From clive at lansink.co.nz Tue Jun 1 04:01:33 2021 From: clive at lansink.co.nz (Clive Lansink) Date: Tue, 1 Jun 2021 04:01:33 +0000 Subject: [Freeswitch-users] Two Freeswitch installations Message-ID: <01000179c5bbdb9b-5af3052c-7c70-4381-9643-ab8e28c5b2ef-000000@email.amazonses.com> An embedded and charset-unspecified text was scrubbed... Name: not available URL: From kaduww at gmail.com Tue Jun 1 04:10:26 2021 From: kaduww at gmail.com (Carlos Eduardo) Date: Tue, 1 Jun 2021 01:10:26 -0300 Subject: [Freeswitch-users] Two Freeswitch installations In-Reply-To: <01000179c5bbdb9b-5af3052c-7c70-4381-9643-ab8e28c5b2ef-000000@email.amazonses.com> References: <01000179c5bbdb9b-5af3052c-7c70-4381-9643-ab8e28c5b2ef-000000@email.amazonses.com> Message-ID: Hey Clive, According to what you said, you didn't set the external ip params on your profile. Try adding this: 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 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 at 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 at 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 at 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 at gmail.com *LinkedIn:* https://www.linkedin.com/in/carlos-eduardo-wagner-96bbb433/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From clive at lansink.co.nz Tue Jun 1 06:05:09 2021 From: clive at lansink.co.nz (Clive Lansink) Date: Tue, 1 Jun 2021 06:05:09 +0000 Subject: [Freeswitch-users] Two Freeswitch installations Message-ID: <01000179c62d041d-57c94897-7a2f-47d1-93e2-35a01dab3897-000000@email.amazonses.com> An embedded and charset-unspecified text was scrubbed... Name: not available URL: From s.safarov at gmail.com Tue Jun 1 06:12:31 2021 From: s.safarov at gmail.com (Sergey Safarov) Date: Tue, 1 Jun 2021 09:12:31 +0300 Subject: [Freeswitch-users] Two Freeswitch installations In-Reply-To: References: <01000179c5bbdb9b-5af3052c-7c70-4381-9643-ab8e28c5b2ef-000000@email.amazonses.com> Message-ID: 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 wrote: > Hey Clive, > > According to what you said, you didn't set the external ip params on your > profile. Try adding this: > > > > > 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 > 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 at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From botelist at gmail.com Tue Jun 1 07:07:09 2021 From: botelist at gmail.com (Bote Man) Date: Tue, 1 Jun 2021 03:07:09 -0400 Subject: [Freeswitch-users] Two Freeswitch installations In-Reply-To: <01000179c62d041d-57c94897-7a2f-47d1-93e2-35a01dab3897-000000@email.amazonses.com> References: <01000179c62d041d-57c94897-7a2f-47d1-93e2-35a01dab3897-000000@email.amazonses.com> Message-ID: <011201d756b4$bd78edc0$386ac940$@gmail.com> "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 On Behalf Of Clive Lansink Sent: Tuesday, 1 June, 2021 02:05 To: FreeSWITCH Users Help 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: 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: So this file is picking up the global variables. I've found these defined in vars.xml: 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 at Lansink.Co.NZ Phone: +64 9 520-4242 Mobile: +64 21 663-999 Fax: +64 21 789-150 -----Original message----- From: Carlos Eduardo To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Two Freeswitch installations Reply-to: FreeSWITCH Users Help 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: 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 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 at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com From clive at lansink.co.nz Tue Jun 1 10:25:37 2021 From: clive at lansink.co.nz (Clive Lansink) Date: Tue, 1 Jun 2021 10:25:37 +0000 Subject: [Freeswitch-users] Two Freeswitch installations Message-ID: <01000179c71b79dc-b7ab8701-8f24-431d-8c5d-c36135a42d37-000000@email.amazonses.com> An embedded and charset-unspecified text was scrubbed... Name: not available URL: From mike at freeswitch.org Tue Jun 1 15:37:58 2021 From: mike at freeswitch.org (Mike Jerris) Date: Tue, 1 Jun 2021 09:37:58 -0600 Subject: [Freeswitch-users] Two Freeswitch installations In-Reply-To: <01000179c71b79dc-b7ab8701-8f24-431d-8c5d-c36135a42d37-000000@email.amazonses.com> References: <01000179c71b79dc-b7ab8701-8f24-431d-8c5d-c36135a42d37-000000@email.amazonses.com> Message-ID: <86B4FA4D-DEA7-4575-8DB7-909B41F7BBFB@freeswitch.org> Ports are negotiated, the config is for the local ports offered in that negotiation. > On Jun 1, 2021, at 4:25 AM, Clive Lansink 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 at Lansink.Co.NZ > Phone: +64 9 520-4242 > Mobile: +64 21 663-999 > Fax: +64 21 789-150 > > -----Original message----- > From: "Bote Man" > To: "'FreeSWITCH Users Help'" > Subject: Re: [Freeswitch-users] Two Freeswitch installations > Reply-to: FreeSWITCH Users Help > 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 On Behalf Of Clive Lansink > Sent: Tuesday, 1 June, 2021 02:05 > To: FreeSWITCH Users Help > 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: > > > > 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: > > > > So this file is picking up the global variables. I've found these defined in vars.xml: > > > 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 at Lansink.Co.NZ > Phone: +64 9 520-4242 > Mobile: +64 21 663-999 > Fax: +64 21 789-150 > > -----Original message----- > From: Carlos Eduardo > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Two Freeswitch installations > Reply-to: FreeSWITCH Users Help > 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: > > > > 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 > 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. From clive at lansink.co.nz Tue Jun 1 18:02:52 2021 From: clive at lansink.co.nz (Clive Lansink) Date: Tue, 1 Jun 2021 18:02:52 +0000 Subject: [Freeswitch-users] Two Freeswitch installations Message-ID: <01000179c8be1c61-a9706fbf-c045-4f1c-b5a3-d27ee03a33c5-000000@email.amazonses.com> An embedded and charset-unspecified text was scrubbed... Name: not available URL: From david.villasmil.work at gmail.com Tue Jun 1 18:42:29 2021 From: david.villasmil.work at gmail.com (David Villasmil) Date: Tue, 1 Jun 2021 19:42:29 +0100 Subject: [Freeswitch-users] FS_Dialer In-Reply-To: <003c01d75600$b87b3530$29719f90$@teleapps.com> References: <003c01d75600$b87b3530$29719f90$@teleapps.com> Message-ID: There are several in github. Or you can make your own. Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 On Tue, Jun 1, 2021 at 4:32 PM Praveenkumar T wrote: > 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 > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From d.mordovin at crestwavetech.ru Tue Jun 1 19:13:59 2021 From: d.mordovin at crestwavetech.ru (Dmitry Mordovin) Date: Tue, 1 Jun 2021 22:13:59 +0300 Subject: [Freeswitch-users] uuid_broadcast hang and start after the call hangup Message-ID: Hello I have LUA script which start a call (new session generated) Code here new_session = freeswitch.Session(destination); if (new_session:ready()) then new_session_disp = new_session:getVariable("endpoint_disposition"); while(new_session:ready() and new_session_disp ~= "ANSWER") do <------>new_session_disp = new_session:getVariable("endpoint_disposition"); end api = freeswitch.API(); api:executeString("bgapi uuid_park " .. tostring(new_session.uuid)); end On answer event I park the call and script finished. I see single active call. Listing below. freeswitch at freeswitch43> show calls uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,presence_id,presence_data,accountcode,callstate,callee_name,callee_num,callee_direction,call_uuid,hostname,sent_callee_name,sent_callee_num,b_uuid,b_direction,b_created,b_created_epoch,b_name,b_state,b_cid_name,b_cid_num,b_ip_addr,b_dest,b_presence_id,b_presence_data,b_accountcode,b_callstate,b_callee_name,b_callee_num,b_callee_direction,b_sent_callee_name,b_sent_callee_num,call_created_epoch 3aa3a274-c23d-11eb-9150-e74dc4cdb20a,outbound,2021-05-31 18:22:59,1622485379,sofia/external/1000,CS_PARK,,0000000000,,1000,,,,ACTIVE,Outbound Call,1000,,3aa3a274-c23d-11eb-9150-e74dc4cdb20a,freeswitch43,,,,,,,,,,,,,,,,,,,,,, 1 total. After all I execute command from cli: Command looks like: uuid_broadcast 3aa3a274-c23d-11eb-9a50-e74dc4cdb20a lua::’next-script.lua' both In logs I see, command was sent to FS success, but not executed! Moveover, after call hangup, FS start execution of command but can’t coz call in terminate state. 2021-05-31 18:23:30.312133 [NOTICE] sofia.c:1079 Hangup sofia/external/1000 [CS_PARK] [NORMAL_CLEARING] 2021-05-31 18:23:30.312133 [DEBUG] switch_ivr.c:625 sofia/external/1000 Command Execute lua(next-script.lua) 2021-05-31 18:23:30.312133 [DEBUG] switch_core_session.c:2668 sofia/external/1000 ZOMBIE EXEC lua(next-script.lua) EXECUTE sofia/external/1000 lua(next-script.lua) What/why is ZOMBIE EXEC ? Could someone help me to find reason of strange behavior FS? Looks like the uuid_broadcast command stay in queueu and wait something… and start after call has hangup. -------------- next part -------------- An HTML attachment was scrubbed... URL: From bullehs at gmail.com Sat Jun 5 06:29:28 2021 From: bullehs at gmail.com (HS) Date: Sat, 5 Jun 2021 11:29:28 +0500 Subject: [Freeswitch-users] Amazon EC2 Instances - Freeswitch Message-ID: Dear all, I have been testing an Opensips + Freeswitch setup on the same instance on EC2. I did follow the guide here for IP setup: https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2 However, for production, it seems better to have separate instances for Opensips and Freeswitch. (Or wrong?). What would be the following variables: bind_server_ip external_rtp_ip external_sip_ip Shall I use the Opensips external IP or Freeswitch (although Freeswitch will have no external access). Thanks for the help. -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Sat Jun 5 12:13:44 2021 From: david.villasmil.work at gmail.com (David Villasmil) Date: Sat, 5 Jun 2021 13:13:44 +0100 Subject: [Freeswitch-users] Amazon EC2 Instances - Freeswitch In-Reply-To: References: Message-ID: If freeswitch has no external (public) access, and you don’t have an rtpproxy/rtpengine, and you will be servicing clients on the public internet, the audio will no work. Your best bet for a simple setup is assign a public ip to freeswitch, allow all rtp port range (UDP) in and out of freeswitch, and set this public ip in external_rtp_ip. Everything else are the private IPs. On Sat, 5 Jun 2021 at 07:30, HS wrote: > Dear all, > > I have been testing an Opensips + Freeswitch setup on the same instance on > EC2. I did follow the guide here for IP setup: > https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2 > > However, for production, it seems better to have separate instances for > Opensips and Freeswitch. (Or wrong?). What would be the following variables: > > bind_server_ip > external_rtp_ip > external_sip_ip > > Shall I use the Opensips external IP or Freeswitch (although Freeswitch > will have no external access). > > Thanks for the help. > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From bullehs at gmail.com Sat Jun 5 14:40:31 2021 From: bullehs at gmail.com (HS) Date: Sat, 5 Jun 2021 19:40:31 +0500 Subject: [Freeswitch-users] Amazon EC2 Instances - Freeswitch In-Reply-To: References: Message-ID: David. Thanks a lot for taking the time to explain. I think in my hurry to post I forgot to add