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DTLS and media description proto
by Johan Pascal 08 Jun '21

08 Jun '21

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Freeswitch status stuck at 600 sessions with no traffic
by Abdirahman Osman 08 Jun '21

08 Jun '21

Hello, My maximum session is set to 600 in the switch.conf.xml, but freeswitch status is stuck at the moment 600 concurrent calls although I don't have a single call on the freeswitch. freeswitch@kvs> status UP 0 years, 2 days, 7 hours, 25 minutes, 30 seconds, 810 milliseconds, 872 microseconds FreeSWITCH (Version 1.10.5 -release-17-25569c1631 64bit) is ready 747 session(s) since startup 600 session(s) - peak 600, last 5min 600 0 session(s) per Sec out of max 30, peak 2, last 5min 1 600 session(s) max min idle cpu 0.00/78.60 Current Stack Size/Max 240K/8192K This is the error I am getting on Freeswitch [DESTINATION_OUT_OF_ORDER] 2021-06-04 19:05:44.404930 [CRIT] switch_core_session.c:2363 Over Session Limit! 600 2021-06-04 19:05:44.404930 [CRIT] mod_sofia.c:4730 Error Creating Session 2021-06-04 19:05:44.404930 [NOTICE] switch_ivr_originate.c:2999 Cannot create outgoing channel of type [sofia] cause: [DESTINATION_OUT_OF_ORDER] 2021-06-04 19:05:44.404930 [DEBUG] switch_ivr_originate.c:3995 Originate Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER] 3:07 Thanks! Abdirahman

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FS_Dialer
by Praveenkumar T 01 Jun '21

01 Jun '21

Dear, Please let me know if any dialer module is available from FS, if one wants to setup an automated outbound call centre. Thanks & Regards Praveen

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Re: [Freeswitch-users] Two Freeswitch installations
by Clive Lansink 01 Jun '21

01 Jun '21

Thanks Mike. That is what I thought. So for the two instances of Freeswitch I want to run, by setting these parameters to ranges that don't overlap, you're saying I can be confident that each instance will not offer local ports that clash with the other instance. Each instance will only listen on an RTP port that is within the range set by rtp-start-port and rtp-end-port. So that means I just need to ensure the router routes those ports correctly. I think I've done that but I will look at that more closely. Cheers. Clive Lansink Email: Clive(a)Lansink.Co.NZ Phone: +64 9 520-4242 Mobile: +64 21 663-999 Fax: +64 21 789-150 -----Original message----- From: Mike Jerris <mike(a)freeswitch.org> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Subject: Re: [Freeswitch-users] Two Freeswitch installations Reply-to: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Date: Tue, 1 Jun 2021 09:37:58 -0600 Ports are negotiated, the config is for the local ports offered in that negotiation. > On Jun 1, 2021, at 4:25 AM, Clive Lansink <clive(a)lansink.co.nz> wrote: > > OK following on from previous messages, can someone clarify exactly how Freeswitch RTP parameters work, rtp-start-port and rtp-end-port in switch.conf.xml. > > I only know the basics of SIP negotiation but I understand the two end points need to agree on RTP ports to use. That means each side must transmit RtP UDP packets to a port the other side has designated it will receive on. I presume these parameters determine the range of ports Freeswitch could listen on. When the remote end point is somewhere else on the internet, I presume the external profile does an address translation so the negotiation is done in terms of the public IP address Freeswitch knows about. > > So I have two Freeswitch servers behind the router. I've set these parameters to ranges that don't clash. And I've set the servers to use different external SIP ports. I've set the router to forward UDP packets from the outside so each range of ports goes to the appropriate server. > > I should point out that until recently, I've only had the one Freeswitch server and I never changed these RTP parameters. I simply opened up UDP ports 16384 to 32767 on the router to go to the Freeswitch server and everything worked properly so I haven't had to give this any thought. It's only because I now have a need to run a separate Freeswitch server that I now have this problem. > > More thoughts much appreciated. > > > Clive Lansink > Email: Clive(a)Lansink.Co.NZ > Phone: +64 9 520-4242 > Mobile: +64 21 663-999 > Fax: +64 21 789-150 > > -----Original message----- > From: "Bote Man" <botelist(a)gmail.com> > To: "'FreeSWITCH Users Help'" <freeswitch-users(a)lists.freeswitch.org> > Subject: Re: [Freeswitch-users] Two Freeswitch installations > Reply-to: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Date: Tue, 1 Jun 2021 03:07:09 -0400 > > "auto-nat" employs UPnP to communicate with your router and which is widely considered a security hole. > > This wiki page discusses NAT pretty well: > https://freeswitch.org/confluence/display/FREESWITCH/NAT > > Keep in mind that those X-PRE-PROCESS lines are only processed at start time. > > Also, check that you do not have the -nonat switch on the freeswitch process start command. > > > John Boteler > BnC Group U.S.A. > > > > -----Original Message----- > From: FreeSWITCH-users <freeswitch-users-bounces(a)lists.freeswitch.org> On Behalf Of Clive Lansink > Sent: Tuesday, 1 June, 2021 02:05 > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Subject: Re: [Freeswitch-users] Two Freeswitch installations > > Hi Carlos. > > Thanks for that. I thought before I change this, I'd let you know the current settings. > > So in one server, in the file SIP profiles\external.xml, there is: > <param name="rtp-ip" value="$${local_ip_v4}"/> > <param name="sip-ip" value="$${local_ip_v4}"/> > <param name="ext-rtp-ip" value="<static IP address>"/> > <param name="ext-sip-ip" value="<static IP address>"/> So that appears to be correct. I think at some stage I must have hard coded those static public IP addresses. > > For the other server, that same file shows: > <param name="rtp-ip" value="$${local_ip_v4}"/> > <param name="sip-ip" value="$${local_ip_v4}"/> > <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> > <param name="ext-sip-ip" value="$${external_sip_ip}"/> So this file is picking up the global variables. I've found these defined in vars.xml: > <X-PRE-PROCESS cmd="stun-set" data="external_rtp_ip=stun:stun.freeswitch.org"/> > <X-PRE-PROCESS cmd="stun-set" data="external_sip_ip=stun:stun.freeswitch.org"/> > I presume the STUN server would return the same static IP address. > > So do you think it would make any difference setting these to auto-nat? > > Cheers. > > > Clive Lansink > Email: Clive(a)Lansink.Co.NZ > Phone: +64 9 520-4242 > Mobile: +64 21 663-999 > Fax: +64 21 789-150 > > -----Original message----- > From: Carlos Eduardo <kaduww(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Subject: Re: [Freeswitch-users] Two Freeswitch installations > Reply-to: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Date: Tue, 1 Jun 2021 01:10:26 -0300 > > 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(a)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. _________________________________________________________________________ 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(a)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(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com

