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I am developing apps using ARI which need suppression of DTMF tones in the audio, and I have been told (back in December) that asterisk depends on SIP providers to suppress DTMF tones in the audio stream. Having sorted out my ARI code to suppress D..
–000000000000cbeb58059d4c637c Content-Type: text/plain; charset=UTF-8For those of you who actually process SIP MESSAGE requests…Do you use any of the AMI events generated by the Message/ast_msg_queue channel? We want to change that channel to an inter..
Gang I have stumbled of this problem. I need the P-Asserted-Identity header in an AGI scrip. In the Dial-Plan I do: same => n,Set(PAI=${PJSIP_HEADER(read,P-Asserted-Identity)}) In the AGI I do: my $pai = $AGI->get_variable(PAI); This works fine, unl..
Are you using chan_sip or chan_pjsip? If chan_pjsip then you need to use the PJSIP_HEADER function [1] instead. For instance the following returned the call-id for me:PJSIP_HEADER(read,Call-ID)How are you attempting to use the channel function [2]? ..
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When we created the External Media addition to ARI we created an ExternalMedia object to be returned from the channels/externalMedia RESTendpoint.This object contained the channel object that was created plus local_address and local_port attributes (wh..
I dont know if its ever really been tried, recording/playback of video is not something anyone has put a focus n. Only really h264 and h263 I believe. Youd need to negotiate h264 on the channel. As it is youre using VP9, and Asterisk does not do vi..
The aco_option_register function is used by modules to register some configuration handling logic. The bridge feature API is defined in bridge_features.h[1] but Im not sure functionality you require (arbitrary adding of features) was ever added to ..
Have you looked at the actual ARI events that are occurring and the (presumably SIP) signaling for the call when it occurs? — Joshua C. Colp Digium – A Sangoma Company | Senior Software Developer 445 Jan Davis Drive NW – Huntsville, AL 35806 – US Ch..
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