similar to: how asterisk detects silence?

We have a system where two calls are in a ConfBridge with recording. This is Asterisk 16.3.0 Channel A seems to work perfectly. Wireshark is showing the RTP to/from working fine and having no jitter/lag issues. This call hears everything from channel B. Channel B we have more issues capturing a wireshark trace because their channel can be in the system for hours. When the two calls are in the

19.03.2015 09:31, Dmitry Melekhov ?????: > Hello! > > As I see there is dsp_drop_silence switch in confbridge. > Could you tell me how asterisk detects silence? > Is it possible to change silence level, > so, let's say some not loud enough background noises will be > recognized as silence > and only loud enough human voice will be recognized as sound? > > Thank

Using asterisk 16.1.1. I'm setting up a test using the cmp2k (Cyber Mega Phone 2K Ultimate Dynamic Edition). I have noticed Chrome 72 had some issues with video streams. I just upgraded to Chrome 73 and see they still have some issues. If I have 2 calls in a confbridge with video set to none. I then set the video source to a Chrome browser and the Remote Video shown to both calls from

All; What I want to do is create a way to easily send callers into a conference room to have an N-way conference call. I created an extension '100' that calls the MeetMe() command. Then all I need to do is transfer a caller using a blind transfer (*2 in my case) to extension 100. Then I can dial a feature code that sends me into that conference (*15 in my case). So far, a piece of

Hi All, I want to set the video_mode of the confbridge dynamically in the dialplan. SO say if 5 users join the conference with follow_talker mode, it should work like that (and it does). But if 6th user changes the video_mode to first_marked and gets marked in the dial plan and joins the conference, he does not become the single video source of the conf. The video mode stays follow_talker. I

Hi all, I'm looking for some serious help. :) I couldn't find a better description for my problem... I think it is quite complex! Here's what I would like to achieve: A SIP caller dials into to my Asterisk 10. He will automatically listen to a specific MP3 stream. Other SIP callers dial also into my Asterisk. They all will automatically listen to the same MP3 stream. All

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! We tried to move Windows 2003 VM with e1000 driver from Centos 7 which runs qemu-kvm-0.12.1.2-2.491.el6_8.7.x86_64 to Centos 7 with qemu-kvm-ev-2.6.0-28.el7_3.6.1.x86_64 and we got problems- tcp sessions, namely smb connections, randomly drops. We didn't test previous qemu-rhev with this VM, so we don't know how it works in them. Could you tell me is this known problem? Any

Hello! I need to do rsync between local drives on win32. Rsync works, but it never ends. I started it with -vv and it writes something like: total: matches=0 tag_hits=0 false_alarms=0 data=0 And then it stays forever. :-( Could you help me?

Hello! I want to create BDC with smbpasswd backend, just because I run ldap master on the same machine as PDC and I don't think that using ldap backend will be far better for me. Only thing I don't understand: I read in howto: <quote> Machine Accounts Keep Expiring This problem will occur when the passdb (SAM) files are copied from a central server but the local Backup

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 --

Hello! I want to start rsync on w2k as service. If I try to start rsync from command line it simply do nothig: $ rsync --daemon Administrator@dm-w2ks /usr/bin $ ps PID PPID PGID WINPID TTY UID STIME COMMAND 480 1 480 480 con 500 04:15:03 /usr/bin/bash 1428 480 1428 1420 con 500 05:26:46 /usr/bin/ps Administrator@dm-w2ks

Hello! Installed packages from SIG on centos7 , at first start it works, but after restart- not: ?glusterd2 --config /etc/glusterd2/glusterd2.toml DEBU[2018-04-04 09:28:16.945267] Starting GlusterD???????????????????????????? pid=221581 source="[main.go:55:main.main]" version=v4.0.0-0 INFO[2018-04-04 09:28:16.945824] loaded configuration from file???????????????

Hello! I have following connections over isdn pri: avaya definity---pri--asterisk--pri-panasonic 500 Just because panasonic 500 can't send user's names. I also want to have reverse callerid for avaya users. But if there is no answer in dial plan: exten => _XXXX,1,Set(CONNECTEDLINE(name)=${DB(names/${EXTEN})}) ;exten => _XXXX,n,Answer exten => _XXXX,n,Dial(DAHDI/g4/${EXTEN})

Hello! Just got new server with TE420. Not all four spans will be used immediately, but spans not configured or not connected blink red light. Is it possible to turn span off, so my colleagues will not eventally tell me that something is wrong with asterisk? :-) Thank you!

18.08.2017 12:21, Atin Mukherjee ?????: > > On Fri, 18 Aug 2017 at 13:45, Raghavendra Talur <rtalur at redhat.com > <mailto:rtalur at redhat.com>> wrote: > > On Fri, Aug 18, 2017 at 1:38 PM, Atin Mukherjee > <amukherj at redhat.com <mailto:amukherj at redhat.com>> wrote: > > > > > > On Fri, Aug 18, 2017 at 12:22 PM,

Hey all, Just wondering about the current status of wrepld. Back in '02 Jean Fran?ois Micouleau was doing some work on it, but there does not appear to be any changes to the code since that spring besides updates to the entire tree that happen to touch the wrepld files. The Samba-HOWTO states that it is in "active development" and I was wondering if I could help with testing or

Can it be that our accounting packages don't work on SMB and only with NT? I would HATE to put a NT Box into the mix. Ruben -- Brooklyn Linux Solutions http://www.mrbrklyn.com http://www.brooklynonline.com 1-718-382-5752

We have a product that uses Asterisk via AMI. I am relatively certain we used to be able to play prompts by actions like the following to make asterisk play the confbridge-join prompt when a new user joins the confbridge. However, that doesn't seem to work now. Action: SetVar ActionID: C58 Channel: PJSIP/1003-00000003 Variable: CONFBRIDGE(bridge,sound_join) Value: en/confbridge-join Does

09.09.2016 13:45, Joshua Colp ?????: > Dmitry Melekhov wrote: >> Hello! >> >> >> Upgraded 13.10 to 13.11.1 today and now I see messages in log: >> >> >> [Sep 9 12:23:16] NOTICE[6064] res_pjsip/pjsip_distributor.c: Request >> 'REGISTER' from '"3563" <sip:3563 at 192.168.32.254>' failed for >>