Hi everyone. I have a need to run two separate Freeswitch PCs behind my router and I'm trying to figure out the best way to do this. For the first PC, in vars.xml I have set external_sip_port to 5060 and its TLS equivalent to 5061, and in switch.conf.xml I have set rtp-start-port to 17408 and rtp-end-port to 17920. For the second PC, in vars.xml I have set external_sip_port to 5040 and its TLS equivalent to 5041, and in switch.conf.xml I have set rtp-start-port to 16384 and rtp-end-port to 16896. On the router I have opened up UDP port 5060 and UDP ports 17408 to 17920 to the IP address of the first PC. I have opened up UDP port 5040 and UDP ports 16384 to 16896 to the IP address of the second PC. Things are half working. But when I phone in on a number that should connect to the first PC, the call goes through and I can answer it, but I'm only getting audio one way. Is there more I have to do to get Freeswitch to only use a certain range of ports for RTP so I can route correctly and avoid clashes? What am I missing here. Cheers. Clive Lansink Email: Clive@Lansink.Co.NZ Phone: +64 9 520-4242 Mobile: +64 21 663-999 Fax: +64 21 789-150
Hey Clive, According to what you said, you didn't set the external ip params on your profile. Try adding this: <param name="ext-rtp-ip" value="auto-nat"/> <param name="ext-sip-ip" value="auto-nat"/> With this, your FS servers will send the correct IP on SIP and SDP and the audio should work both ways. Regards, Em ter., 1 de jun. de 2021 às 01:02, Clive Lansink <clive@lansink.co.nz> escreveu:
Hi everyone.
I have a need to run two separate Freeswitch PCs behind my router and I'm trying to figure out the best way to do this.
For the first PC, in vars.xml I have set external_sip_port to 5060 and its TLS equivalent to 5061, and in switch.conf.xml I have set rtp-start-port to 17408 and rtp-end-port to 17920. For the second PC, in vars.xml I have set external_sip_port to 5040 and its TLS equivalent to 5041, and in switch.conf.xml I have set rtp-start-port to 16384 and rtp-end-port to 16896. On the router I have opened up UDP port 5060 and UDP ports 17408 to 17920 to the IP address of the first PC. I have opened up UDP port 5040 and UDP ports 16384 to 16896 to the IP address of the second PC. Things are half working. But when I phone in on a number that should connect to the first PC, the call goes through and I can answer it, but I'm only getting audio one way.
Is there more I have to do to get Freeswitch to only use a certain range of ports for RTP so I can route correctly and avoid clashes? What am I missing here.
Cheers.
Clive Lansink Email: Clive@Lansink.Co.NZ Phone: +64 9 520-4242 Mobile: +64 21 663-999 Fax: +64 21 789-150
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com
-- *Carlos E. Wagner* *Tecnólogo em Telecomunicações, Opensips Certified Professional* *Fone: +55 48 99981-0894* *E-mail:* kaduww@gmail.com *LinkedIn:* https://www.linkedin.com/in/carlos-eduardo-wagner-96bbb433/
one way audio because RTP packets to from phone do not reach your FreeSwitch. Looks as you have allowed call signaling and RTP packets to pass on the router. But need to "forward all UDP packets from RANGE1 to FS1 and from RANGE2 to FS2". you need to use iptables DNAT. On Tue, Jun 1, 2021 at 7:47 AM Carlos Eduardo <kaduww@gmail.com> wrote:
Hey Clive,
According to what you said, you didn't set the external ip params on your profile. Try adding this:
<param name="ext-rtp-ip" value="auto-nat"/> <param name="ext-sip-ip" value="auto-nat"/>
With this, your FS servers will send the correct IP on SIP and SDP and the audio should work both ways.
Regards,
Em ter., 1 de jun. de 2021 às 01:02, Clive Lansink <clive@lansink.co.nz> escreveu:
Hi everyone.
I have a need to run two separate Freeswitch PCs behind my router and I'm trying to figure out the best way to do this.
For the first PC, in vars.xml I have set external_sip_port to 5060 and its TLS equivalent to 5061, and in switch.conf.xml I have set rtp-start-port to 17408 and rtp-end-port to 17920. For the second PC, in vars.xml I have set external_sip_port to 5040 and its TLS equivalent to 5041, and in switch.conf.xml I have set rtp-start-port to 16384 and rtp-end-port to 16896. On the router I have opened up UDP port 5060 and UDP ports 17408 to 17920 to the IP address of the first PC. I have opened up UDP port 5040 and UDP ports 16384 to 16896 to the IP address of the second PC. Things are half working. But when I phone in on a number that should connect to the first PC, the call goes through and I can answer it, but I'm only getting audio one way.
Is there more I have to do to get Freeswitch to only use a certain range of ports for RTP so I can route correctly and avoid clashes? What am I missing here.
Cheers.
Clive Lansink Email: Clive@Lansink.Co.NZ Phone: +64 9 520-4242 Mobile: +64 21 663-999 Fax: +64 21 789-150
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com
-- *Carlos E. Wagner* *Tecnólogo em Telecomunicações, Opensips Certified Professional*
*Fone: +55 48 99981-0894* *E-mail:* kaduww@gmail.com *LinkedIn:* https://www.linkedin.com/in/carlos-eduardo-wagner-96bbb433/ _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com
participants (3)
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Carlos Eduardo
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Clive Lansink
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Sergey Safarov