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Well, it breaks audio for all NAT endpoints, how can I fix this? > On 18 Mar 2015, at 15:48, Matthew Jordan <mjordan at digium.com> wrote: > > On Wed, Mar 18, 2015 at 7:41 AM, Nick Awesome <jleed at me.com <mailto:jleed at me.com>> wrote: >> Hey guys, >> >> have issues with reinvite, no matter what endpoint is calling asterisk >> always tries switch simple_bridge to native_rtp >> >> Bridge 0422bfa0-9d22-4bba-9108-a3f14d7d1cab: switching from si...

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 <mjordan at digium.com> wrote: > On Fri, May 22, 2015 at 4:41 AM, Nick Awesome <jleed at me.com> 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...

...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 <jleed at me.com> wrote: > recreate Echo, if that is possible. trying to recode all dialplan to > stasis application > > On 22 May 2015, at 19:29, Scott Griepentrog <sgriepentrog at digium.com> > wrote: > > Nick- > > Are you wanting to recreate the dialplan Echo() appl...

Hey guys, have issues with reinvite, no matter what endpoint is calling asterisk always tries switch simple_bridge to native_rtp Bridge 0422bfa0-9d22-4bba-9108-a3f14d7d1cab: switching from simple_bridge technology to native_rtp in endpoints table ?direct_media? sets to ?no? on all endpoints but it doesn?t help. if native_rtp not work for some reason I have oneway audio. how can I fix this?

...native_rtp' basic-bridge <da8840bc-9b71-4ca6-b1d8-9565bf8e5e28> -- <PJSIP/304-00000022>AGI Script /pbx/agi.php completed, returning 4 > On 18 Mar 2015, at 18:26, Matthew Jordan <mjordan at digium.com> wrote: > > On Wed, Mar 18, 2015 at 9:53 AM, Nick Awesome <jleed at me.com> wrote: >> Well, it breaks audio for all NAT endpoints, how can I fix this? >> > > Local (packet to packet) bridging should not do that. Remote (direct > media) can do that. > > Can you confirm - by looking at a verbose level 4 log - how Asterisk > is b...

Can anyone tell me how can I create echo test using ARI stasis application?

...he 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 <mjordan at digium.com <mailto:mjordan at digium.com>> wrote: > On Fri, May 22, 2015 at 4:41 AM, Nick Awesome <jleed at me.com <mailto:jleed at me.com>> 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 wit...

Hey guys,tried to make tls work with pjsip on asterisk 13.2.0 have compiled pjsip with ssl, added transport [tls] type=transport cert_file=/pbx/keys/server.crt ca_list_file=/pbx/keys/ca.key priv_key_file=/pbx/keys/server.key protocol=tls bind=192.168.1.4:5061 local_net=192.168.1.0/24 external_media_address=77.77.77.77 external_signaling_address=77.77.77.77 have configured Grandstream GXP1400

On Wed, Mar 18, 2015 at 7:41 AM, Nick Awesome <jleed at me.com> wrote: > Hey guys, > > have issues with reinvite, no matter what endpoint is calling asterisk > always tries switch simple_bridge to native_rtp > > Bridge 0422bfa0-9d22-4bba-9108-a3f14d7d1cab: switching from simple_bridge > technology to native_rtp > > in e...

On Wed, Mar 18, 2015 at 9:53 AM, Nick Awesome <jleed at me.com> wrote: > Well, it breaks audio for all NAT endpoints, how can I fix this? > Local (packet to packet) bridging should not do that. Remote (direct media) can do that. Can you confirm - by looking at a verbose level 4 log - how Asterisk is bridging the two channels? -- Matthew...

On Thu, Mar 19, 2015 at 1:47 AM, Nick Awesome <jleed at me.com> wrote: > NAT endpoint calling local endpount - switching to native_rtp then no audio, > both of them have direct_media=no, Verbose log: > > -- Executing [99 at dialmap:1] AGI("PJSIP/304-00000022", "/pbx/agi.php") in > new stack > -- Launch...

On Fri, May 22, 2015 at 4:41 AM, Nick Awesome <jleed at me.com> 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 giv...

...request in PJSIP ASTERISK-24986 issues opened already more the 2 month and calls from customers still drops. very annoying :( maybe some one could help me figure out where Received INFO request dies in source so I could patch it to response 200 OK ? > On 20 Apr 2015, at 15:08, Nick Awesome <jleed at me.com> wrote: > > Hi guys, have really annoying problem with trunks when I calling over voip provider.. > > > after awhile provider sends INFO packages but for some reason Asterisk doesn?t answer on it. > after 8 packagers provider just drops the call, here is the packag...

Hi, after update on 12.4.0 asterisk crashes on MeetMe ending on 12.3.2 it worked well. Is some one else have this issues? should someone open a ticket?

hello, is there way to send event to all ami clients from AGI script? Sent from my iPhone

Hi guys, have really annoying problem with trunks when I calling over voip provider.. after awhile provider sends INFO packages but for some reason Asterisk doesn?t answer on it. after 8 packagers provider just drops the call, here is the package <--- Received SIP request (555 bytes) from UDP:192.168.53.1:5060 ---> INFO sip:c2e74184-4d23-43a3-8fd9-26ba2c5ef6c9 at 192.168.53.9:5060 SIP/2.0

Oh god it works ! to switch cisco to upd I used config: <transportLayerProtocol>2</transportLayerProtocol> with udp it works well, thanks for your help :) > On 24 Feb 2015, at 17:02, Joshua Colp <jcolp at digium.com> wrote: > > If you use UDP with force_rport=no it'll work. > If you use TCP then set rewrite_contact=yes so it'll reuse the established TCP

Hay guys, have question. When I do regular dial I use $this->AGI->get_fullvariable('${PJSIP_DIAL_CONTACTS('.$callObj.')}',false,true); to get all contacts of current endpoint and so I dial to all phones at once, but if I exec QUEUE, I have just one phone rings, seems like it take first one as Dial app by default, is there way to fix this?

Stuck with TLS transport, I have 2 phones (both in local network for tests) one connected with up second with tls when I calling TLS to UPD everything is fine, but when UDP calls TLS I getting an error ERROR[44230]: pjsip:0 <?>: tlsc0x7f143012 TLS connect() error: Connection refused [code=120111] pjsip log: -- Called PJSIP/601/sip:601 at 192.168.1.55:5075;transport=tls <---

what may cause this, and how can I fix it ? WARNING[23010]: pjsip:0 <?>: tsx0x7f24f41b2 ..Failed to send Request msg NOTIFY/cseq=15293 (tdta0x7f2480001a70)! err=171064 (Unsuitable transport selected (PJSIP_ETPNOTSUITABLE)) -------------- next part -------------- An HTML attachment was scrubbed... URL: