Author : Michael Maier

PJSIP: Asterisk Can’t Decide Which Codec To Use

Posted May 12, 2017 by Michael Maier & filed under Asterisk Users Comments: 5.

Tags: aar, C language, embedded systems, format changed from none to alaw, incoming call, ooh

Im facing completely choppy sound. The wireshark trace shows, that there are a lot of codec changes without any trigger (means no options or reinvite or any other package).Please note: PJSIP is a free and open source multimedia communication libr..

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Some Questions Regarding Jitterbuffer In Asterisk / Pjsip

Posted May 8, 2017 by Michael Maier & filed under Asterisk Users Comments: 0.

Tags: Jitterbuffer, SBC

Hello!I just implemented a jitterbuffer for pjsip in the dialplan in a SBC:[fromtrunk]exten => _[+0-9]!,1,Set(JITTERBUF..

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Outbound T.38 Via RTP With Pjsip Does Not Work As Expected

Posted April 6, 2017 by Michael Maier & filed under Asterisk Users Comments: 3.

Hello!Im trying to send a fax via T.38 to a destination, which should be T.38capable. My provider supports T.38, too. Unfortunately, it doesnt work. This means:Call is started and SDP is negotiated w/ alaw. Callee sends reinvite -for alaw again (..

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