similar to: ruby-agi-1.1.1 released !

I just released ruby-agi-1.1.0 Here is the release notes: addition of method 'jump_to' jump_to would take three arguments ex. jump_to(context, extension, priority) enhanced callerid, calleridname and calleridnumber. calleridnumber is an addtional method to this release. Regardless of Asterisk version above three caller methods would return uniform value. Such as,

Hi, I've been trying to use the set_callerid function in the AGI. It sets the CallerIDname properly but I can't figure out how to set the CallerIDnumber. Is it at at possible ? Cheers. SL

Anyone out there running 1.2beta with a PRI and having CDR problems? I just upgraded to most recent everything and now my CDR's look like this: "","????","9035646130","copper_routing","????","Zap/65-1","SIP/netl-a3ac","Dial","SIP/+19035646130@netl|60","2005-08-31 13:03:09","2005-08-31

Hello, I have released Ruby Asterisk Gateway Interface (ruby-agi) v0.0.2b. Any feedback, bug report, suggession, feature request is most welcome. ruby-agi homepage: http://www.rubyforge.org/projects/ruby-agi/ Download ruby-agi v0.0.2b here: http://rubyforge.org/frs/download.php/5965/ruby-agi_v0.0.2b.tgz Thanks, Mohammad Khan

I edited the calleridname.agi patch to only overwrite the name if it is missing. The asteridex option still overwrites the name since it is our master list for known numbers. -- Steven calleridname.agi.patch: --- C:\Documents and Settings\steveb\Desktop\calleridname.agi-orig Tue Jun 13 14:37:09 2006 +++ C:\Documents and Settings\steveb\Desktop\calleridname.agi Tue Jun 13 14:37:09 2006 @@ -16,6

Hi all, Does anyone have any advice about returning multiple values from an ActionWebService? That is, I''d like to implement something like the following. api_method :do_something, :expects => [{:id => :string}], :returns => [{:return_code => :int}, {:return_text => :string}] The easy way to do it would be to

I am working with a customer and their SIP provider is IPitimi. The customer needs to sometimes provide various caller id number for the calls going to IPitimi. They are processing calls for multiple businesses who want their caller id to show up. When no caller id is provided, the From must be the DID at ipitimi ip address and caller id is DID at customer IP address. When caller id is

Here is my products.yml version_control_book: id: 1 title: Pragmatic Version Control description: How to use version control image_url: /images/sk_svn_small.jpg price: 29.95 date_available: 2005-01-26 00:00:00 automation_book: id: 2 title: Pragmatic Project Automation description: How to automate your project

Hi all, I'm new in the list, and I have a problem upgrading from asterisk 1.2 to asterisk 1.4: There is a diference from asterisk1.2 to asterisk1.4 in AMI events. When I do a call to a queue (with the same extensions.conf dial plan) with ast1.2 and ast1.4, in ast1.2 apper 3 newcallerid event in ast1.4 apper only 2. It is normal? anyone knows it? what is the reason? I

I'm having trouble getting the manager interface to behave properly; specifically the Originate event. If I create an originate event as below, the calling phone will auto-answer (as it's supposed to) but the receiving phone never rings. It will timeout at 20 seconds. Action: Originate Channel: Local/201@from-sip2 Context: from-sip Extension: 154 Priority: 1 CallerID: John Doe

Hi everybody, I have corrected this line in extensions.conf by stripping spaces off and now it executes: exten => 22999,1,VoiceMailMain(s${CALLERIDNUM}) when it runs, the mail box number is asked and password too. I expected no question were made, because I inserted CALLERIDNUMBER and s in front of box number. Anybody knows why? Thank to you all, very kind members of this list! Ciao Mauro

Guys. Anybody knows why this is happening? Seems every time I make an internal call, the manager shows this and I don't get the callerid on my identapop but rather the calledid.. Event: Dial Privilege: call,all Source: SIP/intruder1-85f0 Destination: SIP/test-f037 CallerID: 201 CallerIDName: Anton Krall SrcUniqueID: 1117038116.7 DestUniqueID: 1117038116.8 Event: Newchannel Privilege:

Hi, I'm using mysql to define my sipfriends. When authenticating, the calleridname from the calling SIP user (phone) seems getting lost. With "sip debug" I can see in the SIP messages: From: "myName" <sip:4912345@1.2.3.4>;tag=22668125 To: <sip:004954321@1.2.3.4> but I can't find "myName" in any channel variable. Both, ${CALLERID} and

Previously, when I wanted to forward to incoming callerid when I forwarded a call to another number I had to set the callerid on the outgoing call to be that of the incoming number. So today I do this: exten => s,n,Set(CALLERID(name)=${CALLERIDNAME}) because I want the outgoing callerid that I forward to not be the normal callerid of the local extension but I want to forward the incoming

> -----Original Message----- > From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users- > admin@lists.digium.com] On Behalf Of Adams, Gavin Is there any additional information I could provide to start tracking this down? I was thinking about looking into the various applications source to see how they access the data elements for callerid. I know where the values are pulled

Folks, I'd like to change the value of ${CALLERIDNAME} for incoming PSTN calls from certain numbers, but haven't found a way that works. The goal is to provide more informative names on my phones' caller ID displays--e.g., I would prefer to display "ROB CELL" instead of "CELLULAR CALL" when I call home from my cell phone. This is what I tried in the context

Hello Everyone, I've been playing with an agi script for paging sip phones. page.agi will take all available sip extensions and assign them to the global variable PAGE_GROUP. Allowing the phones to be paged from the dialplan with the new Page cmd. Extensions to be excluded are presented as arguments to the agi. Each time a page is made this agi refreshes the global variable. This works with

i've got 3 queues: QUEUE A - >(mon,tue,wed,thu,fri) 8-12/14-18 -> Queue the call > else play a mex reporting the service period QUEUE B - >(mon,tue,wed,thu,fri) 8-12-> Queue the call > else play a mex reporting the service period QUEUE C - >(mon,tue,wed,thu,fri,sat,sun) 8--20 -> Queue the call > else play a mex reporting the service period how can i set it?

I use Jack for getting callee sound. Dial with option M(): JACK_HOOK(manipulate,i(rec_737219:input),o(rec_737219:output),c(rec_737219))=on This works fine, but I need to connect the sound channel to Jack *before* the actual answer. As you can see in the AMI log, between "Ringing" to JACK_HOOK there is a 6 second break. I don't want that. I need a way to launch Dialplan function

This is with an x100p (the motorola chipset) Two problems. Looks like CALLERIDNAME is being used uninitialized. On my other phones the callerid is fine and my buttset shows that the callerid passes the checksum. This is the relevant portion of extensions.conf exten => s,1,Answer exten => s,2,SetCallerID(H ${CALLERIDNAME} <${CALLERIDNUM}>) exten => s,2,Dial(${MGCP_ALL}) Here is