similar to: API Originate Action - distinguishing between No Answer and Invalid phone number

Hi all I have installed AAH 2.2 in my P4 PC following AAH handbook PDF and http://mundy.org/blog/index.php?p=62#amp and made as per the guide says and downloaded SJ Phone, and registered user and when i try to dial the 19197543700 i get message that, all circuits are busy now, please try your call later and when i see in the console i get this mesage any help Called easycall/19197543700

I have Meetme conferencing compiled for Debian as per http://powerontech.com/freepbx-on-debian.htm . I drop a callfile in the outgoing dir, and it intitiates a call to a local extension as a channel, but the call seems to block on the Meetme() command. That extension completes the outgoing Dial(SIP) command to my phone, announcing that leg is the only member of the conference, and just waits. If I

Nik, Looks like you're making some progress. When I first started using A@H I had trouble getting the outbound dialing to work. I wasn't sure where to start, so what I did was skip the macros in the dial plan. I wanted to play around with exactly what digits the telco wanted to see. So I put a specific extension in my [default] context like this: exten =>

I'm having some trouble getting the channel status with an AGI script. #!/usr/bin/perl use Asterisk::AGI; $AGI = new Asterisk::AGI; my %input = $AGI->ReadParse(); $AGI->channel_status('Zap/1-1'); I am now stuck, and don't know how to get the return codes: -1 There is no channel that matches the given <channelname> 0 Channel is down and available 1 Channel

Hello everyone Is there a possibility to catch an outbound callleg ID for the follovong scenario: some carrier -----> ------(asterisk1) --->-----asterisk2 ? I can get inbound callid for asterisk1 with a ${SIPCALLID} in extensions.conf or to look it up in cdrs field (are the same). But how about outbound? I have all calls just forwarded through asterisk1, not answered and for every call I

Hi, Reading this old thread, may I ask if keeping hangup handlers from updating CDR values still enforced in Asterisk 15 ? If positive, would it be very complex to add in Asterisk, a configuration option allowing a system administrator to list in cdr.conf, the CDR fields allowed to be updated in hangup handlers ? I'm planning to store some RTCP stats. Saving them in CDR(userfield) would be

The Asterisk Manager Interface is driving me nuts. Whoever wrote it should be drawn and quartered. Sometimes the data comes back separated by \r\n, and sometimes it's separated by \n. The whole thing is completely inconsistent, and trying to write any kind of API for it is -GHASTLY- Doug.

Hello, To detect an answering machine I have found following two commands, BackgroundDetect (comes with asterisk) MachineDetect (asterisk add-ons) First question, does BackgroundDetect works well with g729? I havn't try MachineDetect yet, what is the benefit of MachineDetect over BackgroundDetect. If anybody used any of this command successfully, please help me. If possible, please let me

Ok... I am reluctant to ask this question as I believe that it may be like asking what someones favorite linux distribution is... but I need to make an informed decision. We are getting ready to upgrade from a TE210P to a quad T1 card with echo cancellation. I am trying to decide between the Sangoma card and the Digium card. I need this to have great quality and I need it to work well. I would

Anyone know if #include works in ael yet? extensions.ael: #include "inc/pbx/global.conf" context test_context { }; *CLI> ael reload May 30 13:56:45 NOTICE[8516]: pbx_ael.c:1120 handle_root_token: Unknown root token '#include' May 30 13:56:45 WARNING[8516]: pbx.c:3758 ast_merge_contexts_and_delete: Requested contexts didn't get merged

When Asterisk first starts up, it will attempt to "bring up" the B channels on any PRI circuits. If you are using A@H then you can log on to the Asterisk CLI (asterisk -r) and then do "stop now" to stop asterisk. Start up Asterisk again by typing asterisk -cvvv at the Linux command line. You should see a bunch of messages on the terminal and then you'll get the Asterisk

With endbeforehexten=no I actually get two CDR entries. One for the call and a second one for the "h" extension. "","13","10","sip-locals","""13"" <13>","SIP/13-00000006","SIP/10-00000007","Dial","SIP/10","2015-08-04 06:28:44","2015-08-04

I am sending this again since I haven't get it back for twelve hours: When I originate call to IAXComm, more or less one of tree calls fails for no aparent reason. Originating calls to SIP clients works as expected. Anybody has similar problems? Is it asterisk or client problem? Asterisk log: Jul 15 00:00:04 DEBUG[1179663]: manager.c:1018 process_message: Manager received command

Hi, Is there a way to enable the usage of UNIQUEID CDR field using a MySQL database backend for storing CDRs without having to recompile asterisk-addons as stated here http://www.voip-info.org/wiki-Asterisk+cdr+mysql ? After version 1.4 it is said in release that it can be done (not sure if it applies to mysql backend) How would it be the syntax in cdr.conf? I tried this without success in

Hi, I am trying to find a way to add the following info in CDRs (with asterisk 1.4.23.1): 1. Codec used 2. RTP QoS statistics 3. RTP IP of remote host 4. For answered calls, the peer that requested to end the conversation I have managed to get 1 and 2 for the caller, like that: exten => h,1,Set(CDR(userfield)=RIP=${SIPCHANINFO(recvip)}

Gang, I recall several months ago that there was a company that was giving away a free 1-port T1 card, with some specific conditions. Do any of you recall who that was? My Google searches are coming up empty and now I'm wondering if I was hallucinating... Thanks, MC -------------- next part -------------- An HTML attachment was scrubbed... URL:

Nik, I'm not sure that "NOP" is correct, but I'm in the states so I'll to defer to someone who knows E1/PRI. When I run zttool I have "OK" under the alarms. Is there a way you can call the telco and confirm the settings? Make sure that framing, coding and D channels are set up on their end the same way you're set up. As for asterisk, here's what I get

> you should mv the file (and in the same filesystem, so 'rename' is used) > You might want to chmod or even chown the file first as well. I wrote a little script that does all of this before the .call file is mv'd into the outgoing directory: cp /tmp/test3.call /tmp/test1.call chmod 666 /tmp/test1.call chgrp asterisk /tmp/test1.call chown asterisk /tmp/test1.call mv

Greetings all, This is my first post to this forum, so please be kind. I am looking to integrate Asterisk as our primary voice mail server (IVS) for our NEC NEAX 2000 IVS. We currently use a 6-port Mitel system. The two communicate via analog TDM ports and a MCI serial connection. Is anyone familiar with setting up a NEAX 2000 with Asterisk as an IVR server? I know I would need six FXO and

I would like to send the CDR records from all our machines around the world to a single database. But I need the hostname included with each record for monitoring purposes. Is there a better way than using the userfield and adding SetCDRUserfield for every call to set the userfield to the name of the host? Thanks...