Hi everyone, I'm building a PABX system with Asterisk and FreePBX and I need trustworthy phones to perform tests, that's why I'm using Linphone. The problem is that I need to secure all communications so I'm using `sips` for the URIs, however, I can see in the asterisk logger a mix between sip and sips schemes.
Why is this happening? Apparently once I use the sips scheme all communications should be forced to also use sips. As you can see in this asterisk log https://jfernandz.me/~wyre/linphone-linphone_2.log, both contacts (phones) are apparently registered using the sips scheme: asc3*CLI> pjsip show contacts Contact: <Aor/ContactUri..............................> <Hash....> <Status> <RTT(ms)..> ========================================================================================== Contact: 035000/sips:[email protected]:41856;transport b131d6fe7e Avail 105.646 Contact: 052002/sips:[email protected]:45504;transport= 2a6084e8d3 Avail 131.221 Objects found: 2 I've placed a test call between two linphone clients and you can see a mix between sip and sips schemes, for example, I can see Contact: <sip:[email protected]:41856 ;transport=tls>;expires=599;+sip.instance="<urn:uuid:5e85b13b-b9e6-008a-b342-fdcb81341b3a>";+org.linphone.specs="ephemeral,groupchat/1.1,lime" I think this could be causing the placed call is hungup immediately. Is there some way to force sips usage in your Linphone client app? Thank you all. BR. Javier. *Javier Fernández Aparicio* *DevOps Engineer* Headquarters - Paseo Castellana, 200 - SPACES - Madrid 28046 ES Labs - Calle Innovación, 17 - Getafe, Madrid 28906 ES https://www.joifilabs.com/
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