similar to: Is it possible to use Stasis to control both legs of a Local channel created using ARI?

HI Matt, I am trying to replicate this particular problem. We are seeing more frequently where the Event: AsyncAGIExec is never being sent. The two scenarios I have seen in tests yesterday and today... We sendl an AMI action. For example, play a short file or hangup. AMI Events will indicate it did the work, but we never receive the Event: AsyncAGIExec with a result at all. Asterisk debug

Hi I am experiencing Asterisk Crash. Log got stopped when asterisk crashed. Please help me to identify the reason and fix this issue. Asterisk: 1.8.20 I am using AMI and fastAGI to control the call. Some part of dial plan is also defined in extensions.conf I am experiencing this crash when app_meetme conference functionality is used with more than 3 parties. I faced this issue with

Hi Team, Thanks for your great job an Asterisk new features developments. I installed asterisk-12 Beta and found some changes as well which i notice to put in-front of your knowledge, don't know that bug of new modification into objects or old version (asterisk-11) mistake corrected that time, anyway *Asterisk-12:* Array ( [Event] => ConfbridgeMute [Privilege] => call,all [Conference]

Is it possible to simply send a message to appear as an AMI message/event, from the dialplan? For example exten =>123,1,ami(myEvent, param1, param2) and in the AMI a message appears like: Event: myEvent Privilege: call,all Channel: PJSIP/misspiggy-00000001 Uniqueid: 1368479157.3 ChannelState: 3 ChannelStateDesc: Up CallerIDNum: 657-5309 CallerIDName: Miss Piggy

On Thu, Aug 6, 2020 at 1:52 PM Dan Cropp <dan at amtelco.com> wrote: > I understand how to control the first local channel, but an having trouble > getting the second local channel to enter stasis. > > > > I setup have the following extensions.conf to handle 1000 (basically had > it setup so if first stasis not there try second, but believe second > channel never

We are communicating with Asterisk via AMI. Running Asterisk version 13.18.5 on an Ubuntu box. If you look at the event response, the Result field is filled with random characters. I'm not sure what to do because that is a completely random result. It makes no sense. We send the following command to asterisk via AMI Action: AGI ActionID: C44415 Channel: SIP/192.168.40.105-00001338

Hello, I am using Asterisk 11.0.1 and do not notice any changes regarding the Transfer on newer Asterisk 11.x versions. I am using AMI and controlling a channel via AsyncAGI. I send a Transfer cmd (such as the following) Action: AGI ActionID: C8 Channel: SIP/1004-00000002 CommandID: C8 Command: EXEC Transfer SIP/1003 Destination phone starts ringing. If it answers the

Hello List, Good day, We have an application, where we redirect a channel to meet me. Sometimes the channel is getting hanged up by Asterisk, and we get an hang-up event. Please reply back, if any one faced such issue. Here is the hangup event info, -HANGUP {calleridname=<unknown>, connectedlinename=<unknown>, uniqueid=1384413814.79523, cause=0,

Please help find the cause of strange behavior res_pjsip. Making outgoint call to other sip server (CommuniGatePro), my asterisk suddenly sends BYE after picking up! Partial log of an outgoing call with full debug is attached and on web: http://pastebin.com/tLNCpx4d No diagnostic messages why asterisk suddenly decided to hangup i don't found :( There are suggestions or strong belief

HI, Is there anybody using package maps, available at ftp://ftp.mcs.vuw.ac.nz/pub/statistics/map/ I have installed it (version maps_0.1-3.tar.gz wich seems to be latest, on R 1.5.1, linux), but whatever I do, I get a segfault. Discussion in the mailing list suggests that the package is not finished yet but is still usable, at least on some platforms. Does anybody has experience with similar

An additional follow-up question, if I need to set the P-Asserted-Identity on the create (originate), is there a way to do this with ARI? From: asterisk-users <asterisk-users-bounces at lists.digium.com> On Behalf Of Dan Cropp Sent: Friday, August 7, 2020 11:51 AM To: 'asterisk-users at lists.digium.com' <asterisk-users at lists.digium.com> Subject: [asterisk-users] With ARI,

Hi Jöran, Would it be possible to see an example using curl of how you are passing the PAI Header through ARI create? Dan From: asterisk-users <asterisk-users-bounces at lists.digium.com> On Behalf Of Jöran Vinzens Sent: Friday, August 7, 2020 12:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] With

## Hello.. I have asked a similar question, but this is not fixed as before. ## I am running the following using Ubuntu OS: R version 2.13.1 (2011-07-08) Copyright (C) 2011 The R Foundation for Statistical Computing ISBN 3-900051-07-0 Platform: x86_64-pc-linux-gnu (64-bit) ## when I do this: > install.packages("mapproj", dependencies=T) ## I get this: Installing package(s)

I'm trying to transition from AMI to ARI. Running into a small hiccup when I try to create (originate a call) with the caller id name and number I can pass the Name and Number if the name has no spaces in it and it shows up in my PhonerLite application. curl -v -u asterisk:asterisk -X POST http://asterisk:astersk at localhost:8088/ari/channels/mycallerid.1?endpoint=PJSIP/1003 at

Hi Dan, i did it wrong, sorry: curl -v -H "Content-Type: application/json" -u asterisk:asterisk -X POST " http://localhost:8088/ari/channels/newChannelId" <http://localhost:8088/ari/channels/1400609726.3/play?media=sound:hello-world> --data '{ "endpoint": "SIP/Alice", "variables": { "CALLERID(name)": "Alice" ,

Hi Dan, as far as PPI and PAI Header, we use the channel Vars in order to do that. In Latest Asterisk you can set Channel vars within the create command in the Body. Just set the PJSIP(add,PAI) as you would do it in Dialplan. https://blogs.asterisk.org/2020/07/22/ari-create-channel-with-variables/ BR Jöran On Fri, Aug 7, 2020 at 7:06 PM Dan Cropp <dan at amtelco.com> wrote: > An

Hi, I installed the maps and mapdata libraries on my R-1.8.0 on MacOSX 10.2.8 (jaguar on a powerbook G4), and failed to make the map function work properly: R : Copyright 2003, The R Development Core Team Version 1.8.0 (2003-10-08) [...] > library(maps) > map() Error in file(file, "r") : unable to open connection In addition: Warning message: cannot open file

Hi Dan, i would do something like this (it is not a copy of what we are doing but an example of how i would do it) Important here is the "--data" and "-H" Option as well as the "variables" section within the Body. I added the callerid function here as well as it is the sample in the asterisk wiki. curl -v -H "Content-Type: application/json" -u

The original S maps package, previously only available in R on Unix systems is now available in R for Windows. maps_2.0-2 and mapdata_2.0-2 (see below) in both source and Windows binary form (for rw1071) have been uploaded to CRAN, hopefully soon to appear. The key aspect of this package which differentiates it from other GIS related R packages under development is that it is standalone within

The original S maps package, previously only available in R on Unix systems is now available in R for Windows. maps_2.0-2 and mapdata_2.0-2 (see below) in both source and Windows binary form (for rw1071) have been uploaded to CRAN, hopefully soon to appear. The key aspect of this package which differentiates it from other GIS related R packages under development is that it is standalone within