details. Instance 1 Opensips + RTPProxy (Registrar + routing). Instance 2 Freeswitch (IVR/Conferencing/VoiceMail etc.) Does that setup make sense? What would be the variables (bind_server_ip, external_rtp_ip, external_sip_ip) be please? Thanks again. -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Sat Jun 5 15:33:51 2021 From: david.villasmil.work at gmail.com (David Villasmil) Date: Sat, 5 Jun 2021 16:33:51 +0100 Subject: [Freeswitch-users] Amazon EC2 Instances - Freeswitch In-Reply-To: References: Message-ID: You need to configure rtpproxy in multihomed mode (bridge mode), so it does external/internal relay of rtp. Then opensips will instruct rtpproxy to do it. On freeswitch just configure the private IPs. That’s really all there is to it. If it’s configured correctly, you should see in the SDPs the following: - Client—[INVITE]—>opensips: the client will send its public IP (or maybe its private, but if opensips is doing NAT translation, shouldnt matter, but I suggest testing with a client which sends its public IP) - Opensips—>[INVITE]—>freeswitch: opensips should send rtpproxy’s private IP The on the 200 OKs: - Freeswitch—[200 OK]—>opesnsips: fs should send its private IP - Opensips—[200 OK]—>client: opensips should send rtpproxy’s public IP address. If you don’t see exactly this, something mis-configured somewhere. Hope this help. On Sat, 5 Jun 2021 at 15:41, HS wrote: > David. > > Thanks a lot for taking the time to explain. I think in my hurry to post I > forgot to add details. > > Instance 1 > Opensips + RTPProxy (Registrar + routing). > Instance 2 > Freeswitch (IVR/Conferencing/VoiceMail etc.) > > Does that setup make sense? What would be the variables (bind_server_ip, > external_rtp_ip, external_sip_ip) be please? > > Thanks again. > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From nhall at unixlan.com.ar Sat Jun 5 18:58:13 2021 From: nhall at unixlan.com.ar (Normando Hall) Date: Sat, 5 Jun 2021 15:58:13 -0300 Subject: [Freeswitch-users] Amazon EC2 Instances - Freeswitch In-Reply-To: References: Message-ID: <8cd42b75-f2e4-ca2d-10d5-9612fa1431ff@unixlan.com.ar> A simple and secure workaround, use a VPN like OpenVPN. No open ports, no proxies, and it's secure and fast. Works on desktop, mobiles, etc. Install OpenVPN in your server, and each client. Today every sip phone can use openvpn in its setup. Regards El 05/06/2021 a las 03:29, HS escribió: > Dear all, > > I have been testing an Opensips + Freeswitch setup on the same > instance on EC2. I did follow the guide here for IP setup: > https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2 > > > However, for production, it seems better to have separate instances > for Opensips and Freeswitch. (Or wrong?). What would be the following > variables: > > bind_server_ip > external_rtp_ip > external_sip_ip > > Shall I use the Opensips external IP or Freeswitch (although > Freeswitch will have no external access). > > Thanks for the help. > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From nhall at unixlan.com.ar Sat Jun 5 19:19:03 2021 From: nhall at unixlan.com.ar (Normando Hall) Date: Sat, 5 Jun 2021 16:19:03 -0300 Subject: [Freeswitch-users] Amazon EC2 Instances - Freeswitch In-Reply-To: References: Message-ID: <16c77719-68e2-d886-28ed-12b8992e8fe6@unixlan.com.ar> A simple and secure workaround, use a VPN like OpenVPN. No open ports, no proxies, and it's secure and fast. Works on desktop, mobiles, etc. Install OpenVPN in your server, and each client. Today every sip phone can use openvpn in its setup. Regards PD: Sorry duplicate email, because previous wrong sended El 05/06/2021 a las 03:29, HS escribió: > Dear all, > > I have been testing an Opensips + Freeswitch setup on the same > instance on EC2. I did follow the guide here for IP setup: > https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2 > > > However, for production, it seems better to have separate instances > for Opensips and Freeswitch. (Or wrong?). What would be the following > variables: > > bind_server_ip > external_rtp_ip > external_sip_ip > > Shall I use the Opensips external IP or Freeswitch (although > Freeswitch will have no external access). > > Thanks for the help. > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From bullehs at gmail.com Sun Jun 6 07:31:41 2021 From: bullehs at gmail.com (HS) Date: Sun, 6 Jun 2021 12:31:41 +0500 Subject: [Freeswitch-users] Amazon EC2 Instances - Freeswitch In-Reply-To: References: Message-ID: Hi again. David - thx a lot, will try and revert with issues. I just saw that you are on the Opensips list also :) Norman - thx for the suggestion. However, I am quite new to this - unsure what problem OpenVPN is meant to solve in my scenario. Best wishes. -------------- next part -------------- An HTML attachment was scrubbed... URL: From johan.pascal at linphone.org Mon Jun 7 07:55:42 2021 From: johan.pascal at linphone.org (Johan Pascal) Date: Mon, 7 Jun 2021 09:55:42 +0200 Subject: [Freeswitch-users] DTLS and media description proto Message-ID: <5422ff67-fb60-b77d-631c-3ed9211ad5ab@linphone.org> An HTML attachment was scrubbed... URL: From brian at freeswitch.com Mon Jun 7 14:35:23 2021 From: brian at freeswitch.com (Brian West) Date: Mon, 7 Jun 2021 09:35:23 -0500 Subject: [Freeswitch-users] DTLS and media description proto In-Reply-To: <5422ff67-fb60-b77d-631c-3ed9211ad5ab@linphone.org> References: <5422ff67-fb60-b77d-631c-3ed9211ad5ab@linphone.org> Message-ID: Try setting webrtc_enable_dtls=true in {} on the originate. On Mon, Jun 7, 2021 at 8:47 AM Johan Pascal wrote: > Hi, > > I'm trying to get freeswitch to direct a call to a linphone client using > DTLS, as described here: > > > https://freeswitch.org/confluence/display/FREESWITCH/Certificates#Certificates-DTLS > > > The A leg with a DTLS client works fine but the B leg, while enabling > correctly DTLS in the SDP INVITE(fingerprint and setup attributes) produces > a media description using the proto "RTP/SAVPF". > > Linphone does not accept this proto as a DTLS call but expect > "UDP/TLS/RTP/SAVPF". I'm using freeswitch v1.10.6, I found in the > freeswitch code there > > > https://github.com/signalwire/freeswitch/blob/v1.10.6/src/switch_core_media.c#L9715 > > a test to switch to the expected proto but I can't find a way in the > freeswitch settings/dialplan to get the CF_AVPF_MOZ flag on. When forcing > the outcome of the test to be always true, I get my particular use case to > work but I guess there is a more harmless way to proceed. Any suggestion? > > Thanks, > > Johan > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Brian West | Co-founder and Developer Need Commercial support? email sales at freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: brian at freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com [image: https://www.facebook.com/signalwireinc?src=email] [image: https://twitter.com/freeswitch] -------------- next part -------------- An HTML attachment was scrubbed... URL: From abdirahman.osm at gmail.com Tue Jun 8 14:11:26 2021 From: abdirahman.osm at gmail.com (Abdirahman Osman) Date: Tue, 8 Jun 2021 10:11:26 -0400 Subject: [Freeswitch-users] Freeswitch status stuck at 600 sessions with no traffic Message-ID: 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 at 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: From botelist at gmail.com Tue Jun 8 16:54:22 2021 From: botelist at gmail.com (Bote Man) Date: Tue, 8 Jun 2021 12:54:22 -0400 Subject: [Freeswitch-users] Freeswitch status stuck at 600 sessions with no traffic In-Reply-To: References: Message-ID: <010801d75c86$eeec5e00$ccc51a00$@gmail.com> Is there a script called from the dialplan that might be interfering with proper call completion? Anything else that might interfere with SIP signaling, particularly the BYE message? Bote Man http://www.botecomm.com/bote/radio/streaming.html From: Abdirahman Osman Sent: Tuesday, 8 June, 2021 10:11 To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Freeswitch status stuck at 600 sessions with no traffic Hello, My maximum session is set to 600 in the switch.conf.xml, but freeswitch status is stuck at the moment 600 concurrent calls although I don't have a single call on the freeswitch. freeswitch at kvs> status UP 0 years, 2 days, 7 hours, 25 minutes, 30 seconds, 810 milliseconds, 872 microseconds FreeSWITCH (Version 1.10.5 -release-17-25569c1631 64bit) is ready 747 session(s) since startup 600 session(s) - peak 600, last 5min 600 0 session(s) per Sec out of max 30, peak 2, last 5min 1 600 session(s) max min idle cpu 0.00/78.60 Current Stack Size/Max 240K/8192K This is the error I am getting on Freeswitch [DESTINATION_OUT_OF_ORDER] 2021-06-04 19:05:44.404930 [CRIT] switch_core_session.c:2363 Over Session Limit! 600 2021-06-04 19:05:44.404930 [CRIT] mod_sofia.c:4730 Error Creating Session 2021-06-04 19:05:44.404930 [NOTICE] switch_ivr_originate.c:2999 Cannot create outgoing channel of type [sofia] cause: [DESTINATION_OUT_OF_ORDER] 2021-06-04 19:05:44.404930 [DEBUG] switch_ivr_originate.c:3995 Originate Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER] 3:07 Thanks! Abdirahman -------------- next part -------------- An HTML attachment was scrubbed... URL: From abdirahman.osm at gmail.com Tue Jun 8 17:13:58 2021 From: abdirahman.osm at gmail.com (Abdirahman Osman) Date: Tue, 8 Jun 2021 13:13:58 -0400 Subject: [Freeswitch-users] Freeswitch status stuck at 600 sessions with no traffic In-Reply-To: <010801d75c86$eeec5e00$ccc51a00$@gmail.com> References: <010801d75c86$eeec5e00$ccc51a00$@gmail.com> Message-ID: Thanks Bote man, I already figured out,, As you suggested it was SIP point not sending BYE message, left 600 UDP ports connected Abdirahman On Tue, 8 Jun 2021 at 13:07, Bote Man wrote: > Is there a script called from the dialplan that might be interfering with > proper call completion? Anything else that might interfere with SIP > signaling, particularly the BYE message? > > > > > > Bote Man > > http://www.botecomm.com/bote/radio/streaming.html > > > > > > > > *From:* Abdirahman Osman > *Sent:* Tuesday, 8 June, 2021 10:11 > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] Freeswitch status stuck at 600 sessions > with no traffic > > > > 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 at 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 > > > > > > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From thesipguy at gmail.com Tue Jun 8 17:43:19 2021 From: thesipguy at gmail.com (S.Rosenberg) Date: Tue, 8 Jun 2021 20:43:19 +0300 Subject: [Freeswitch-users] Amazon EC2 Instances - Freeswitch In-Reply-To: References: Message-ID: I have a similar setup, I have OpenSIPs server without a rtp engine, my Asterisk/Freeswitch servers are on public IP's but port 5060 is blocked and can only receive traffic from the OpenSIPs IP's but the RTP ports are open on the Asterisk and Freeswitch servers, this way I avoid another hop on the audio, another benefit is that I can have Asterisk/Freeswitch servers in different parts of the world and route local traffic through them without needing to travel half the world to my OpenSIPs server and back. On Sat, Jun 5, 2021, 09:47 HS wrote: > Dear all, > > I have been testing an Opensips + Freeswitch setup on the same instance on > EC2. I did follow the guide here for IP setup: > https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2 > > However, for production, it seems better to have separate instances for > Opensips and Freeswitch. (Or