No audio with all configuration made

General Help
1
  • I have a successful FreePBX 13 installation on Debian as per the steps in FreePBX wiki, I configured 2 extensions, where both can connect and place a call but can not hear each other, also if one party end the call then the call is not ended on the other party.
  • I want the server and the clients to use TLS and SRTP only using non standard TLS and RTP ports, also I created FreePBX self-signed-certificate and applied in FreePBX settings.
  • My FreePBX PC connect to a wireless router (LAN 192.168.1.xx) that is connected to internet (WAN) via static IP (so I believe I am behind NAT)
  • I went through FreePBX SIP audio issues wiki with no luck to resolve it
  • Here are all of the configuration I have edited where I left the rest to their default values:

Advanced Settings

  • Device Settings
    — SIP nat = yes
    — SIP encryption = yes

Asterisk SIP Settings >> General SIP Settings

  • NAT Settings
    — External Address = (My Public IP)
    — Local Networks = 192.168.1.0 / 255.255.255.0
  • RTP Settings
    — RTP Port Ranges = start/end = custom range but not (10000-20000)

Asterisk SIP Settings >> Chan SIP Settings

  • NAT Settings
    — NAT = Yes
  • TLS/SSL/SRTP Settings
    — Enable TLS = yes
    — Certificate Manager = default (self-signed generated by FreePBX certificate management)
    — SSL Method = sslv2
  • Advanced General Settings
    — Bind port = custom port but not 5060
    — TLS bind port = custom port but not 5061

Extensions >> Advanced

  • Edit Extension
    — NAT Mode = yes (force_rport,comedia)
    — Port = custom port which is the same as default TLS port
    — Transport = TLS only
    — Enable Encryption = yes (SRTP only)

  • I have forwarded all the above custom ports from my router to the FreePBX PC.

  • I allowed all traffic in debian firewall.

  • As I used custom RTP ports, I have un-comment the line “include rtp_additional.conf” from /etc/asterisk/rtp.conf and I assured that rtp_additional.conf contains my custom RTP start/end ports.

  • I restarted asterisk after every change.

  • I configured 2 android phones using Zoiper to connect to FreePBX externally over 3g/4g network

Please let me know if there are extra settings to do.

2

If using non default TLS port, in the following section:
Asterisk SIP Settings >> Chan SIP Settings >> Other SIP Settings :
Add the following:
externtlsport= (Your non default TLS port)

3

Hi all…

any solution regarding this issue?

thanks in advance

4

Not really - the first is a common problem when interacting with a POTS phone through a VOIP system.

The second requires some specific steps (that have been covered in detail in other posts in these fora) including limitations on certificate authorities and other features of TLS and SRTP.

5

For the first issue, i just found =

  1. The endpoint (the extension) softphone or handset can established the call and make the conversation with success( no drop) if the ip address of the extension same to the subnet of freepbx server.But, the extension will established the call and the calling will dropped after 31-32 second, if the extension is in the another subnet with the freepbx server.

  2. The endpoint (the extension) softphone or handset that has the same subnet with the freepbx server when make and established a call to another extension and when the extension desconected the call, the call will be disconected in each extension. But, it will not working if the extension did and established a call from another subnet of the freepbx server( when we disconected from 1 side, the other side wont disconnected).

And second issue, i still have not gotten the solution.

Thanks