Archives : January-2016
can you share your best practices for ARI reconnect when asterisk is restarted or when ari app is started before asterisk is fullybooted?we are using node.js + ari-client so we are thinking about these options:1) wait for AMI event FullyBooted2) w..
How should I configure Asterisk (13.7.0) to get persistently PJSIP SIPmessages in a log file and not in console ?I would expect adding debug=yes in pjsip.conf to produce the same output as pjsip set logger on. Am I understanding correctly ?Be..
Would greatly appreciate any input into this currently-unanswered question on the forum:http://forums.asterisk.org/viewtopic..
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Hihow to flush user input before READ()?I wrote a small script to ask for user password before granting access to outside, but some telefones memorize the full user input, including #.So, when the user press redial, for instance 5556789#123, aster..
Hi.I cant find X-RTP-Stat SIP header in my packets. Im using Asterisk 13.6and PJSIP.Is there something that I should configure to Asterisk add this header?Thanks.Marcelo Hartmann Terres Fones: +55 51 3024-3568 | +55 11 4063-8864 | +55 92 3090-0115Pro..
I am running 11.20.0 (64 bit) as a user other than root and using the Console/dsp port (soundcard) output and HDMI.I am getting a warble or clicking noise on the audio. Im connected into the pulseaudio for the logged in user. Pulseaudio works fine ..
I am turning up a PJSIP Endpoint and am having problems when they send an INVITE to my server. Asterisk is returning a 421 Extenstion Required. Sinceextension means different things in the SIP stack versus Asterisk, Idont know what it is complain..
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