Re: [asterisk-users] ARI echo test

2015-05-24 Thread Ilya Awesome
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 sgriepent...@digium.com 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 jl...@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 sgriepent...@digium.com 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 mjor...@digium.com 
 wrote:
 On Fri, May 22, 2015 at 4:41 AM, Nick Awesome jl...@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 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
 
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 Digium, Inc · Software Developer
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 direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090
 Check us out at: http://digium.com · http://asterisk.org
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Re: [asterisk-users] Asterisk switching bridge to native_rtp even with direct_media=no

2015-03-22 Thread Ilya Awesome
Ok, if this is normal why I have oneway audio when nat endpoint calling to 
local.
if mixmonitor or srtp is enabled audio is ok. 
Issues with native_rtp for sure

Sent from my iPhone

 On 19 Mar 2015, at 23:08, Matthew Jordan mjor...@digium.com wrote:
 
 On Thu, Mar 19, 2015 at 1:47 AM, Nick Awesome jl...@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@dialmap:1] AGI(PJSIP/304-0022, /pbx/agi.php) in
 new stack
-- Launched AGI Script /pbx/agi.php
-- AGI Script Executing Application: (Dial) Options:
 (PJSIP/99/sip:99@192.168.1.73:5060,20)
-- Called PJSIP/99/sip:99@192.168.1.73:5060
-- PJSIP/99-0023 is ringing
-- PJSIP/99-0023 answered PJSIP/304-0022
-- Channel PJSIP/304-0022 joined 'simple_bridge' basic-bridge
 da8840bc-9b71-4ca6-b1d8-9565bf8e5e28
-- Channel PJSIP/99-0023 joined 'simple_bridge' basic-bridge
 da8840bc-9b71-4ca6-b1d8-9565bf8e5e28
 Bridge da8840bc-9b71-4ca6-b1d8-9565bf8e5e28: switching from
 simple_bridge technology to native_rtp
 Locally RTP bridged 'PJSIP/99-0023' and 'PJSIP/304-0022' in
 stack
 Locally RTP bridged 'PJSIP/99-0023' and 'PJSIP/304-0022' in
 stack
 0x7f4b50145420 -- Probation passed - setting RTP source address to
 194.204.157.200:8972
 0x7f4b5014f140 -- Probation passed - setting RTP source address to
 192.168.1.73:5004
-- Channel PJSIP/304-0022 left 'native_rtp' basic-bridge
 da8840bc-9b71-4ca6-b1d8-9565bf8e5e28
-- Channel PJSIP/99-0023 left 'native_rtp' basic-bridge
 da8840bc-9b71-4ca6-b1d8-9565bf8e5e28
-- PJSIP/304-0022AGI Script /pbx/agi.php completed, returning 4
 
 Correct - and per the log, they shouldn't be in a direct media bridge:
 
 Locally RTP bridged 'PJSIP/99-0023' and
 'PJSIP/304-0022' in stack
 Locally RTP bridged 'PJSIP/99-0023' and
 'PJSIP/304-0022' in stack
 
 Locally RTP bridged means media is still flowing through Asterisk, it
 just isn't being decoded and passed through the core.
 
 
 -- 
 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
 
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[asterisk-users] Sent ami event from AGI?

2014-10-02 Thread Ilya Awesome
hello, is there way to send event to all ami clients from AGI script?

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