt${EXTEN:3}.gsm) exten => _*99XX,n,Verbose(2,Doing custom

I am running asterisk 1.6.0.1. I have a Grandstream GXP280 phone I use for testing. In addition I register a zoiper SIP soft phone. For the Grandstream I have busylevel=1 in sip.conf. If I place a call from the GXP280 to zoiper and then put that call on hold from the zoiper side and then call GXP280's extension, asterisk indicates the phone is ringing. As the GXP280 is a single line phone it

I sent this email a few days ago but did not see any responses to it: > I am running asterisk 1.6.0.1. I have a Grandstream GXP280 phone I use for > testing. In addition I register a zoiper SIP soft phone. > > For the Grandstream I have busylevel=1 in sip.conf. > > If I place a call from the GXP280 to zoiper and then put that call on hold > from the zoiper side and then

I have an extension defined like this: exten => do_monitor,1,Answer() exten => do_monitor,n,NoOp(Just got '${CfMC_ActionID}') exten => do_monitor,n,ChanSpy(${CfMC_WhoHear},qX) exten => do_monitor,n,Hangup() I use an AMI packet like this: Action: Originate Channel: Agent/1001 Exten: do_monitor Context: cfmc_cdi_private Priority: 1 Variable: CfMC_ActionID=callE1334 Variable:

I would like to be able to stop the chanspy application and go to the next step in the dialplan but I do not see a way to do that. I have looked at the code and I do not see a way to stop the chanspy application. Even if there are no channels that match the chanprefix pattern the chanspy application is not exited. Hitting the * key stops spying on a channel but then starts spying on the same

Jim, and all: Thanks for the response. If I can repeat what you are saying: you don't have to define the multiple lines in sip.conf? For comparison, with my current esi setup, we have 10 outgoing lines. If one line is busy, then the service just rolls to the next number. This is set up with the phone service. That doesn't have to done with outgoing sip lines? Does the dialstatus

I am using dtrace on a Mac OS X Leopard beta. Currently I''m tracing read operations. I''d like to get info about the process which is creating the reads, as well as information about parent processes. The least amount of information I need is pid and process name, but more would certainly be a plus! I know that in the probe entry, pid, ppid, and process name are easily

Hi! We're using sernet-samba 3.0.28-21 on debian. (The smb.conf can be found below) There are 2 subfolders in root share directory "ponudbe" and "projects": srv03:/srv/data/test# getfacl * # file: ponudbe # owner: root # group: root user::rwx user:COMPANY+administrator:rwx group::r-x group:COMPANY+ponudbe_rw:rwx mask::rwx other::--- default:user::rwx

I work at home with standard residential cable Internet service and I wanted to test CPA for use with our dialer solution. The first problem I ran into is that CPA only works with a SIP provider that does IP based authentication opposed to usename/password authentication. After I got an account setup to solve that problem I thought I was on my way to being able to test. No so. I got asterisk

We have a client that currently has a Microsoft Lync setup. I must admit I know nothing about this setup. What we would like to be able to do is allow the phones on desks connected to this server the ability to dial something that would allow the phone to access an asterisk box to be able to do an agent login over their LAN. Is there any way to do this? Can the Lync server have a SIP trunk to

1250628579|MANAGER|S1140|Agent/265|ADDMEMBER| Above is an example of a line from the file. Is there an option to have this date/time output in a way that can be read without conversion? The date/time in full and messages is formated so it can be read directly. -- Jim Dickenson mailto:dickenson at cfmc.com CfMC http://www.cfmc.com/

It seems as if there is a problem if one uses Local/SomeName at some_context in a Dial application. If the extension is changed to somename then things work. I have an extension SomeName defined. In another extension I try to dial this extension and it does not work saying extension/context not found. The extension displayed has uppercase characters so the case is not lost. If I change the

Is it possible to SIP trunk to this Cisco device so that phones connected to the Cisco box can dial extensions on the Asterisk box? What I would like to be able to do is dial some extension(s) on phones connected to the Cisco box and have the call routed into extension(s) on the Asterisk box. One of our clients has a call center with 65 analog phones connected to the Cisco box. We would like to

I have an asterisk system in Costa Rica that connects to a SIP provider in Atlanta. Sometimes SIP packets seem get dropped or retransmitted too quickly. In trying to debug this I turned on SIP debug in Asterisk and the SIP provider enabled packet capture on his end. What I saw was me sending an invite, them sending a 100 Trying, me sending a cancel, me sending a retransmit of the cancel, me