similar to: CAS E1 signalling

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 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

Hi! Anybody familiar with streaming rtp from asterisk. Preferably with the xorcom asterisk patch which streams rtp from asterisk to oreka audio server. Any ideas will do just fine though! Regards / Marcus

Hi List, Asterisk 's *ControlPlayback* will used for play any recorded file as an audio player. Is it possible that we can use it for multiple forward and rewind ? ex:- original: ControlPlayback(filename,skipms,ff,rew,stop,pause) expected ControlPlayback(filename,skip1,skip2,skip3,forward1,rewind1,forward2,rewind2,forward3,rewind3,stop,pause) : ----- Thanks and regards Virendra Bhati

Hello, I'm trying to cut off the "+" sign if part of a telephone number, but not succeeding : exten => test,n,Set(cid=+99999600) exten => test,n,Set(regx="([0-9])") exten => test,n,Set(cid2=$["${cid}" : ${regx}]) exten => test,n,NoOp(cid2=${cid2}) cid2 is empty afterwards... What I want is to make sure there are only numbers and no other

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

I am running Asterisk 1.16.2.13, dahdi 2.4.0 and libpri 1.4.11.4 on an HP ML110 G6 using Ubuntu Linux 10.04 LTS. I have two Digium TE121 single T1 port cards and a Digium AEX800 8-port FXS card. All PCI Express cards. Co-workers are hearing hissing sounds on some calls, and I am getting IRQ errors when running "dahdi show status". I see that sharing IRQs for Digium cards isn't

Hi, Anyone who has experience with ISDN PRI voice gateways? My customer wants to connect some E1 equipment to a gateway that converts from ISDN PRI E1 to SIP/RTP. The data will be transmitted over a WAN, and into an Asterisk-1.8 server. It's one E1 on each site at 8 sites, and they are asking for our recommendations on gateways. Your advice on this topic is very appreciated! -- Johan

Greetings- I'm about to dive into the process of virtualizing some of my Asterisk (primarily 1.4.x) infrastructure. In the past, when looking at virt solutions, the primary issue preventing me from moving was the lack of proper timing. We do not need it for MeetMe but rather for IAX2 trunking. I'd like to use either OpenVZ or KVM, but each seem to have independent "issues" that

Hi, I write some php code in AMI to working with asterisk command. I don't know exactly what is the different between AMI and AGI and witch one is better for my planning. Im planning to call party users that their number is is my panel on web. We have some operator and they can call party users via client softphone by clicking on their number, so they have to limited to call just listed

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

On 04/07/2015 10:48 AM, Johan Wilfer wrote: > Den 2015-04-07 15:41, Ikka Tirtawidjaja skrev: >> Dear all, >> >> Is anyone has experience making Asterisk server with virtual server >> OPEN-VZ (in proxmox 3.4 box) ? >> >> My boss want to build a production server with it, and it will have +/- >> 300 sip user (concurrent call maybe < 150 call) >>

Dear SAMBA gurus, I browsed the list archives but found nothing relevant about this big (IMHO) problem. Only just another problem like mine, unfortunately unanswerd :(. I have here samba-2.0.3 on a Linux box, libc5, manually compiled without problems. All is well EXCEPT WinPopUp messages, I can send them OK but I cannot receive. As receiver I set LinPopUp but I don't think that is ever

Hi, is there another IAX-CLient for Linux beside Gnophone? The problem with Gnophone is, when there is an incoming call. There is no pop-up-window or any sound that caughts my attention. So calls easily remain unanswerd. Or can I configure Gnophone in the desired way? Thx, Holger

Dear SAMBA gurus, I browsed the list archives but found nothing relevant about this big (IMHO) problem. Only just another problem like mine, unfortunately unanswerd :(. I have here samba-2.0.3 on a Linux box, libc5, manually compiled without problems. BTW kernel is 2.2.7ac1. All is well EXCEPT WinPopUp messages, I can send them OK but I cannot receive. As receiver

Hi! Anyone that have tried using Asterisk 11 with SIP + Confbridge as a VMware virtual machine? Any issues to be aware of? Of course the hardware node needs to to be powerful enough - but say you have just one virtual machine on the node - will the performance be drastically less than running asterisk on the metal? Or can I expect roughly the same performance? Thanks! -- Johan Wilfer

Hi, We have some Asterisk servers that we are moving behind a NAT to preserve public addresses and make room for growth. This is Asterisk 1.4 NAT works very good with the externip/localnet-setting when we are connected directly to our teleco. But when I try to use NAT and put them behind our Kamailio something interesting happens: The media-address in the SDP is the internal ip and not the

According to CarrierAccess, the Adit 600 uses CAS for voice signalling. What is this? This should not be a problem for Asterisk? Does the Asterisk server need to reencode CAS into aLaw when going to Euro ISDN? BR Daniel Nystr?m

Hi folks, I had a survey online but there i couldnt find a clean sample of CAS signalling on E1 interfaces. I defined a span with CAS framing and HDB3 line coding but dont know which signalling to use for channels. I'd use 3 bit CAS signalling and 20 incoming channels and 10 outgoing ones. Anyone could help me define the signalling for these channels. PS. Im using Sangoma cards. Any