I have a functioning Freepbx 184.108.40.206 system. We are totally soft phone, which work flawlessly. We are trying to add a few Sangoma P310 into the system to see how they work. We have had little luck on this TBH, but got one to connect.
we are able to dial but no audio is heard from the phone. Even when we try DTMF in VM.
The phones are inside a NAT but the PBX is DMZd.
The Sangoma Phones are at Firmware 3.5.1 which as far as I can tell is the latest… from my research it sounds like this is something to do with the firmware on the phone?? or??
im seeing these entries in the log during the call…
[2021-11-04 07:20:38] VERBOSE[C-00000019] res_rtp_asterisk.c: 0x7fa95004de10 – Strict RTP learning complete - Locking on source address 192.168.1.35:4012
[2021-11-04 07:20:38] VERBOSE[C-00000019] res_srtp.c: SRTCP unprotect failed on SSRC 162886699 because of authentication failure
[2021-11-04 07:20:43] VERBOSE[C-00000019] res_srtp.c: SRTCP unprotect failed on SSRC 162886699 because of unable to perform desired validation
[2021-11-04 07:20:48] VERBOSE[C-00000019] res_srtp.c: SRTCP unprotect failed on SSRC 162886699 because of unable to perform desired validation
Those particular errors are media encryption related. Perhaps there’s an issue with how the encryption keys are being setup? I’d call Sangoma tech support to figure out further why that might be happening as it could be a number of issues.
I was hoping this wasn’t the answer… i’ve had a ticket in for over a week without being touched on another issue…
Thanks, ill go that route…
Are you using a dynamic IP address?
Not on the server, but on the phones yes…
I had a similar issue that is connected to the changing dynamic IP address. But your problem seems to be somewhere else.
Does it work with UDP? You’re using PJSIP?
Could you make a call while pjsiplogger is activated and show the result of it? You can replace sensitive data like IP addresses with placeholders.
I was able to fix this by removing as many codecs as possible. I haven’t confirmed this, but when I reduced them to under 10, then he call audio worked… MTU maybe?
You removed them from FreePBX or the phone?
in “Settings” --> “Asterisk SIP Settings” --> “General SIP Settings” --> “Audio Codecs”.
The Save, Apply config and to be safe restarted Asterisk…
Then in EPM, Did a rebuild and Update Phones…
This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.