similar to: OriginateResponse 'reason' property.

Displaying 20 results from an estimated 20000 matches similar to: "OriginateResponse 'reason' property."

2010 Dec 01
1
Reasons of OriginateResponse
Good morning everyone. I wonder where I can find a list of the reasons the event OriginateResponse. I found this list [1]. But in my Asterisk has other reasons too. [1] 0 = no such extension or number 1 = no answer 4 = answered 8 = congested or not available Thanks in advanced, -- Rodrigo Lang Opening your mind - Just another Open Source site<http://openingyourmind.wordpress.com/>
2009 Oct 05
3
OriginateResponse Event
Hi people, I'm executing some parallel Originate async, is there a way to know the result of each originate?... I was looking at the OriginateResponse event, but I don't know how to get it from my web service. Also, if I have 3 originate in parallel, how can I identify the OriginateResponse of each one? Thanks in advance... Anahi Ludue?a
2006 Oct 13
3
OriginateEvent reason codes.
Hi. I'm making calls via the Manager OriginateAction. My action is set to be async and therefore I receive originiate events. Within the originate event that I receive there is a reason code. In the event of failure I need to dermine why the call failed (no pickup, rejected, no such number, circuit busy, ect) and inform the user with a meaningful message. I assume that one is suppose to
2009 Jul 29
1
Matching Originate action with its NewChannel event
An application commanding asterisk with AMI is going to launch lots of concurrent calls in very few seconds using the Originate AMI command but it's also going to need to be able to cancel very quickly any call of them even before each OriginateResponse event comes in. All the calls will be done by the same trunk (a trunking enabled channel). But there's a problem for canceling any call:
2013 Aug 22
2
How to get the original SIP result code
B.H. Hello, i'm using AMI Originate action (with async=true) to send outgoing calls to a SIP trunk (using asterisk-java library to connect to AMI). The problem is that in case of failed originate, OriginateResponse event is returning only the reason code which is sometimes not sufficient to determine the real cause of failure. Also, there's no way to link between the channel that was
2015 Aug 06
2
Asterisk uses "Anonymous", but why?
On Thu, Aug 6, 2015 at 1:25 PM, Murthy Gandikota <murthy64 at hotmail.com> wrote: > > > ________________________________ > > Date: Thu, 6 Aug 2015 12:55:28 -0500 > > From: rmudgett at digium.com > > To: asterisk-users at lists.digium.com > > Subject: Re: [asterisk-users] Asterisk uses "Anonymous", but why? > > > > > > > > On
2015 Aug 06
3
Asterisk uses "Anonymous", but why?
On Thu, Aug 6, 2015 at 12:33 PM, Murthy Gandikota <murthy64 at hotmail.com> wrote: > > > ________________________________ > > Date: Thu, 6 Aug 2015 12:07:35 -0500 > > From: rmudgett at digium.com > > To: asterisk-users at lists.digium.com > > Subject: Re: [asterisk-users] Asterisk uses "Anonymous", but why? > <snip> > >> Here
2007 Aug 13
0
Originate and tracking
I am originating calls through the Manager Originate API command. I can track failures (through the OriginateResponse event) I can track answered calls through the OriginateResponse event) There may be occasions where I need to cancel some outbound calls whilst they are ringing. Here's my problem: How do I know what the channels are in order to cancel them ? I can get a
2009 May 12
2
Asterisk Manager API Action Originate
Has anyone else had issues with Originate returning the wrong error code? According to the docs, the following errors are supposed to be returned: 0 = no such extension or number 1 = no answer 4 = answered 8 = congested or not available Now in Asterisk 1.4.23 I get some error code 5's but since they're so few I tend not to worry. But what is concerning is the number of Error 0's I
2010 Nov 24
0
Originate Response.