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Two Freeswitch installations
by Clive Lansink 01 Jun '21

01 Jun '21

OK following on from previous messages, can someone clarify exactly how Freeswitch RTP parameters work, rtp-start-port and rtp-end-port in switch.conf.xml. I only know the basics of SIP negotiation but I understand the two end points need to agree on RTP ports to use. That means each side must transmit RtP UDP packets to a port the other side has designated it will receive on. I presume these parameters determine the range of ports Freeswitch could listen on. When the remote end point is somewhere else on the internet, I presume the external profile does an address translation so the negotiation is done in terms of the public IP address Freeswitch knows about. So I have two Freeswitch servers behind the router. I've set these parameters to ranges that don't clash. And I've set the servers to use different external SIP ports. I've set the router to forward UDP packets from the outside so each range of ports goes to the appropriate server. I should point out that until recently, I've only had the one Freeswitch server and I never changed these RTP parameters. I simply opened up UDP ports 16384 to 32767 on the router to go to the Freeswitch server and everything worked properly so I haven't had to give this any thought. It's only because I now have a need to run a separate Freeswitch server that I now have this problem. More thoughts much appreciated. Clive Lansink Email: Clive(a)Lansink.Co.NZ Phone: +64 9 520-4242 Mobile: +64 21 663-999 Fax: +64 21 789-150 -----Original message----- From: "Bote Man" <botelist(a)gmail.com> To: "'FreeSWITCH Users Help'" <freeswitch-users(a)lists.freeswitch.org> Subject: Re: [Freeswitch-users] Two Freeswitch installations Reply-to: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Date: Tue, 1 Jun 2021 03:07:09 -0400 "auto-nat" employs UPnP to communicate with your router and which is widely considered a security hole. This wiki page discusses NAT pretty well: https://freeswitch.org/confluence/display/FREESWITCH/NAT Keep in mind that those X-PRE-PROCESS lines are only processed at start time. Also, check that you do not have the -nonat switch on the freeswitch process start command. John Boteler BnC Group U.S.A. -----Original Message----- From: FreeSWITCH-users <freeswitch-users-bounces(a)lists.freeswitch.org> On Behalf Of Clive Lansink Sent: Tuesday, 1 June, 2021 02:05 To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Subject: Re: [Freeswitch-users] Two Freeswitch installations Hi Carlos. Thanks for that. I thought before I change this, I'd let you know the current settings. So in one server, in the file SIP profiles\external.xml, there is: <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="<static IP address>"/> <param name="ext-sip-ip" value="<static IP address>"/> So that appears to be correct. I think at some stage I must have hard coded those static public IP addresses. For the other server, that same file shows: <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> So this file is picking up the global variables. I've found these defined in vars.xml: <X-PRE-PROCESS cmd="stun-set" data="external_rtp_ip=stun:stun.freeswitch.org"/> <X-PRE-PROCESS cmd="stun-set" data="external_sip_ip=stun:stun.freeswitch.org"/> I presume the STUN server would return the same static IP address. So do you think it would make any difference setting these to auto-nat? Cheers. Clive Lansink Email: Clive(a)Lansink.Co.NZ Phone: +64 9 520-4242 Mobile: +64 21 663-999 Fax: +64 21 789-150 -----Original message----- From: Carlos Eduardo <kaduww(a)gmail.com> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Subject: Re: [Freeswitch-users] Two Freeswitch installations Reply-to: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Date: Tue, 1 Jun 2021 01:10:26 -0300 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(a)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(a)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(a)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(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > https://freeswitch.com -- *Carlos E. Wagner* *Tecnólogo em Telecomunicações, Opensips Certified Professional* *Fone: +55 48 99981-0894* *E-mail:* kaduww(a)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(a)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(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com _________________________________________________________________________ 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(a)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(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com

