similar to: Media flow between them

I'm having an issue with CDR. Basically, I expect to have all "legs" of a call having the same linkedid and differing only by the sequence value. That does happen, but I'm getting null dst values after doing an attended transfer. I'm not sure if this is a bug or I'm doing something wrong. I'm running Asterisk 13.2.0. Here's the console log, step by step: First,

Hello a call goes out and is answered : [Sep 17 11:29:52] VERBOSE[23420][C-00000051] app_dial.c: SIP/myprovider-0000010b is making progress passing it to SIP/mysippeer-00000108 [Sep 17 11:30:05] VERBOSE[23420][C-00000051] app_dial.c: SIP/myprovider-0000010b answered SIP/mysippeer-00000108 [Sep 17 11:30:05] VERBOSE[23522][C-00000051] bridge_channel.c: Channel SIP/myprovider-0000010b joined

NAT endpoint calling local endpount - switching to native_rtp then no audio, both of them have direct_media=no, Verbose log: -- Executing [99 at dialmap:1] AGI("PJSIP/304-00000022", "/pbx/agi.php") in new stack -- Launched AGI Script /pbx/agi.php -- AGI Script Executing Application: (Dial) Options: (PJSIP/99/sip:99 at 192.168.1.73:5060,20) -- Called

Hullo Everybody ; I am failing to start my new VM and it tells me unkown bootloader . Below are my parameters and output . Could somebody be having an idea of putting right this problem? [root@virtualintranet /]# xe vm-param-list uuid=70645ba3-bcbc-683b-099e-ed197301fcc2 uuid ( RO) : 70645ba3-bcbc-683b-099e-ed197301fcc2 name-label ( RW): PVG1

    Some users have complained that their calls drop after about 30 seconds.  Not all, just some.  After looking at the log files the only difference I can find from the dropped calls is the following line: [2020-09-07 11:29:59] VERBOSE[21666][C-00000055] bridge.c: Bridge 14410400-5e04-4358-af0c-45fd71f6f5cd: switching from simple_bridge technology to native_rtp     Most calls just do:

Hi, with canreinvite=no and directmedia=no I and getting the message in the logs for all calls "switching from simple_bridge technology to native_rtp" -- Executing [102 at mkg:1] Dial("SIP/101-00000017", "SIP/102") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/102 -- SIP/102-00000018 is ringing -- SIP/102-00000018 answered SIP/101-00000017

Hey guys, have issues with reinvite, no matter what endpoint is calling asterisk always tries switch simple_bridge to native_rtp Bridge 0422bfa0-9d22-4bba-9108-a3f14d7d1cab: switching from simple_bridge technology to native_rtp in endpoints table ?direct_media? sets to ?no? on all endpoints but it doesn?t help. if native_rtp not work for some reason I have oneway audio. how can I fix this?

Hello! Just installed asterisk 13.2.0 and see many such messages in log, I see them in console during calls, really something like this: -- Executing [6166 at kanbaikal:2] Dial("OOH323/kanbaikal-6", "SIP/6166 at asterisk") in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called SIP/6166 at asterisk > 0x7fa9d4007660 --

On 9/12/18 1:22 PM, Joshua Colp wrote: > On Wed, Sep 12, 2018, at 2:19 PM, sean darcy wrote: >> I understand that HangUp() hangs up the calling channel. I want to >> hangup the called channel. >> >> SIP/mycall-xxxxx calls and bridges with DAHDI/1-1. >> >> I send SIP/.... to listen to a long, very long, file. > > Define "send". How are you

ok switching to "Console/default" does show the text --- <("<) --- Call to device 'default' on console from 'default' <2564286000> --- (>")> --- --- <("<) --- Auto-answered --- (>")> --- However I don't hear any audio. Thanks Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am banging my head over a simple asterisk trick I was seeing on one asterisk server. An extension dials an international premium number, the called number answers, then the extension hangups, but the call continue to run on the international number side, generating an high profit for the premium number company and a big loss for the asterisk owner. I think some sort of "transfer"

No, there is no Music On Hold starting and the bad thing is the call duration reported by asterisk was just few seconds while the call duration reported by the provider was few thousand seconds, the max allowed. So they will be able to terminate the call on the asterisk side and have it run on the provider side. Leandro 2016-09-15 19:18 GMT+02:00 Max Grobecker <max.grobecker at

Well, it breaks audio for all NAT endpoints, how can I fix this? > On 18 Mar 2015, at 15:48, Matthew Jordan <mjordan at digium.com> wrote: > > On Wed, Mar 18, 2015 at 7:41 AM, Nick Awesome <jleed at me.com <mailto:jleed at me.com>> wrote: >> Hey guys, >> >> have issues with reinvite, no matter what endpoint is calling asterisk >> always tries

05.03.2015 11:42, Dmitry Melekhov ?????: > 05.03.2015 11:29, Dmitry Melekhov ?????: >> Hello! >> >> Just installed asterisk 13.2.0 and see many such messages in log, I >> see them in console during calls, really something like this: >> >> >> -- Executing [6166 at kanbaikal:2] Dial("OOH323/kanbaikal-6", >> "SIP/6166 at

Hello! I need to use n-way call as it described here: http://habrahabr.ru/sandbox/52259/ It is in russian, but dial plan is quite clear. It works in asterisk 11: -- Blind transferring OOH323/7272-6385 to '0' (context fromtransfer) priority 1 -- Executing [0 at fromtransfer:1] NoOp("OOH323/7272-6385", "") in new stack -- Executing [0 at fromtransfer:1]

Ok, commented out that line. It's still doing it. Reloaded dialplan. Please don't tell me I have to restart asterisk. Thomas M. Peters | Systems Administrator | tpeters at mcts.org Desk: 414.343.1720 | Helpdesk: x3400 or helpdesk at mcts.org >>> Tiago Geada <tiago.geada at gmail.com> 6/26/2015 12:07 PM >>> messages => error states to log error messages to

Léandre BASSOLE PhD Student CNRS-CERDI 65 Bd Francois Mitterrand Boite Postale 320 63009 Clermont-Ferrand CEDEX 1 FRANCE Tel : +33 4 73 17 74 45 Fax : +33 4 73 17 74 28 ----- Forwarded Message ---- From: Leandre Bassole <leandrebassole@yahoo.co.uk> To: r-help@r-project.org Sent: Saturday, 27 October, 2007 8:41:05 PM Subject: Hi all!! I am a new user of R. I am very familar to Stata, but

Hi again! I tried to attach two VoIP-phones to my new Asterisk 13.14.1 on a Banana PI with Armbian/Debian 9. First test was to call a test service that say the time. Works! Second test was to record my voice and play it again. Works! Third test was to call the other VoIP-phone. It does NOT work... :( Then I noticed that, by starting, Asterisk says the following messages: [Feb 15 18:42:54]

n 20/07/2020 14:19, Nick Howitt via samba wrote: > > > On 20/07/2020 11:14, Rowland penny via samba wrote: >> >> >> I have reviewed all the posts in this thread and I 'think' I know >> what is going on and also answers a question I asked. >> >> You have in your smb.conf: >> >> unix password sync = Yes >> >> This possibly

On 20/07/2020 10:37, Nick Howitt via samba wrote: > Bump, please. I have reviewed all the posts in this thread and I 'think' I know what is going on and also answers a question I asked. You have in your smb.conf: unix password sync = Yes This possibly means that you have a group in /etc/group called allusers with the ID of 63000 I would replace the line with: ldap password sync