similar to: Manager API - ExtensionState help please.

Hi, I am trying to get the extension status (weather it has dialed outgoing call via SIP or IAX2), using the following piece of code however it always returns -1 on all the extensions (valid/invalid). Am i missing something ? Any help. Thanks ----------------------------------- #!/usr/bin/perl use Asterisk::Manager; use lib './lib', '../lib'; $|++; my $astman = new

New installation from AsteriskNow 3.0.0 with asterisk 11 and freepbx, running Digium D40 and D70 phones. Direct-dialed extensions work fine, but extensions in RingGroup won't ring - dialparties.agi apparently removes them from the dialstring pre-emptively: dialparties.agi: EXTENSION_STATE: 4 (UNKNOWN) dialparties.agi: Extension 2010 has ExtensionState: 4 What can I do? Any extension

Hello List, I have been working on a PHP application in order to build a BLF style script. Until now everything is going Ok but something a little (in my oppinion) strange is going on with the 'ExtensionState' command; The problem is that it does not returns the 'Status' as it's suposed to, mentioned in the A.T.F.O.T book - version 2., where it sais something like:

This is my first attempt to write software of any sort. What I am trying to is to use a .php page to query asterisk Manager and get the ExtensionState for each particular extension. Then when it has the answer it outputs an XML file for use as the directory on a Cisco 7960 phone. What I am thinking is that when the user hits the directory button to veiw the directory that is at this URL it will

Upgraded from 16.2.14 to 1.6.2.15 on Fedora 13, with aastra 9133i and 57i. On 9133i and 57i: #<extension># works for a blind transfer. Xfer<extension>Xfer doesn't! All this worked on 1.6.2.14. Nothing useful on cli, verbose 3, DEBUG. Here extension 169 answers an outside call, and tries to transfer it to 145 using the Xfer button: -- SIP/169-0000009c answered

I appologize for the long, new-ish question, but after a few days of trying to work a solution by reading through the list archives and WIKI and coming up with what I thought would work, I think I'm just not getting a fine detail. I titled this thread "Setting up Security Groups" because I'm trying to set up some sip user groups with certain calling rights, e.g., one group of

I'm trying to create several contexts for extentions with different levels of access to features and I'm wondering how the heck do I include all the contexts so that you can call internal to any extention in another context without giving the features of the higher level context to the lower level context? ie..... [admin] include => local include => longdistance include =>

I'm trying to use the FollowMe app with Asterisk 1.4.17. I've followed the WIKI page on setting it up but I can't seem to get it to work. Here is my Asterisk version: pbx1*CLI> core show version Asterisk 1.4.17 built by root @ pbx1 on a i686 running Linux on 2008-01-10 12:08:48 UTC Here is a log of when the FollowMe is being called: NOTE: I've tried to use the AstDB as

Hi, This problem may be related to a configuration problem but I believe it is a bug in the FOP since restarting the FOP server clears the problem. Here is the scenario: Using AgentCallBackLogin and have four agents logged in a call is made to one of the agents directly from an internal phone. Okay so far. Call is hung up and the same extension is used to call another agent okay again, no

I have an interesting dilemma with sequencing in the dialplan. Up to now, I have assumed that the extensions in the dial plan were tested in the order that they appear in extensions.conf. In other words, I have the following fragment which was designed to dial toll free on the PSTN and all other long distance on VoIP: [longdistance] include => local

We are trying to implement "area-code dialing" in our asterisk PBX. Basically: we will have a number of customers, who may be in different area codes, that want to direct-dial each other's extensions. We want this to work like a "real" centrex, in that seven-digit numbers should try (1) "local" VoIP extensions, and then (2) "local" PSTN numbers.

Hi all, I am looking for examples of the extensions.conf that puts all incoming calls into a context where extensions can be dials, and all phones in a context where extensions and outside calls can be dialed. i.e. I have seen: [incoming] include => sip-extensions [sip-extensions] include => longdistance [longdistance] .... Doesn't this allow any internal callers to make external

I have been experiencing a rather odd issue with my (newly connected) T100P card. It is installed in my Asterisk server, which is working great with SIP clients. The card is connected to a channelized T-1 line (24 channels). The odd bit is that outgoing calls (Asterisk --> T-1 --> PSTN) works just fine. However, incoming calls (PSTN --> T-1 --> Asterisk) are dropped after about 20

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

Still trying to upgrade to 1.6.0.9 for 1.4. It worked - it worked all day yesterday, but this morning: -- Executing [646xxxyyyy at longdistance:1] Answer("SIP/172-08276a60", "") in new stack .......... -- Executing [646xxxyyy at longdistance:6] Dial("SIP/172-08276a60", ""DAHDI/g2"/1646xxxyyyy") in new stack May 27 09:56:57]

Can someone please explain what I am doing wrong here? I only want the extensions listed in long-users to be able to access the longdistance context. If I do this, I get a congestion tone no matter what I dial. If I add a [default] context and include => longdistance, then the local callers can call the long distance number fine, which is not what I want, but I still want long-users to be

A call comes in from the PSTN, Asterisk answers it, it goes to the directory, and then to the extension the caller designates and the user at that extension answers. The user at the extension then wants to transfer the call to another extension; on the Cisco 7940 they push the ?more? soft key, then the ?Transfer? soft key, then enter the extension number they want to transfer to, and hit the

Howdy, My first forray into using res_mysql.conf for realtime access of sip users and extensions. I have the following relevant section of extensions.conf: --- [trunklocal] exten => _NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}}) [local] include => trunklocal include => trunktollfree [longdistance] include => local include => trunkld [international] include

HI, I dont understand the syntax of the dial application when used like this: Dial(Local/${EXTEN}@longdistance) i want to know what is this "Local" doing instead of Tech like SIP, IAX, H323? -- Regards Rizwan Hisham Software Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am trying to receive both CID and DNIS from the telco through a non-pri T1. Currently I have the T1 setup and operational both outbound and inbound calls are completed as should be expected. The calls came in and were placed in the context specified in zapata.conf on exten => s,1. I have requested that the telco provide callerid (they call it ANI) along with 10 digit dnis for my 800