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@example.com:41856;transport b131d6fe7e Avail 105.646
Contact: 052002/sips:firstname.lastname@example.org: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
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 Fernández Aparicio
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