similar to: Best timing source?

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 >>

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 >>

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 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

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]:

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.

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

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 Mon, Sep 12, 2016 at 2:31 PM, George Joseph <gjoseph at digium.com> wrote: > > > On Mon, Sep 12, 2016 at 2:14 PM, Carlos Chavez <cursor at telecomabmex.com> > wrote: > >> 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: >>

I use realtime on my asterisk installation. I have always used mysql for my realtime connection but as mysql seems to be on the "soon to be deprecated" list of asterisk features I am trying to move to ODBC (still using MariaDB/Mysql on backend). I find ODBC support in Asterisk very unstable. Just today my asterisk server (a test server and a production server) has crashed often

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>

Is it possible to use serveral protocols for a single transport section in pjsip.con? In sip.conf you could use transport=udp,ws,wss so you cound use webrtc along with your phones but if I try: [transport-udp] type=transport protocol=udp,ws,wss bind=0.0.0.0 I get an error that transport-udp is not found. Do I need a dedicated interface for WebRTC? [Feb 15 12:42:10] ERROR[3308]:

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

I am unable to force a hangup on a channel that has been stuck for over two days: IAX2/from-CD-11006 oficina 2770 1 Up Dial IAX2/to-CD/2883 3467130007 46:24:59 Sotelo Sotelo IAX2/to-CD-20713 I have tried "hangup request IAX2/from-CD-11006" several times but no joy. I also see the following in the CLI: [Nov 3

I am having a small problem that is driving me nuts. I can make calls over my Webrtc client without any problems and audio sounds fine. The only problem I have is that when I call an internal SIP extension on my PBX I do not hear the ring while I wait for the call to be answered. My dial command does include the rR options. If I make an external call to a land line or a mobile phone I do

On 2/15/16 12:50 PM, Joshua Colp wrote: > Carlos Chavez wrote: >> Is it possible to use serveral protocols for a single transport section >> in pjsip.con? In sip.conf you could use transport=udp,ws,wss so you >> cound use webrtc along with your phones but if I try: >> >> [transport-udp] >> type=transport >> protocol=udp,ws,wss >> bind=0.0.0.0 >

>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

Many Thanks Carlos, I was hoping to check whether the remote server is available before I issue the dial in my dial plan. Is there a better way to do it in asterisk without using unix commands? Many Thanks, Ashwin On 5/30/15, 2:06 AM, "Carlos Chavez" <cursor at telecomabmex.com> wrote: >On 5/29/15 1:16 PM, Ashwin Surendran wrote: >>> Hi, >> I have multiple

Can anyone recommend a good WebRTC phone to use with Asterisk? I do not mind if it is commercial or open source. Customers are starting to ask for web solutions and we need to start testing. -- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez +52 (55)9116-91161