The trunk seems to always negotiate to G729, so Asterisk ends up transcoding the ulaw to G729 between the two, and faxes have lots of issues. Asterisk PJSIP Troubleshooting Guide - Asterisk Project Wiki ael set debug {read|tokens|macros|contexts|off} -- Enable AEL debugging flags. git.asterisk.org Git - asterisk/asterisk.git/blob - res/res_pjsip.c Asterisk new PJSIP driver security option - Server Fault Bundling allows a self-contained PJSIP to exist within Asterisk and be used by all functionality within it. The first day, I made my configurations and all chan_sip and chan_pjsip extensions were working fine. git.asterisk.org Git - asterisk/asterisk.git/log (typically /etc/asterisk/) noload => res_pjsip.so noload => res_pjsip_pubsub.so noload => res_pjsip_session.so noload => chan_pjsip.so Asterisk 14: Coming with improved PJSIP DNS Support! CHAN_SIP / CHAN_PJSIP 401 Unauthorized on INVITE ncinta December 20, 2021, 7:26pm #1 Brief Description - Asterisk supports RTT (real time text) with an addition of a couple statements in the configuration files, at least using chan_sip. I'm trying write softphone app with pjsua. ; First, manually written examples to serve as a handy reference. Everything works well, sound is transmitted bidirectional, when I use Asterisk and softphones in the same local network - 192.168.10.XXX, but when I hide my softphone behind NAT, I can't hear any incoming sound, outcoming sound works OK. git.asterisk.org Git - asterisk/asterisk.git/log PJSIP-pjproject - Asterisk Project - Asterisk Project Wiki No voice transmission, PJSIP behind NAT - Stack Overflow
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