Re: [asterisk-dev] Idle Timers and Keep-Alives

2018-01-29 Thread André Valentin
tinfo/asterisk-dev >> <http://lists.digium.com/mailman/listinfo/asterisk-dev> >> >> -- >> _____ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >>

Re: [asterisk-dev] Transcoding: Codec 2, iLBC 20, SILK, GSM-EFR, AMR(-WB)

2015-11-27 Thread André Valentin
Hello Alexander, great work!! If you ask, it would be nice if transcoding support is there for GSM-EFR and both AMR Codes. If you need testing, I can support you. Kind regards, André Am 24.11.2015 um 16:08 schrieb Alexander Traud: > Thanks to the codec/format changes which were introduced wi

[asterisk-dev] PJSIP and persistent TLS

2014-03-03 Thread André Valentin
Hi! I'm just trying to move my function ality from chan_sip to pjsip. I stumbled upon one problem. With chan_sip and a via persistant TLs connected phone everything works as expected. Calls in/out work. Even if asterisk tries to reach the phone, it reuses the existing TLS connection. If I swit