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 had this working in chan_sip and using TIPCon1 soft-phone (TIPcon1 download | SourceForge.net). This call scenario was executed successfully as audio / text, video text and audio/video/text. Text meaning real time text as in ITU T.140. Another essential configuration on the soft-phone to make this work was to disable 401 unauthorized on the INVITE message, otherwise the TIPCon1 / SIPCon1 would not respond to an 401 unauthorized message!
This was possible by not providing any “secret” in the FreePBX extensions configurations.
However, using chan_PJSIP and following the same (blank secret) it does not disable the 401 unauthorized message, at least according to what I have experienced.
Question: Hoe do I disable the 401 unauthorized message in PJSIP? The reason being is the TIPCon1 soft-phone and other endpoints that would not respond to such a message.
Any input would be welcome!