wrong?). What would be the following variables: > > bind_server_ip > external_rtp_ip > external_sip_ip > > Shall I use the Opensips external IP or Freeswitch (although Freeswitch > will have no external access). > > Thanks for the help. > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From itadebayo at gmail.com Wed Jun 9 07:19:37 2021 From: itadebayo at gmail.com (I. Adebayo) Date: Wed, 9 Jun 2021 08:19:37 +0100 Subject: [Freeswitch-users] Amazon EC2 Instances - Freeswitch In-Reply-To: References: Message-ID: Hi Rosenber, We have been trying to get OpeSIPS to work with freeswitch without success for some time now. Can you share a basic configuration we use ASTPP. Thanks. Ismail Rosenber, Virus-free. www.avast.com <#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2> On Tue, Jun 8, 2021 at 7:12 PM S.Rosenberg wrote: > I have a similar setup, I have OpenSIPs server without a rtp engine, my > Asterisk/Freeswitch servers are on public IP's but port 5060 is blocked and > can only receive traffic from the OpenSIPs IP's but the RTP ports are open > on the Asterisk and Freeswitch servers, this way I avoid another hop on the > audio, another benefit is that I can have Asterisk/Freeswitch servers in > different parts of the world and route local traffic through them without > needing to travel half the world to my OpenSIPs server and back. > > On Sat, Jun 5, 2021, 09:47 HS wrote: > >> Dear all, >> >> I have been testing an Opensips + Freeswitch setup on the same instance >> on EC2. I did follow the guide here for IP setup: >> https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2 >> >> However, for production, it seems better to have separate instances for >> Opensips and Freeswitch. (Or wrong?). What would be the following variables: >> >> bind_server_ip >> external_rtp_ip >> external_sip_ip >> >> Shall I use the Opensips external IP or Freeswitch (although Freeswitch >> will have no external access). >> >> Thanks for the help. >> _________________________________________________________________________ >> >> 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From Raveendra.B at vectone.com Wed Jun 9 12:04:28 2021 From: Raveendra.B at vectone.com (Raveendra Battala) Date: Wed, 9 Jun 2021 12:04:28 +0000 Subject: [Freeswitch-users] How to build Sofia-sip stack 32bit version for android Message-ID: Hi, I am trying to build sofia-sip-1.12.11 or sofia-sip-1.13.3 32bit version for android but build failed always with the following error. Could you share information me whether Sofia-sip stack supports arm 32bit version build for android? export AR=$NDK/toolchains/arm-linux-androideabi-4.9/prebuilt/darwin-x86_64/bin/arm-linux-androideabi-ar export AS=$NDK/toolchains/arm-linux-androideabi-4.9/prebuilt/darwin-x86_64/bin/arm-linux-androideabi-as export CC=$NDK/toolchains/arm-linux-androideabi-4.9/prebuilt/darwin-x86_64/bin/arm-linux-androideabi-gcc export CXX=$NDK/toolchains/arm-linux-androideabi-4.9/prebuilt/darwin-x86_64/bin/arm-linux-androideabi-g++ export LD=$NDK/toolchains/arm-linux-androideabi-4.9/prebuilt/darwin-x86_64/bin/arm-linux-androideabi-ld export RANLIB=$NDK/toolchains/arm-linux-androideabi-4.9/prebuilt/darwin-x86_64/bin/arm-linux-androideabi-ranlib export STRIP=$NDK/toolchains/arm-linux-androideabi-4.9/prebuilt/darwin-x86_64/bin/arm-linux-androideabi-strip export NM=$NDK/toolchains/arm-linux-androideabi-4.9/prebuilt/darwin-x86_64/bin/arm-linux-androideabi-nm export PATH=$NDK/toolchains/arm-linux-androideabi-4.9/prebuilt/darwin-x86_64/bin:$PATH ~/WORK/SofiaSIP/sofia-sip-1.12.11$ ./configure --host=arm-linux-androideabi --build=arm-linux-androideabi checking build system type... Invalid configuration `arm-linux-androideabi': system `androideabi' not recognized configure: error: /bin/bash ./config.sub arm-linux-androideabi failed Thanks & Regards, Raveendra NOTICE AND DISCLAIMER This email contains Vectone information, which may be privileged or confidential. It's meant only for the individual(s) or entity named above. If you're not the intended recipient, note that disclosing, copying, distributing or using this information is prohibited. If you've received this email in error, please let me know immediately on the email address above. We monitor our email system, and may record your emails. Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: From johan.pascal at linphone.org Tue Jun 8 22:04:53 2021 From: johan.pascal at linphone.org (Johan Pascal) Date: Wed, 9 Jun 2021 00:04:53 +0200 Subject: [Freeswitch-users] DTLS and media description proto In-Reply-To: <5422ff67-fb60-b77d-631c-3ed9211ad5ab@linphone.org> References: <5422ff67-fb60-b77d-631c-3ed9211ad5ab@linphone.org> Message-ID: <286707ab-6a7b-0c89-c41f-5049c88ede31@linphone.org> An HTML attachment was scrubbed... URL: From igor.potjevlesch at gmail.com Fri Jun 11 09:11:51 2021 From: igor.potjevlesch at gmail.com (Igor Potjevlesh) Date: Fri, 11 Jun 2021 11:11:51 +0200 Subject: [Freeswitch-users] Call drop with 406 after being answered Message-ID: <001a01d75ea1$d1a3af70$74eb0e50$@gmail.com> Hello! I'm facing a problem with a call scenario with an in-dialog INVITE just after the call has been picked up. Freeswitch receive on his leg B this in-dialog INVITE. Not forwarded to leg A. In a SIP trace, I can see 406 Not Acceptable on leg B whereas there is no difference in the Media except "Bandwidth Information" which is "AS:80" versus "AS:82" in the RE-INVITE. I turn on the logs on debug, but nothing specific appears. The first thing: 3f6d8dd1-5415-4a58-a948-551d841cf5dd 2021-06-11 10:34:12.873542 [NOTICE] sofia.c:1012 Hangup sofia/internal/ABC [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 3f6d8dd1-5415-4a58-a948-551d841cf5dd 2021-06-11 10:34:12.873542 [DEBUG] switch_ivr_bridge.c:787 BRIDGE THREAD DONE [sofia/internal/ABC] 3f6d8dd1-5415-4a58-a948-551d841cf5dd 2021-06-11 10:34:12.873542 [DEBUG] switch_core_state_machine.c:653 (sofia/internal/ ABC) State EXCHANGE_MEDIA going to sleep [.] 