similar to: Originate with Local channel to any app-only extension hangs up immediately?

Hi All, I am in the process of migrating from 1.4.20 to 1.6.2.x and have stumbled across a number of "differences" between the 2 versions that are forcing me to use local channels in my dialplan (mainly centered around the different behavior of CDR fields in the 2 versions) . Previously, I would place a call via an AMI Originate action similar to: action:.Originate..

I have an app which handles a Mitel's command port to change the MWI lights. It detects dial tone, plays some DTMF digits, listens for dialtone-or-busy, does a manager event on what it finds, and returns. Since the Mitel command port does not give answer supervision (looks like it's ringing), and since I run this app via a AMI "originate" command, I set up an extension in

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

Hello, I'm giving hangup-handlers a try on a new Asterisk 11.2.1 setup. My plan is to use this handler to update my CDRs with values such as Asterish and Tech cause (see function HANGUP_CAUSE). I want to have my custom hangup-handler be run only once and when "the second channel" hangs up. At the moment, I'm issuing a couple of "channel originate Local/1 at mycontext1

Hello, I want originate a call to some destination, and when B side answes to play a prompt. Asterisk version is 1.6.0.5. But also I need to insert a SIP header to Invite, that's why I'm using Local Channel. This is my extension.ael: context autodialer-local { _X. => { SipAddHeader(P-Asserted-Identity: <sip:${CALLERID(num)}@xxx.xxx.xxx.xxx;user=phone>);

2013/3/26 Richard Mudgett <rmudgett at digium.com> > > On 03/25/2013 05:17 PM, Olivier wrote: > > > Hello, > > > > > > I'm giving hangup-handlers a try on a new Asterisk 11.2.1 setup. > > > My plan is to use this handler to update my CDRs with values such > > > as > > > Asterish and Tech cause (see function HANGUP_CAUSE). >

I have been trying to get a DIALSTATUS output from a call started with originate. I searched a fair bit and have found several references to using local channels to do this. However, I could not find enough of the specifics to get it working myself. What I need to do is dial a zap channel and run various scripts if the channel is answered, busy, no-answer,etc. Here is the dial plan I am

I would be grateful if someone could tell me where to find the docs to get started on the following problem: A program needs to be written to place a SIP call to a certain extension on another Asterisk system, and see whether the call state ratchets up to "ringing", then drop, and take action on the results. Can anyone refer me to the appropriate starting point to read up on this?

Hi all, I'm trying to use the CDR() function on the "Variable" header of the Originate AMI action, but it isn't working. Anyone knows anything about this problem? asterisk 1.4.26 Thanks, Gabriel Ortiz -------------- next part -------------- An HTML attachment was scrubbed... URL:

hello List, i am in a situation where i cannot get cdr records for call originated from CLI , i am not able to get when i used application or extension. is there any solution regarding this ,i working since last 3 days onto this. regards Dhaval -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am trying to make a data channel using ISDN and i need to set the caller id num field. Can any body tell me how i can set the caller id field since i notice in chan_dahdi.conf callerid field doesn't work with channel originate. Thanks, Pawel -------------- next part -------------- An HTML attachment was scrubbed... URL:

Here's my cmd: originate MOTIF/8447/+12122064431 at voice.google.com extension s at greeting Greeting: [greeting] exten=> s,1,Wait(2) same=>n,Background("hello") same=>n,Wait(3) I can see the call go out (also in, since testing on one our own numbers), but [greeting] never executes. I'm expecting to see that when the motif call comes in and is answered,

We relay on 'failed' extensions after AMI ORIGINATE command. When moving from Asterisk 1.8.22 to Asterisk 13.2, it has stopped to work. I belive that it is due to a change in pbx.c => ast_pbx_outgoing_exten. Thanks, Valter -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, is there a way to know if originate call channel ended the call *before* connecting to context/extension/priority? DIALSTATUS is empty, HANGUPCAUSE is always 16, nothing in SIP Headers nor in AST_CONTROL_FRAME_[HANGUP|ANSWER] Asterisk is 1.6.2.16 Thanks for any hint -- Daniel

Hi guys I seem to be having a problem, I don't know if it's a bug or whether I'm just doing it incorrectly. I want to set about 3 channel variables when I originate a call via AMI. All the documentation I have found says to do it like this: Variable: variable1=value|variable2=value|variable3=value However when I do this it runs them all together and I end up with:

As far as I can tell, after discussing this matter with other asterisk users in my area, my telco _does_ provide disconnect supervision.. It seems that the problem is actually related to the Sangoma A200 card I'm using, as two other people both using this same card have expressed the same problem.. Are there any other users on this list using the Sangoma A200 FXO port card, and experiencing

Hi, I would like to originate my first call from CLI. As I'm new to this, I'm wondering if it's possible. When I type "originate" from CLI, I've got this : " There are two ways to use this command. A call can be originated between a channel and a specific application, or between a channel and an extension in the dialplan. This is similar to call files or the

Hello, I'm using an AMI Originate command to send a fax. The fax is sent by a script, and I'd like my script to send the fax, wait until it has succeeded or failed, then exit with an appropriate error code (it is driven by a mail system, so the exit code will tell the mail system whether to retry the fax later). The script works great if the fax succeeds, or if the line is busy or

Hello, I am using asterisk manager interface (http) for originating calls. How can I get the name of the channel which is created by originate? I want to use this channel for other manager commands like Atxfer, Monitor, Hangup etc. If I do action=originate, channel=SIP/200 then it creates a channel like 'SIP/200-0865ff80' which I can see in the asterisk console using "core show

Everyone - thought I would pass on a useful piece of information. Finally got a solution to my phantom ringing problem. Problem - the zhone is triggered into detecting ringing by the Automatic Line Insulation Tests (ALIT or LIT) run nightly automatically by the telco. Here it is twice between 8pm and 9pm on my particular lines. My first approach before I knew specifically the buzzword for