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Issues with calls over TLS on FreePBX (Asterisk)

Issues with calls over TLS on FreePBX (Asterisk)

Owned by Alex | Ringotel Team, created
Last updated: Apr 09, 2023
1 min read

Problem

Users register normally but calls are rejected from PBX side with the code 488 Not Acceptable Here or no audio after the connection is estanblished.

Solution

The problem occurs because of the protocol misconfiguration on the PBX side.

To solve the issue with the calls, go to the extension settings and open the Advanced tab.

If PJSIP extension:

  1. Set Medial encryption parameter to SRTP via in-SDP

  2. Ensure that the DTLS is disabled

If CHAN_SIP extension:

  1. Set Enable Encryption to Yes

  2. Ensure that the DTLS is disabled

Submit and Apply changes

During the call from PBX to Shell user, there were no ciphers presented in SDP session description in INVITE package from PBX side. Should be like this:

v=0

o=5001 1593011580 1 IN IP4 52.213.1.74

s=SIP Call

c=IN IP4 52.213.1.74

t=0 0

m=audio 10034 RTP/SAVP 8

a=rtpmap:8 PCMA/8000

a=ptime:20

a=sendrecv

a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:iuKEYTgW5dva6Zk6eTeKsXHdUfVrEmTArW82/Mq+

a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:S4FBRSR3ZcWJm4ISXgjblGtPWTEmWl149GX9RmEJ——

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