Change audio port settings

[toolfolks]

Hi,

I have my fusionPBX running on our office server.

We have changed the sip port to 10060 ( for security ) and our TalkTalk router does not allow port forwarding of 5060.

So we port forward on the office router 10060 and can make calls between extensions outside of the domain but no sound.

So tired to port forward 16384 to 32768 but the router states:

The range of external port cannot include 7547,514,68,53,546,500,1701,8443,8080,990,5060,28090,50000-50020.

So I presume I have to change the RTP ports but I don't know which setting this is in the sip profile so I can port forward.

Also I cannot connect any IPPhones that are on the same network. What is the setting in rules.v4 to allow all devices on 192.168.1.xxxxx to connect.

 

[TheOperator]

The RTP ports can be changed in /etc/freeswitch/autoload_configs/switch.conf.xml. Look for:

<!-- RTP port range -->
<!-- <param name="rtp-start-port" value="16384"/> -->
<!-- <param name="rtp-end-port" value="32768"/> -->

My settings for a SOHO setup are:
<param name="rtp-start-port" value="20000"/>
<param name="rtp-end-port" value="20500"/>

Don't forget to restart freeswitch afterwards.

What error are you getting concerning registrations. Don't forget that sip clients should normally connect to the "internal" profile and you should not add your local networks to the "apply-inbound-acl" settings.

On my system I don't have a setting called "rules.v4" in my internal profiles.

 

[toolfolks]

Thanks for the info.
I have changed the /etc/freeswitch/autoload_configs/switch.conf.xml file and saved.
restarted freeswitch
Added the port forwarding.
I'm still not getting any audio.

I have the following port forwarding.

5432 (PostgreSql)
Sip 10060 TCP&UDP
Sip 5080 TCP&UDP
20000 - 20500 TCP&UDP

Anything else I'm missing ?

 

[ad5ou]

Have you updated your iptables on fusionpbx server to allow the new port ranges?

 

[toolfolks]

I have no changed the port setting in rules.v4.

Still no audio

ngrep shows from one app one way
#
U 192.168.1.2:20021 -> 2.99.151.76:4013 #19
..........................................9ac88cd5.%FreeSWITCH.org -- Come to ClueCon.com...
#
U 192.168.1.2:20021 -> 2.99.151.76:4013 #20
..........................................9ac88cd5.%FreeSWITCH.org -- Come to ClueCon.com...
#
U 192.168.1.2:20021 -> 2.99.151.76:4013 #21
..........................................9ac88cd5.%FreeSWITCH.org -- Come to ClueCon.com...
^Z

Reversed......


#
U 192.168.1.2:20347 -> 2.99.151.76:4015 #1
......`C..............................`C..baca6043.%FreeSWITCH.org -- Come to ClueCon.com...
#
U 192.168.1.2:20347 -> 2.99.151.76:4015 #2
......`C..............................`C..baca6043.%FreeSWITCH.org -- Come to ClueCon.com...

 

[TheOperator]

Could be NAT related. From what I can see you have 192.168.x.x as your internal PBX IP. Alas I can not see your external (public) IP.

I have a similar setup with my PBX on premisis and the following settings in my ipV4 external profile im Advanced -> variables -> IP Address:

external_rtp_ip: autonat:[external IP on router]
external_sip_ip: autonat:[external IP on router]

And in the profile:
rtp-ip: $${local_ip_v4}
sip-ip: $${local_ip_v4}

As far as your port setup is concerned - RTP only uses UDP.
And make sure that any SIP ALG is disabled.

if you use sngrep instead of ngrep you will get more details as to the ip information passed on in the SIP headers.

 

[toolfolks]

And in the profile:
rtp-ip: autonat: $${local_ip_v4}
sip-ip: $${local_ip_v4}

Is the the internal sip profile ?
Is the setting for rtp-ip = 'autonat: $${local_ip_v4}'

 

[TheOperator]

Yes, internal sip profile - had a copy&paste error there, just updated the post.

 

[toolfolks]

Got that working now.
I had to restart freeswitch.
Thanks for the help.

Worldwide domination here we come.........