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check this line in sip.conf : canreinvite = no ; allow RTP voice traffic to bypass Asterisk if it set to yes RTP traffic will _try_ to go directly between SIP endpoints. If it set to no - all traffic will be sent over PBX.


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It depends. If you do any transcoding (in other words the two endpoints use different codecs) then Asterisk will definately be in the RTP stream. If your two endpoints are on opposite sides of NAT, then Asterisk will likely be in the RTP stream (also depending on reinvite settings). If both endpoints are on the same side of NAT then likely no.


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Instead of using CDR, I log CEL to a table. Then I can see the call which are started (eventtype 1, CHANNEL_START), and see if the linkedid has corresponding events with eventtype 4 ANSWER and events with eventtype 2 CHANNEL_END I then find the lastest customer-id with that phone number (a customer might register multiple time with the same number), and ...


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Either by dtmf-ing *1 (Usually *2 to make the transfer, check in features.conf) or bridge the original channels manually. The "Atxfer" action actually sends dtmf internally. You might want to take a peek at the "t" and "T" flags of the Dial application as they decide who can dtmf to a specific channel. ...


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There are two ways. Either 1-using 183 Early media. The other way is using Answer() application. In both cases, the media will come from the Asterisk side. ; 183 - session progress method. exten=> _X.,1,Progress() exten=> _X.n,playback(tt-monkeys) ; 200 - Answer exten=> _X.,1,Answer() exten=> _X.,n,playback(tt-monkeys)



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