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...productive btw we are using tools like sipcapture.org,voipmonitor.org, callstats.io, elasticsearch+filebeat, ... but without informations whats happening inside asterisk is harder to solve problems Dne 12/12/2019 v 16:00 Joshua C. Colp napsal(a): > On Thu, Dec 12, 2019 at 10:57 AM marek <cervajs64 at gmail.com > <mailto:cervajs64 at gmail.com>> wrote: > > thank you very much. this is exactly whats needed for debug > > example output for your info > > [Dec 12 15:39:19] DEBUG[2182][C-00000000]: pjproject: <?>:         > icess0x7f5d44081e8...

Hello, any plans for astricon videos? Thanks Marek

...e router ip - customers router ip - customers private ip ) to stun request pjsproject debug config pjproject.conf [startup] log_level=4 type=startup btw some examples will be very helpfull Marek Dne 12/12/2019 v 14:05 Joshua C. Colp napsal(a): > On Thu, Dec 12, 2019 at 8:57 AM marek <cervajs64 at gmail.com > <mailto:cervajs64 at gmail.com>> wrote: > > Asterisk is on public IP (as described in the first email) > > i have 10 years experience in voip, 4 years webrtc in production. > i know about ICE/STUN/DTLS-SRTP. yes, not every detail but the >...

...TP. this is not only about ICE. its about RTP engine too which is Asterisk specific and Asterisk DEBUG is not helping ... going back to read res_rtp_asterisk.c & decrypting pcaps with wireshark Dne 12/12/2019 v 13:02 Joshua C. Colp napsal(a): > On Thu, Dec 12, 2019 at 7:57 AM marek <cervajs64 at gmail.com > <mailto:cervajs64 at gmail.com>> wrote: > > with wireshark i need decrypt traffic every call which is time > consuming. get debug from pjnat through asterisk is not possible > because of technical reasons or nobody did it? > > > in my...

...from    1.1.1.1:15026 (type 08, seq 021201, ts 000800, len 000160) looking for the part where RTP engine switch from *10.2.152.36 to **2.2.2.2* it looks like**its somewhere in the learning phase* * Dne 12/12/2019 v 11:51 Joshua C. Colp napsal(a): > On Thu, Dec 12, 2019 at 6:39 AM marek <cervajs64 at gmail.com > <mailto:cervajs64 at gmail.com>> wrote: > > hi, > > i have following topology > > PSTN - Asterisk ---- internet -----  router - jssip client (wss) > > Asterisk 13.29.1 on public IP, chan_pjsip for wss, chan_sip/udp > for SIP...

hi, i have following topology PSTN - Asterisk ---- internet -----  router - jssip client (wss) Asterisk 13.29.1 on public IP, chan_pjsip for wss, chan_sip/udp for SIP connection to PSTN router - public IP/private IP (NAT) jssip client - private IP - sip over websocket to Asterisk PJSIP ~30% of calls has problem with no audio. reason is that Asterisk is sending RTP to private IP of jssip

On Tue, Jan 21, 2020 at 9:51 AM marek <cervajs64 at gmail.com> wrote: > Hello, > > any plans for astricon videos? > If you are referring to AstriCon 2019 there was no filming of sessions. I did record my own, though, about PJSIP[1]. [1] https://www.youtube.com/watch?v=wCOa04g1c7w -- Joshua C. Colp Asterisk Technical Lead Sango...

hi, its possibe to "dont start" music on hold when caller (from sip operator trunk) press HOLD (i.e. on mobile phone) Asterisk acts on SDP a=sendonly i want pass trough media from SIP trunk provider Marek

there are new versions of Asterisk but mailing list is empty http://lists.digium.com/pipermail/asterisk-users/ Marek

hi, anybody using AEAP in production? https://www.asterisk.org/asterisk-external-application-protocol-speech-to-text-engine/ i'm trying, but i have problems with opus (no response from google) another problem is i need alaw and it looks like only ulaw is supported any experience with AEAP and STT from IBM?