Re: [asterisk-users] ARI echo test
Thanks for answer, AGI/AMI looks still rocks, will think about using ARI just for queues and conferences. Sent from my iPhone > On 25 May 2015, at 04:55, Scott Griepentrog wrote: > > I'm pretty sure there isn't a way to do that currently. My best guess would > be that a new special type of bridge technology could be created that would > implement the per-channel echo (no audio bridged between channels in the > bridge). That would require new C code in Asterisk for the bridge, and then > the usual methods of moving channels in to bridges with ARI could be used. > >> On Sat, May 23, 2015 at 1:33 AM, Nick Awesome wrote: >> recreate Echo, if that is possible. trying to recode all dialplan to stasis >> application >> >>> On 22 May 2015, at 19:29, Scott Griepentrog wrote: >>> >>> Nick- >>> >>> Are you wanting to recreate the dialplan Echo() application in stasis? >>> >>> Why not just send the call to Echo() instead of Stasis()? >>> On Fri, May 22, 2015 at 11:25 AM, Matthew Jordan wrote: On Fri, May 22, 2015 at 4:41 AM, Nick Awesome wrote: > Can anyone tell me how can I create echo test using ARI stasis > application? > I'm not sure an 'echo' test really makes much sense with ARI, but we do have some nice documentation on getting started with ARI on the wiki. The basic tutorial example should give you an ARI event over a WebSocket connection. https://wiki.asterisk.org/wiki/display/AST/Getting+Started+with+ARI -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users >>> >>> >>> >>> -- >>> >>> Scott Griepentrog >>> Digium, Inc · Software Developer >>> 445 Jan Davis Drive NW · Huntsville, AL 35806 · US >>> direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 >>> Check us out at: http://digium.com · http://asterisk.org >>> -- >>> _ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> New to Asterisk? Join us for a live introductory webinar every Thurs: >>> http://www.asterisk.org/hello >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >>http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > > Scott Griepentrog > Digium, Inc · Software Developer > 445 Jan Davis Drive NW · Huntsville, AL 35806 · US > direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 > Check us out at: http://digium.com · http://asterisk.org > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Load Balancing with DNS SRV without DUNDI
HiI want to load balance SIP calls between two(or more) Asterisks with only DNS SRV. I used bidirectional sync Unison to synchronize configuration files and internal database file between two Asterisk boxes.The problem is when a calls come to Asterisk1 but SIPendpoint is registered on Asterisk2.How we can check a SIP endpoint is registered or not and what is Contact information in Dialplan ? Regardsbabak -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ARI echo test
I'm pretty sure there isn't a way to do that currently. My best guess would be that a new special type of bridge technology could be created that would implement the per-channel echo (no audio bridged between channels in the bridge). That would require new C code in Asterisk for the bridge, and then the usual methods of moving channels in to bridges with ARI could be used. On Sat, May 23, 2015 at 1:33 AM, Nick Awesome wrote: > recreate Echo, if that is possible. trying to recode all dialplan to > stasis application > > On 22 May 2015, at 19:29, Scott Griepentrog > wrote: > > Nick- > > Are you wanting to recreate the dialplan Echo() application in stasis? > > Why not just send the call to Echo() instead of Stasis()? > > On Fri, May 22, 2015 at 11:25 AM, Matthew Jordan > wrote: > >> On Fri, May 22, 2015 at 4:41 AM, Nick Awesome wrote: >> > Can anyone tell me how can I create echo test using ARI stasis >> application? >> > >> >> I'm not sure an 'echo' test really makes much sense with ARI, but we >> do have some nice documentation on getting started with ARI on the >> wiki. The basic tutorial example should give you an ARI event over a >> WebSocket connection. >> >> https://wiki.asterisk.org/wiki/display/AST/Getting+Started+with+ARI >> >> -- >> Matthew Jordan >> Digium, Inc. | Director of Technology >> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA >> Check us out at: http://digium.com & http://asterisk.org >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >>http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > > -- > [image: Digium logo] > Scott Griepentrog > Digium, Inc · Software Developer > 445 Jan Davis Drive NW · Huntsville, AL 35806 · US > direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 > Check us out at: http://digium.com · http://asterisk.org > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [SOLVED] Re: asterisk 13 webrtc
dtlsenable=yes was missing thank you joshua Dne 21.5.2015 v 22:53 Marek Cervenka napsal(a): hi, is there someone with working asterisk13+chan_sip+SIP.js/sipml5 ? or is chan_pjsip better supported? or the recommended way for asterisk is use respoke.io? my problem with asterisk13+chan_sip+sipml5(the same problem is with SIP.js) chan_sip.c:10496 process_sdp: Can't provide secure audio requested in SDP offer " sip.conf (important parts) [vr1a882] ... nat=force_rport,comedia canreinvite=no encryption=yes avpf=yes force_avp=yes icesupport=yes directmedia=yes transport=wss,ws dtlsrekey=60 dtlsverify=no dtlscertfile=/etc/pki/tls/certs/rapidssl.crt dtlsprivatekey=/etc/pki/tls/private/rapidssl.key dtlssetup=actpass -- --- Marek Cervenka === -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
