Use your PBX web interface to edit your Flowroute Trunk. 25; "config show help res_pjsip", then you can drill down through the various The first day, I made my configurations and all chan_sip and chan_pjsip extensions were working fine. 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/blob - configs/samples ... [SOLVED] How to disable directmedia in all pjsip endpoints Set the port to 5160 through one of the following methods: In /etc/asterisk/, open sip.conf with a text editor; or. projects / asterisk/asterisk.git / history commit grep author committer pickaxe ? 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. It was done in a generic fashion though so other modules could use it and additional . This is particularly useful when switching between different versions of Asterisk that may have introduced . When PJSIP was being written it was decided that a new data (not specifically configuration) layer would be written. Asterisk PJSIP Troubleshooting Guide - Asterisk Project Wiki Example of old configuration: alwaysauthreject=yes domain = asterisk.mydomain.com allowexternaldomains = no allowguest = no deny=0.0.0.0/0.0.0.0 permit=10.0.0.0/255.0.0.0 And as the author of the question wrote, you have to use a firewall and fail2ban. Sorcery. I'm trying write softphone app with pjsua. As of Asterisk 13.8.0 another simpler option will be available instead: bundling. In the asterisk advanced settings there is a section for device defaults. res_pjsip: Default endpoints to the "offline" status. Asterisk 13.8.0: Now With Easier PJSIP Install Method! As of Asterisk 17's release, there will be at least a 4-year time frame before the potential removal of chan_sip from Asterisk may happen. PSA: chan_sip status changed to "deprecated" & Asterisk 17.0.0-rc2 Release

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