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Re: [Freeswitch-users] Two Freeswitch installations
by Clive Lansink 01 Jun '21

01 Jun '21

Hi Carlos. Thanks for that. I thought before I change this, I'd let you know the current settings. So in one server, in the file SIP profiles\external.xml, there is: <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="<static IP address>"/> <param name="ext-sip-ip" value="<static IP address>"/> So that appears to be correct. I think at some stage I must have hard coded those static public IP addresses. For the other server, that same file shows: <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> So this file is picking up the global variables. I've found these defined in vars.xml: <X-PRE-PROCESS cmd="stun-set" data="external_rtp_ip=stun:stun.freeswitch.org"/> <X-PRE-PROCESS cmd="stun-set" data="external_sip_ip=stun:stun.freeswitch.org"/> I presume the STUN server would return the same static IP address. So do you think it would make any difference setting these to auto-nat? Cheers. Clive Lansink Email: Clive(a)Lansink.Co.NZ Phone: +64 9 520-4242 Mobile: +64 21 663-999 Fax: +64 21 789-150 -----Original message----- From: Carlos Eduardo <kaduww(a)gmail.com> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Subject: Re: [Freeswitch-users] Two Freeswitch installations Reply-to: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Date: Tue, 1 Jun 2021 01:10:26 -0300 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(a)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(a)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(a)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(a)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(a)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(a)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(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com

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Two Freeswitch installations
by Clive Lansink 01 Jun '21

01 Jun '21

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(a)Lansink.Co.NZ Phone: +64 9 520-4242 Mobile: +64 21 663-999 Fax: +64 21 789-150

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Unregister all devices
by jean rouquet 29 May '21

29 May '21

Hello, If bob is registered from two devices, is it possible from one device to cancel both registrations? regards, Jean

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Integration with Opensips
by HS 29 May '21

29 May '21

Hi all. Just been trying to integrate Opensips with Freeswitch by following this old guide: https://www.opensips.org/Documentation/Tutorials-OpenSIPSFreeSwitchIntegrat… I think I have managed to get the other stuff working alright. However, it seems that the user extensions aren't being pulled from the opensips database. When I run: list_users group default I get the following error: 2021-05-25 02:57:32.297372 [ERR] mod_lua.cpp:202 /usr/share/freeswitch/scripts/xml_handler.lua:40: attempt to index global 'params' (a nil value) stack traceback: /usr/share/freeswitch/scripts/xml_handler.lua:40: in main chunk 2021-05-25 02:57:32.297372 [ERR] mod_lua.cpp:270 LUA script parse/execute error! I am using Opensips 3.0 on Amazon EC2. Any help would be appreciated. Best wishes.

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Re: [Freeswitch-users] rtp-timeout-sec VS media_timeout
by David P 28 May '21

28 May '21

We're using FS 10.5 on Debian Stretch installed via packages. <param name="rtp-timeout-sec" value="300"/> in our sip_profiles/internal.xml does effect calls. We have no media_timeout* config.

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