similar to: AMI potential memory leak

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 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]

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 processes the dial plan so even if second line was hello-world2 it would not matter. [mycontext] exten =>

Hello. I've experience a rather surprising behaviour of the AMI 1.1 > Event: Newstate^M > Privilege: call,all^M > Channel: SIP/XXXXXX-089c63b8^M > ChannelState: 4^M > ChannelStateDesc: Ring^M > CallerIDNum: XXXXXXXX^M > CallerIDName: YYYYYYYYY^M > Uniqueid: 1256089773.59^M Usually ChannelStateDesc gives me 'Ringing' but sometimes it only gives me

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

In 1.4.43 I would see things from "core show channels" like DAHDI/18/xxxxx for line 18 in Asterisk 11 its DAHDI/i4/xxxx How do I get the line number back? Jerry

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

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

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

I use Jack for getting callee sound. Dial with option M(): JACK_HOOK(manipulate,i(rec_737219:input),o(rec_737219:output),c(rec_737219))=on This works fine, but I need to connect the sound channel to Jack *before* the actual answer. As you can see in the AMI log, between "Ringing" to JACK_HOOK there is a 6 second break. I don't want that. I need a way to launch Dialplan function

Hi to all. I have a problem in the AMI. Sometimes the AMI don't generate the event NewState when the exten of destiny is Ringing and sometimes don't show me the callerid in this events. The event NewState what i refer: Event: Newstate Privilege: call,all Channel: SIP/17-00006fd6 ChannelState: 5 ChannelStateDesc: Ringing CallerIDNum: 4191920902 CallerIDName: 4191920902 Uniqueid:

Hi I am using following say.conf file. Its a default file, which comes with Asterisk installation. When I call SAY DATETIME AGI function, it simply returns without playing date & time. Where as if I use mode=old setting, it works. Is this a bug or mode=new is not implemented for SAY DATETIME AGI function? [general] mode=new ; method for playing numbers and dates ;

Dear all, I have this setup on Ubuntu 12.04 and using puppetmaster/puppet 2.7.14 and puppetdb/puppetdb-terminus 0.9.0 from puppetlabs. My puppetmaster also run puppetdb. I also use hiera in this setup. hadoop4 is puppetmaster and hadoop02 is puppet client. puppet node status hadoop4.west.net hadoop4.west.net Currently active Last catalog: 2012-06-05T23:23:33.159Z Last facts:

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,

Issuing the AMI Status command results in a list of active channels. But how to figure out which channels are related before the call is answered? 2 channels below are somehow associated, but how can I be 100% sure they are related in order to implement a redirect of the incoming call to another phone ("attended" call pickup respecting call/pickupgroups). Uniqueid seems to be a

I am trying to set add a SIP Header to a call before adding it to the Queue. The dial plan sends the call to my macro to perform the work. When I use chan_sip, everything works as expected. When I use PSJIP support, it's not adding the SIP header. Looking at the output, I see the macro is called in both cases. In the PJSIP case, the added sip header never is showing up in the asterisk logs

<!doctype html> <html> <head> <meta charset="UTF-8"> </head> <body> <div> <br> </div> <blockquote type="cite"> <div> On 25 November 2018 at 06:29 Joan Moreau < <a href="mailto:jom@grosjo.net">jom@grosjo.net</a>> wrote: </div> <div>

On Fri, 23 Nov 2018 10:45:56 -0500, Brad Smith stated: >On 11/23/2018 9:31 AM, The Doctor wrote: > >> On Fri, Nov 23, 2018 at 04:06:53PM +0300, Odhiambo Washington wrote: >>> On Fri, 23 Nov 2018 at 15:29, Timo Sirainen <tss at iki.fi> wrote: >>> >>>> https://dovecot.org/releases/2.3/dovecot-2.3.4.tar.gz >>>>

It's still missing core dump (or bt full from it) Aki On 27.11.2018 8.39, Joan Moreau wrote: > > Thank you Aki > > here the requested data (below) > > Please not as well that we have numerous subfolders (>50) and pretty > big mailbox sizes (>20G) > > Bug appears mostly in auth process and index-worker > > > dovecot -n : > > # 2.4.devel

See https://dovecot.org/bugreport.html#coredumps Without a backtrace it's not really possible to figure out where it's crashing. > On 28 Nov 2018, at 13.20, Joan Moreau <jom at grosjo.net> wrote: > > Where to get that ? > > > On 2018-11-27 08:50, Aki Tuomi wrote: > >> It's still missing core dump (or bt full from it) >> >> Aki