similar to: Video Call

Your characterization may be true but Skype works much better than SIP when it comes to sound quality. I have SIP softphone with Asterisk server and Skype on the same workstation. Skype just works better over the same network. Ron On 12/03/2015 9:26 AM, A J Stiles wrote: > On Thursday 12 Mar 2015, Thufir wrote: >> I'm testing Asterisk at home, crummy connection. Skype works fine

Hi, I'm using gotoiftime on asterisk, but it seems  there is a difference between the asterisk time and the system time. could it be because i adjusted the system timezone on my linux? do asterisk not detect the change of timezone on the system? How can I fix this prob? Regards, nhadie -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi List, Im planning on setting up asterisk for a large scale enviorment, with multiple sites. We will be doing quite a bit of inner office calling at each site, and want to place a smaller scale * box at each site with no PRI's, and have that connect to our main * servers at our data center that will have the PRI connections. Can this be done? I havent seen to much of this on the mailing

Good Day list, I have a friend who is interested in implementing an asterisk implementation at his offices. The configuration would consist of the following Site A ---- Asterisk Box With 12 incoming lines and 15 phones Extensions 101-115 Site B ---- Asterisk Box With 4 incoming lines and 7 phones Extensions 201-207 Site C ---- Asterisk Box With 4 incoming lines and 6 phones

I know this has come up before but... Has anyone done anything to get an IAX client built on Windows? I thought someone had started one, but I haven't heard anything about it since - and that was months ago? Anyone have any idea what the status is? -- Ron Gage - Saginaw, Michigan I am looking for work - resume at http://www.rongage.org/resume.doc Electrical Engineering, Linux Programming,

Hello everyone! I''m simply trying to put a PRIO inside an HTB (used to throttle). I''ve got interactive traffic on the network that I want to give priority (VoIP + Citrix + Video). I''ve used the filters in a CBQ script fine, but am having trouble adjusting them to this setup such that they properly assign the traffic. tc qdisc del root dev $e tc qdisc add dev $e

Hi My head hurts... Can anyone help out here, my remote IAX can see my local IAX and visa versa, conversation starts, I can dial my remote (POTS) landline number, remote end answers, trys to route to local iax2, I see it start the conversation here, the extension (SIP) rings once and then it dies... Both ends are defined with accept IPADDRESS to keep it in the family and simple.. Debug info

Using DIAX softphone which seems to be working OK can get to VM/echotest etc in the demo context Am trying to setup FWD but get the following problems Can hear it ringing when dialing FWD no 612 for time. Connects but no sound from remote end. Does anyone have any suggestions. Softphone on 192.168.0.2 asterisk on 192.168.0.3 Netgear RP114 doing NAT to the internet port 5060 being forwarded to

I have a question for using gastman. I have set up extensions for my IAX users as IAX2/username, and I keep getting the following Dunno how to tell if IAX2/username/6 is IAX2/username I was wondering if there is some sort of wildcard character that can be used here? The number changes every time, so I do not think that I can put in seperate extensions. Thank You, Ron Frederick --------------

I'm installing an office in a couple of weeks that will have some nice Polycom FX video units in the conference rooms. I'm thinking that with asterisk-oh323 http://www.inaccessnetworks.com/ian/projects/asterisk-oh323/#section2 I should hopefully get the ability for phone users to dial an extension and participate in video conferences, or just simply phone conference with users in the

Good Day list I am looking to buy a few Wooksung Video phones to try with my asterisk box.... http://www.wooksung.com/eng/html/pro/pro_001.html has anyone had any experience using these with asterisk? Thanks Ron

how to make g.729 preferred, but failover to gsm I've purchased a few g.729 licences, and would like to set up iax.conf such that g.729 is used if they are available, but then it fails over to gsm. I'm not sure how to specify such a preference. I'll let the server transcode from ulaw (from the sip phones) to g.729. Got plenty of CPU for the number of phones we run off that

Sorry for the empty message. Pressed the wrong button. I have been wrestling with a pretty generic Asterisk configuration (version 11.11.0 ) set up with FreePBX. The trunk SIP is setup to allow ulaw,alaw,gsm, Video is disabled. I was using Eyebeam and am now trying Jitsi. Jitsi has a number of codecs enabled - opus, SILK, G722, speex,PCMU, PCMA, iLBC, GSM, G723 and telephone-event The

Hi All, if i do this setup: |---[ext 100] |--[router/nat gw]--| | |---[ext 101] | [asterisk]--[internet]---| | | |---[ext 200] |--[router/nat gw]--|

Hi All, I'm trying to test video calls on asterisk, my issue is i have two asterisk servers linked via an IAX peer and users register on the asterisk via SIP. if both video phones are registered on the same asterisk server, video call works, but call is via SIP. but,if one video phone 1 is registered on asterisk 1 and video phone 2 is registered on asterisk 2, video only works one way,

Hi, all I have couple of SIP phones and they are in [from-sip] context. I also have an IAX2 phone. I have put this one in [iax-user] context. I want to make calls between SIP and IAX2 phones. If I put them all in same context all is fine, however when they are in different contexts they will not call each other and I will get message (in * CLI) that particular extension does not exist in a

Hi I'm browsing around the internet looking for signs that the IAX client library and app_meetme support video. I stumbled across this post by SteveK on the 27th of Feb 2006. "My company is looking to hire a full-time developer, who will be working about 25-50% of the time on iaxclient; in particular to finally integrate, build, polish and enhance video in iaxclient, add video

Hello Ron, Which kernel do you run Asterisk/Freepbx with ? Cheers 2017-12-14 16:57 GMT+01:00 Ron Wheeler <rwheeler at artifact-software.com>: > CentOS 7 works well with Asterisk. > Install latest CentOS7 with updates install asterisk > > I am running FreePBX on CentOS 7. > > Ron > > On 14/12/2017 10:38 AM, Olivier wrote: > > Hello, > > I'm used to

hi, hs anyone able to make video to work on asterisk? i tried following this: http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+xten+eyeBeam i can see that eyebeam is trying to broadcast a video but the other eyebeam is not receiving it. i tested the same setup but this time using ser with rtpproxy and eyebeam video works fine. any ideas? where do you think should i start

I've created a pretty complete HOWTO on creating a Linux Bridge (using Fedora) to shape LAN <--> WAN traffic. It includes installation instructions, a script to configure the bridge (which you install as a service), and 2 scripts to configure the network interfaces using traffic control. http://www.burnpc.com/website.nsf/all/3a64a6369757819686256f960068ad75!OpenDocument If anyone