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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 bhati wrote: > Hi John Wilfer, > > Thanks for replay. Now all things is working on asterisk 1.6.2.18 > version. But When...

...find this option. Any clues? Also - setting quiet=yes still plays join/leave sound. My current work-around is: sound_join=silence/1 sound_leave=silence/1 But this seems a bit ineffective... In Meetme the quiet-flag also disabled join/leave sounds. Is this by design or an oversight? -- Johan Wilfer

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 +/- >&...

...s How do you think about adding webrtc to a existing Asterisk/Kamailio environment? Do you use kamailio (websockets) as a front, a dedicated webrtc asterisk or something like webrtc2sip? How do you use / plan to implement webrtc in your environment? Any feedback is welcome. Thanks! -- Johan Wilfer

Hi all, Sorry for cross posting. I attached my chan_ooh323 patches (asterisk-addons-1.4.5) when codec negotiation patch changes applied to asterisk-1.4.17. Please let me know whether my patches are correct or not. thanks in advance, Ganbold -------------- next part -------------- An HTML attachment was scrubbed... URL:

...do I handle answer / hangup with CAS? Will DAHDI keep this channels up, or should I query the state of the channels (how?) and bring them up myself if they are down (Dial?) 3. Other suggestions? CAS is unknown territory for me so I appreciate all the pointers you have. Thank you! -- Johan Wilfer

...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! 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

...pbx (bypassing kamailio) Asterisk correctly uses the externip as the rtp-ip in the SDP. I know this is an old and unsupported version of Asterisk, but any input on the topic is welcome. If this is supported in later versions we can maybe work around until we migrate later. Thanks! -- Johan Wilfer

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

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) Is it good to go, or not ? I really hope someone who have experience with it willing to share with me... Thanks in advance... Best Regards, Ikka - Jakarta,

Hi, I have some servers that are dedicated to do meetme conferencing. From some previous test i concluded that I need to use dahdi_dummy as it is more accurate. If I did use the core timers in dahdi (not loading dahdi_dummy) I got bad quality in the conferences and dahdi_test showed 99.6% as worst. I thought maybe the issue as bad hardware for the timing or something else. But today I

...Solaris containers or BSD jails. Docker is mostly using the same Kernel api:s that OpenVZ uses, but OpenVZ also has some cusom stuff. If you need Dahdi you will need to give the VE's access to these devices, there are articles out there that explain how this is done. Good luck! -- Johan Wilfer

...x, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mitul at enterux.in DID: +91-22-71967196 Cell: +91-9820332422 On Tue, Apr 7, 2015 at 9:57 PM, Jeff LaCoursiere <jeff at jeff.net> wrote: > 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...

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

Greetings! I have an Asterisk 1.4 box and due to hardware limitations I cannot upgrade atm. So, as long as I understood from different posts, SIP-TLS is not available for 1.4 Then I set up VPN and route all inter-Asterisk traffic into VPN. But for some reason, with all the RTP inside the VPN I start getting packet losses up to 30%. Maybe CPU is too weak, that is yet to be discovered. What

...uipment 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 Wilfer

Hello, Many times, I need to test in a lab Asterisk servers before sending them to customer locations. I'm currently having trouble to test SIP trunks without touching SIP configuration. So, how should I change my testing lab so that I could now test SIP trunks without modifying Asterisk server under test ? A typical set up is: Asterisk server1 under test <---SIP----> Router

Hello All; I'm attempting to build the dahdi on an OpenVZ instance: Linux serverx 2.6.18-274.7.1.el5.028stab095.1 #1 SMP Mon Oct 24 20:49:24 MSD 2011 x86_64 x86_64 x86_64 GNU/Linux Now, the kernel says that I have the proper one installed, as you can see from above. However, when I run the make all, this is what I see: You do not appear to have the sources for the