Hi to all. I am conducting several tests with the Asterisk manager and I noticed something that I believe to be a problem. When I generate a call with the Action Originate with the Async option true, the event OriginateResponse returns normally. But when I generate a call in the same way, without the Async option, the event OriginateResponse does not come. Is this a bug? It was fixed in some
2010 Oct 21
1
How to kill AMI ORIGINATE on-the-fly
My application fires several calls thru AMI ORIGINATE command. For example if I have 3 operators I do 3 ORIGINATEs. My trouble is when one operator quit for some reason, I should kill the corresponding ORIGINATE. Of course, I could let the call ring and hangup after the customer pick-up. But this is not the case, I do have to kill the corresponding ORIGINATE. I could execute a soft hangup,
2007 Sep 26
1
Manager Originate Action and Cancel
I'm using the Originate Action on the Asterisk Manager to place calls between two extensions in async mode. Is there any way to cancel the Originate Action before I get the OriginateResponse action? I'm unable to perform a Hangup because I can't know the channel name before I get the response... thanks in advance! -- santiago aguiar *netlabs* / Palmar 2548 Montevideo, Uruguay Tel.
2006 Feb 27
1
Problems dialing to another Asterisk server
Hi, I have a problem dialing a SIP phone which is logged in as different Astesrik machine from the one I am working with. I want to call a phone in Another astersik machine in , if it answers, calling a SiP phone registered in my ASterisk: My dialplan is: [mariaSIP] exten => _1.,1,Wait(1) exten => _1.,2,Dial(SIP/6021@192.168.0.51:5038,20) exten => _1.,3,HangUp() exten =>
2010 Oct 01
2
AMI Originate
When calling Originate Action, it rings my phone. If I do not answer, I receive a Channel Event: Hangup, followed by receiving an OriginateResponse event with a Failure Response, Reason 3. My phone continues to ring. If I answer the phone at this point, it attempts to answer, but does not succeed. Looking at the asterisk debug, it appears to destroy the SIP dialog for the call. It also
2009 Oct 01
3
What are the reasons for VoIP echo?
I have an Asterisk 1.4.2 system that has been installed for about 3 months now in our home. We converted all of our phones to SIP phones, and use two different trunk providers (BroadVoice for incoming & FlowRoute for outgoing). Most of the time its working flawlessly. But about 1/3rd of the calls that come into us complain of an echo and what is best described as latency issues. Its
2007 Apr 18
0
Phones working with 1.2.17, not with 1.4.2
Hello, I've got various phones (mostly SPA-922) behind NAT registered to Asterisk. I've set nat=yes and canreinvite=no, and everything seemed to work great with 1.2.17. After upgrading to 1.4.2 using users.conf and macro-stdexten my spa-922 can't call other extensions. -- Executing [23@default:1] Macro("SIP/22-b72006f0", "stdexten|23| SIP/23") in new stack
2005 Sep 15
3
${DIALSTATUS} problems
Hi. I'm dialling two numbers - one that's unobtainable, one that's busy. ${DIALSTATUS} is coming back ANSWER each time right before the channels hang up. Am using the following dialplan macro to dial out. [macro-advdial] exten => s,1,Dial(${ARG1},20,g) ; Ring the interface, 20 seconds maximum exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status
2005 Sep 14
0
Dial Application Return Codes - Help needed
Hi. I'm dialling two numbers - one that's unobtainable, one that's busy. ${DIALSTATUS} is coming back ANSWER each time right before the channels hang up. Am using the following dialplan macro to dial out. [macro-advdial] exten => s,1,Dial(${ARG1},20,g) ; Ring the interface, 20 seconds maximum exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status
2010 Apr 18
1
problems originating an outgoing IAX2 call
Dear all i'm trying to originate an outgoing call with the command originate, from Asterisk's CLI i'm typing: CLI> originate IAX2/my-iax-provider/number2call application wait 10 [Apr 18 19:31:12] DEBUG[32331]: chan_iax2.c:4000 create_addr: prepending 40 to prefs -- Call accepted by 62.149.202.150 (format ilbc) -- Format for call is ilbc -- Hungup
2008 Feb 14
1
Error checking asterisk method - suggestions?
Hi there dear users and dear developers of Asterisk! I've got a maybe strange idea, let's see if you laugh or think it's reasonable J I'm using Asterisk with Digium TDM800P cards with 24 lines (as an answering machine). Each analog line can be reached through a phonenumber, since they are each connected to my telephone provider. Yes expensive I know J The challenge: I'd