similar to: Asterisk 11 under VMware?

Hi, I spent the past week experimenting with webrtc + asterisk 11.9.0-rc1 + opus/vb8 codec patch. This is interesting technology and I try to find out how to connect all the moving parts. Firefox: Neither sipml5 or jssip works with calls to asterisk, audio/video doesn't matter. WARNING[977][C-00000005] chan_sip.c: Rejecting secure audio stream without encryption details: audio 35684

On 4/5/16 3:17 PM, Joshua Colp wrote: > Carlos Chavez wrote: >> I am currently having a voice quality problem with one of our Asterisk >> servers. We have checked the network and we have found no problems that >> could cause the voice to sound cracked and with small interruptions. I >> am looking at the timing source for Asterisk and it is currently using >>

Hi, I'm doing an evaluation of Confbridge (migrating from Meetme). Looking at: https://wiki.asterisk.org/wiki/display/AST/ConfBridge+10 Under the heading "User Profile Configuration Options" the option announce_only_user is present. The sample config looks like this: -- ;announce_only_user=yes ;Sets if the only user announcement should be played when a channel enters a empty

I am currently having a voice quality problem with one of our Asterisk servers. We have checked the network and we have found no problems that could cause the voice to sound cracked and with small interruptions. I am looking at the timing source for Asterisk and it is currently using timerfd even though we have an E1 card installed. Is timerfd better than dahdi? Any recommendations to

Humm... Seems like my message didn't make it. Here we go again.. /Johan -------- Original Message -------- Subject: Re: [asterisk-users] ControlPlayback's options Date: Sun, 05 Jun 2011 22:19:18 +0200 From: Johan Wilfer <lists at jttech.se> To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> On 2011-06-05 19:54, virendra

Hi, I try to find some information about CAS E1 signalling and how it's handled by Asterisk. My customer wants to connect to a BT ITS Netrix by CAS E1 E&M. The system is intended to take the channels and mix them (meetme / confbridge) and send the audio back mixed to each. The layout: BT ITS Netrix: CAS E1 E&M <-> MUX - WAN - MUX <-> Digium TE220, Asterisk I've

On 11/14/17 3:38 PM, Joshua Colp wrote: > On Tue, Nov 14, 2017, at 05:23 PM, Carlos Chavez wrote: >> I am trying to get the "Mega Phone" demo working on my office PBX >> but there seems to be a problem when trying to set the default bridge to >> sfu mode. I have the following configuration in confbridge.conf in the >> default_bridge section: video_mode

I am trying to get the "Mega Phone" demo working on my office PBX but there seems to be a problem when trying to set the default bridge to sfu mode. I have the following configuration in confbridge.conf in the default_bridge section: video_mode = sfu but when I do a "confbridge show profile bridge default_bridge" I see: Video Mode: no video I can change it

On 11/14/17 3:55 PM, Joshua Colp wrote: > On Tue, Nov 14, 2017, at 05:47 PM, Carlos Chavez wrote: >> I followed the blog post and I can get video from the conference if >> I configure the bridge as follow_talker so I know everything is working >> on the pjsip side. The only problem is that video_mode = sfu is >> apparently not valid in either confbridge.conf or

On 11/14/17 4:27 PM, Joshua Colp wrote: > On Tue, Nov 14, 2017, at 06:25 PM, Carlos Chavez wrote: >> On 11/14/17 3:55 PM, Joshua Colp wrote: >> >>> On Tue, Nov 14, 2017, at 05:47 PM, Carlos Chavez wrote: >>>> I followed the blog post and I can get video from the conference if >>>> I configure the bridge as follow_talker so I know everything

All; I have a customer who is looking for a good speech to text solution, either open source or reasonably priced commercial product, I'm open to suggestions. Thanks; John V -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150826/64d6c224/attachment.html>

hello list, i use chanspy with the code below [app-chanspy] exten => _007.,1,Macro(user-callerid,) exten => _007.,n,Answer exten => _007.,n,Authenticate(1111) exten => _007.,n,ChanSpy(SIP/${EXTEN:3},dqs) exten => _007.,n,Hangup i have a question related to chanspy i have created extension from 100 to 300 and i will give the permission with group of extension i want to use

We've been having some problems with an E1 PRI line for a few days. We get the following errors: [Mar 24 10:13:39] ERROR[22009] chan_dahdi.c: PRI Span: 2 ROSE REJECT: [Mar 24 10:13:39] ERROR[22009] chan_dahdi.c: PRI Span: 2 INVOKE ID: 316 [Mar 24 10:13:39] ERROR[22009] chan_dahdi.c: PRI Span: 2 PROBLEM: Invoke: Unrecognized Operation The telephone company says that

On 4/5/16 3:17 PM, Joshua Colp wrote: > Carlos Chavez wrote: >> I am currently having a voice quality problem with one of our Asterisk >> servers. We have checked the network and we have found no problems that >> could cause the voice to sound cracked and with small interruptions. I >> am looking at the timing source for Asterisk and it is currently using >>

I am still a little confused about how to activate MWI with PJSIP on Asterisk 13.9.1. I use realtime for configuration. So far I have tried setting both the mailboxes field on ps_endpoints and the mailboxes field in ps_aors but I cannot get the indicator lamp to blink on any of my phones (Digium, Aastra and Yealink). I have tried just the number of the mailbox and also adding the context.

>Hi, I have multiple Asterisk servers in various parts of the world all connected using dedicated VPN?s. Each of these servers have iax and dahdi TRUNK configured on them. Occasionally the VPN?s fail. What I want to be able to do is on my dial plan, use IAX if the asterisk server can reach the remote server using the internet OR, use TRUNK only if it can?t use IAX. Any ideas on how this

Has anyone successfully used Mysql realtime PJSIP with Asterisk 13.11? I have tried 13.11, 13.11.1 and 13.11.2 but I always get the following error now: Sep 12 14:42:35] WARNING[24498]: res_config_mysql.c:1162 require_mysql: Realtime table general at ps_contacts: column 'qualify_timeout' cannot be type 'int(10)' (need char) [Sep 12 14:42:35] WARNING[24498]:

Hello everyone, I am trying to activate Music On Hold using DB on Asterisk 13. It works fine but in order to use new Music On hold definitions I have to reload the moh module. - The following is my configuration in extconfig.conf - I added the following line: musiconhold.conf => mysql,asterisk,bit_ast_config - The following is the table in the database: mysql> select * from

Hi, I'm facing a strange dialplan issue with a PJSIP_DIAL_CONTACTS. When I try to call an offline endpoint with PJSIP_DIAL_CONTACTS, the dial command breaks and the call control go to hangup block instead of next priority. The error in CLI says "*Dial requires an argument (technology/resource)*". This error seems legit as there are no contacts for an offline endpoint. The dialplan

Hello, I can do simple, "yum install asterisk18-*" and it installs Asterisk and Dahdi-tools/Dahdi-Linux on my OpenVZ container. Everything runs well and smooth. However, if I want to compile dahdi-linux on the same openvz then I get the error, *"You do not appear to have the source for the 2.6.32-4-pve kernel installed".* * * 1- Based on above error and Google search I have