5f1866f7-f6c6-43c9-83bb-07cbb51cc3a5 2021-06-11 10:34:12.893545 [DEBUG] switch_core_session.c:2815 sofia/lega/+FROM at W.X.Y.Z. skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) Is there any reason for this? SDP is not duplicated, but there is no specific reason for replying with 406. Regards, Igor. -- L'absence de virus dans ce courrier électronique a été vérifiée par le logiciel antivirus Avast. https://www.avast.com/antivirus -------------- next part -------------- An HTML attachment was scrubbed... URL: From bullehs at gmail.com Fri Jun 11 09:16:24 2021 From: bullehs at gmail.com (HS) Date: Fri, 11 Jun 2021 14:16:24 +0500 Subject: [Freeswitch-users] Amazon EC2 Instances - Freeswitch In-Reply-To: References: Message-ID: Pray tell Mr. Rosenberg :) -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Fri Jun 11 10:45:03 2021 From: david.villasmil.work at gmail.com (David Villasmil) Date: Fri, 11 Jun 2021 11:45:03 +0100 Subject: [Freeswitch-users] Call drop with 406 after being answered In-Reply-To: <001a01d75ea1$d1a3af70$74eb0e50$@gmail.com> References: <001a01d75ea1$d1a3af70$74eb0e50$@gmail.com> Message-ID: Try getting a trace for the call. On Fri, 11 Jun 2021 at 10:12, Igor Potjevlesh wrote: > Hello! > > > > I'm facing a problem with a call scenario with an in-dialog INVITE just > after the call has been picked up. > > Freeswitch receive on his leg B this in-dialog INVITE. Not forwarded to > leg A. > > In a SIP trace, I can see 406 Not Acceptable on leg B whereas there is no > difference in the Media except "Bandwidth Information" which is "AS:80" > versus "AS:82" in the RE-INVITE. > > I turn on the logs on debug, but nothing specific appears. The first thing: > > 3f6d8dd1-5415-4a58-a948-551d841cf5dd 2021-06-11 10:34:12.873542 [NOTICE] > sofia.c:1012 Hangup sofia/internal/ABC [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] > > 3f6d8dd1-5415-4a58-a948-551d841cf5dd 2021-06-11 10:34:12.873542 [DEBUG] > switch_ivr_bridge.c:787 BRIDGE THREAD DONE [sofia/internal/ABC] > > 3f6d8dd1-5415-4a58-a948-551d841cf5dd 2021-06-11 10:34:12.873542 [DEBUG] > switch_core_state_machine.c:653 (sofia/internal/ ABC) State EXCHANGE_MEDIA > going to sleep > > […] > > 5f1866f7-f6c6-43c9-83bb-07cbb51cc3a5 2021-06-11 10:34:12.893545 [DEBUG] > switch_core_session.c:2815 sofia/lega/+FROM at W.X.Y.Z. skip receive message > [APPLICATION_EXEC_COMPLETE] (channel is hungup already) > > > > Is there any reason for this? SDP is not duplicated, but there is no > specific reason for replying with 406. > > > > Regards, > > > > Igor. > > > ------------------------------ > [image: Avast logo] > > L'absence de virus dans ce courrier électronique a été vérifiée par le > logiciel antivirus Avast. > www.avast.com > > <#m_-3255753254414001689_DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2> > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From igor.potjevlesch at gmail.com Fri Jun 11 16:42:47 2021 From: igor.potjevlesch at gmail.com (Igor Potjevlesh) Date: Fri, 11 Jun 2021 18:42:47 +0200 Subject: [Freeswitch-users] Call drop with 406 after being answered In-Reply-To: References: <001a01d75ea1$d1a3af70$74eb0e50$@gmail.com> Message-ID: <004301d75ee0$cfdc1030$6f943090$@gmail.com> Hi David, I got a SIP trace. Do you think of another kind of trace? Something in addition of log level debug? Regards, Igor. De : FreeSWITCH-users De la part de David Villasmil Envoyé : vendredi 11 juin 2021 12:45 À : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] Call drop with 406 after being answered Try getting a trace for the call. On Fri, 11 Jun 2021 at 10:12, Igor Potjevlesh > wrote: Hello! I'm facing a problem with a call scenario with an in-dialog INVITE just after the call has been picked up. Freeswitch receive on his leg B this in-dialog INVITE. Not forwarded to leg A. In a SIP trace, I can see 406 Not Acceptable on leg B whereas there is no difference in the Media except "Bandwidth Information" which is "AS:80" versus "AS:82" in the RE-INVITE. I turn on the logs on debug, but nothing specific appears. The first thing: 3f6d8dd1-5415-4a58-a948-551d841cf5dd 2021-06-11 10:34:12.873542 [NOTICE] sofia.c:1012 Hangup sofia/internal/ABC [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 3f6d8dd1-5415-4a58-a948-551d841cf5dd 2021-06-11 10:34:12.873542 [DEBUG] switch_ivr_bridge.c:787 BRIDGE THREAD DONE [sofia/internal/ABC] 3f6d8dd1-5415-4a58-a948-551d841cf5dd 2021-06-11 10:34:12.873542 [DEBUG] switch_core_state_machine.c:653 (sofia/internal/ ABC) State EXCHANGE_MEDIA going to sleep […] 5f1866f7-f6c6-43c9-83bb-07cbb51cc3a5 2021-06-11 10:34:12.893545 [DEBUG] switch_core_session.c:2815 sofia/lega/+FROM at W.X.Y.Z . skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) Is there any reason for this? SDP is not duplicated, but there is no specific reason for replying with 406. Regards, Igor. _____ L'absence de virus dans ce courrier électronique a été vérifiée par le logiciel antivirus Avast. www.avast.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 at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com -- Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -- L'absence de virus dans ce courrier électronique a été vérifiée par le logiciel antivirus Avast. https://www.avast.com/antivirus -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Fri Jun 11 21:30:05 2021 From: david.villasmil.work at gmail.com (David Villasmil) Date: Fri, 11 Jun 2021 22:30:05 +0100 Subject: [Freeswitch-users] Call drop with 406 after being answered In-Reply-To: <004301d75ee0$cfdc1030$6f943090$@gmail.com> References: <001a01d75ea1$d1a3af70$74eb0e50$@gmail.com> <004301d75ee0$cfdc1030$6f943090$@gmail.com> Message-ID: I hadn't registered this: Freeswitch receive on his leg B this in-dialog INVITE. is the codec coming from that re-invite acceptable by fs? " 406 Not Acceptable" usually means FS doesn't support the offered codec. Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 On Fri, Jun 11, 2021 at 5:43 PM Igor Potjevlesh wrote: > Hi David, > > > > I got a SIP trace. Do you think of another kind of trace? Something in > addition of log level debug? > > > > Regards, > > > > Igor. > > > > *De :* FreeSWITCH-users *De > la part de* David Villasmil > *Envoyé :* vendredi 11 juin 2021 12:45 > *À :* FreeSWITCH Users Help > *Objet :* Re: [Freeswitch-users] Call drop with 406 after being answered > > > > Try getting a trace for the call. > > > > On Fri, 11 Jun 2021 at 10:12, Igor Potjevlesh > wrote: > > Hello! > > > > I'm facing a problem with a call scenario with an in-dialog INVITE just > after the call has been picked up. > > Freeswitch receive on his leg B this in-dialog INVITE. Not forwarded to > leg A. > > In a SIP trace, I can see 406 Not Acceptable on leg B whereas there is no > difference in the Media except "Bandwidth Information" which is "AS:80" > versus "AS:82" in the RE-INVITE. > > I turn on the logs on debug, but nothing specific appears. The first thing: > > 3f6d8dd1-5415-4a58-a948-551d841cf5dd 2021-06-11 10:34:12.873542 [NOTICE] > sofia.c:1012 Hangup sofia/internal/ABC [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] > > 3f6d8dd1-5415-4a58-a948-551d841cf5dd 2021-06-11 10:34:12.873542 [DEBUG] > switch_ivr_bridge.c:787 BRIDGE THREAD DONE [sofia/internal/ABC] > > 3f6d8dd1-5415-4a58-a948-551d841cf5dd 2021-06-11 10:34:12.873542 [DEBUG] > switch_core_state_machine.c:653 (sofia/internal/ ABC) State EXCHANGE_MEDIA > going to sleep > > […] > > 5f1866f7-f6c6-43c9-83bb-07cbb51cc3a5 2021-06-11 10:34:12.893545 [DEBUG] > switch_core_session.c:2815 sofia/lega/+FROM at W.X.Y.Z. skip receive message > [APPLICATION_EXEC_COMPLETE] (channel is hungup already) > > > > Is there any reason for this? SDP is not duplicated, but there is no > specific reason for replying with 406. > > > > Regards, > > > > Igor. > > > ------------------------------ > > [image: Avast logo] > > L'absence de virus dans ce courrier électronique a été vérifiée par le > logiciel antivirus Avast. > www.avast.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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com > > -- > > Regards, > > > > David Villasmil > > email: david.villasmil.work at gmail.com > > phone: +34669448337 > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From thesipguy at gmail.com Sun Jun 13 15:23:19 2021 From: thesipguy at gmail.com (S.Rosenberg) Date: Sun, 13 Jun 2021 18:23:19 +0300 Subject: [Freeswitch-users] Amazon EC2 Instances - Freeswitch In-Reply-To: References: Message-ID: It depends a lot on your design, you need to know what you want OpenSIPs to do and what do you want Freeswitch to do, here is one tutorial by my friend Giovanni Maruzzelli, he wrote the Freeswitch book and is very active on this forum. https://www.opensips.org/Documentation/Tutorials-OpenSIPSFreeSwitchIntegration On Wed, Jun 9, 2021 at 10:44 AM I. Adebayo wrote: > Hi Rosenber, > > We have been trying to get OpeSIPS to work with freeswitch without success > for some time now. > > Can you share a basic configuration we use ASTPP. > > Thanks. > > Ismail > Rosenber, > > > > > Virus-free. > www.avast.com > > <#m_2870777761763276444_DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2> > > On Tue, Jun 8, 2021 at 7:12 PM S.Rosenberg wrote: > >> I have a similar setup, I have OpenSIPs server without a rtp engine, my >> Asterisk/Freeswitch servers are on public IP's but port 5060 is blocked and >> can only receive traffic from the OpenSIPs IP's but the RTP ports are open >> on the Asterisk and Freeswitch servers, this way I avoid another hop on the >> audio, another benefit is that I can have Asterisk/Freeswitch servers in >> different parts of the world and route local traffic through them without >> needing to travel half the world to my OpenSIPs server and back. >> >> On Sat, Jun 5, 2021, 09:47 HS wrote: >> >>> Dear all, >>> >>> I have been testing an Opensips + Freeswitch setup on the same instance >>> on EC2. I did follow the guide here for IP setup: >>> https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2 >>> >>> However, for production, it seems better to have separate instances for >>> Opensips and Freeswitch. (Or wrong?). What would be the following variables: >>> >>> bind_server_ip >>> external_rtp_ip >>> external_sip_ip >>> >>> Shall I use the Opensips external IP or Freeswitch (although Freeswitch >>> will have no external access). >>> >>> Thanks for the help. >>> _________________________________________________________________________ >>> >>> 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 at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From itadebayo at gmail.com Sun Jun 13 20:52:49 2021 From: itadebayo at gmail.com (I. Adebayo) Date: Sun, 13 Jun 2021 21:52:49 +0100 Subject: [Freeswitch-users] Amazon EC2 Instances - Freeswitch In-Reply-To: References: Message-ID: Thanks. We go through the tutorial again. Thanks. Ismail On Sun, Jun 13, 2021 at 4:41 PM S.Rosenberg wrote: > It depends a lot on your design, you need to know what you want OpenSIPs > to do and what do you want Freeswitch to do, here is one tutorial by my > friend Giovanni Maruzzelli, he wrote the Freeswitch book and is very > active on this forum. > > > https://www.opensips.org/Documentation/Tutorials-OpenSIPSFreeSwitchIntegration > > On Wed, Jun 9, 2021 at 10:44 AM I. Adebayo wrote: > >> Hi Rosenber, >> >> We have been trying to get OpeSIPS to work with freeswitch without >> success for some time now. >> >> Can you share a basic configuration we use ASTPP. >> >> Thanks. >> >> Ismail >> Rosenber, >> >> >> >> >> Virus-free. >> www.avast.com >> >> <#m_4692813517024008278_m_2870777761763276444_DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2> >> >> On Tue, Jun 8, 2021 at 7:12 PM S.Rosenberg wrote: >> >>> I have a similar setup, I have OpenSIPs server without a rtp engine, my >>> Asterisk/Freeswitch servers are on public IP's but port 5060 is blocked and >>> can only receive traffic from the OpenSIPs IP's but the RTP ports are open >>> on the Asterisk and Freeswitch servers, this way I avoid another hop on the >>> audio, another benefit is that I can have Asterisk/Freeswitch servers in >>> different parts of the world and route local traffic through them without >>> needing to travel half the world to my OpenSIPs server and back. >>> >>> On Sat, Jun 5, 2021, 09:47 HS wrote: >>> >>>> Dear all, >>>> >>>> I have been testing an Opensips + Freeswitch setup on the same instance >>>> on EC2. I did follow the guide here for IP setup: >>>> https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2 >>>> >>>> However, for production, it seems better to have separate instances for >>>> Opensips and Freeswitch. (Or wrong?). What would be the following variables: >>>> >>>> bind_server_ip >>>> external_rtp_ip >>>> external_sip_ip >>>> >>>> Shall I use the Opensips external IP or Freeswitch (although Freeswitch >>>> will have no external access). >>>> >>>> Thanks for the help. >>>> >>>> _________________________________________________________________________ >>>> >>>> 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From bullehs at gmail.com Mon Jun 14 12:43:36 2021 From: bullehs at gmail.com (HS) Date: Mon, 14 Jun 2021 17:43:36 +0500 Subject: [Freeswitch-users] Amazon EC2 Instances - Freeswitch In-Reply-To: References: Message-ID: My turn to help this time. So I followed the following tutorial on one instance of Amazon EC2 and Opensips 3.0 (+RTPProxy): https://www.opensips.org/Documentation/Tutorials-OpenSIPSFreeSwitchIntegration There were some changes in the .cfg file - which I did over come (or so I think). The second was an error that showed up in the xml_handler.lua file. Calls get to an IVR, however, I did have the following issues (and remain unsolved): 1. IVR/calls to Freeswitch disconnect after 30-32s. 2. Unsure if users are being pulled from the database or the profiles. 3. DTMF input isn't working (system doesn't respond to input.) Hope that helps and maybe someone has a solution. > -------------- next part -------------- An HTML attachment was scrubbed... URL: From julien.terrasson at gmail.com Mon Jun 14 14:10:04 2021 From: julien.terrasson at gmail.com (Julien Terrasson) Date: Mon, 14 Jun 2021 16:10:04 +0200 Subject: [Freeswitch-users] channel variable for time left when using sched_hangup ? Message-ID: Hello, Does anybody know if there is a way to capture the time remaining before a channel is hanged_up with sched_hangup ? Is there somewhere a channel variable that can reflect this information ? Regards, Julien -------------- next part -------------- An HTML attachment was scrubbed... URL: From wayne_w_chang at yahoo.com Mon Jun 14 05:41:45 2021 From: wayne_w_chang at yahoo.com (Wayne Chang) Date: Mon, 14 Jun 2021 05:41:45 +0000 (UTC) Subject: UDP port Operation not permitted References: <527708163.4877841.1623649305719.ref@mail.yahoo.com> Message-ID: <527708163.4877841.1623649305719@mail.yahoo.com> Hi, I am using the fs_cli command to originate a call to a media server. However I am receiving the 503 Service Unavailable. I have checked the media server listening on port 5060. I am able to use sip client to send the invite. Can you help me understand why when using the originate command it failed? Thanks. Here is the log. 2021-06-13 22:08:10.682603 [DEBUG] switch_core_state_machine.c:585 (sofia/external/PaymentIVR2-34117) Running State Change CS_CONSUME_MEDIA (Cur 1 Tot 29)soa_static.c:1028 soa_sdp_mode_set() soa_sdp_mode_set(0x7f1347ffc5c0, (nil), ""): calledsoa_static.c:1445 offer_answer_step() soa_static(0x7f133c0412b0, soa_generate_offer): storing local descriptionsoa.c:1268 soa_get_local_sdp() soa_get_local_sdp(static::0x7f133c0412b0, [(nil)], [0x7f1347ffe720], [0x7f1347ffe718]) called2021-06-13 22:08:10.682603 [DEBUG] switch_core_state_machine.c:663 (sofia/external/PaymentIVR2-34117) State CONSUME_MEDIAnta.c:2694 nta_tpn_by_url() nta: selecting scheme sip2021-06-13 22:08:10.682603 [DEBUG] switch_core_state_machine.c:663 (sofia/external/PaymentIVR2-34117) State CONSUME_MEDIA going to sleeptport.c:3285 tport_tsend() tport_tsend(0x7f133c004e80) tpn = UDP/10.201.21.140:5060tport.c:4071 tport_resolve() tport_resolve addrinfo = 10.201.21.140:5060tport.c:4708 tport_by_addrinfo() tport_by_addrinfo(0x7f133c004e80): not found by name UDP/10.201.21.140:5060tport.c:3663 tport_send_fatal() tport_vsend(0x7f133c004e80): Operation not permitted with (s=35 UDP/10.201.21.140:5060)tport.c:3521 tport_send_msg() tport_vsend returned -1nta.c:8542 outgoing_print_tport_error() nta: INVITE (37269853): Operation not permitted (1) with UDP/[10.201.21.140]:5060nua_session.c:4135 signal_call_state_change() nua(0x7f136c038fd0): call state changed: init -> calling, sent offersoa.c:1268 soa_get_local_sdp() soa_get_local_sdp(static::0x7f133c0412b0, [0x7f1347ffe730], [0x7f1347ffe738], [(nil)]) callednua_stack.c:269 nua_stack_event() nua(0x7f136c038fd0): event i_state INVITE sentnua_stack.c:359 nua_application_event() nua: nua_application_event: enteringnua_stack.c:271 nua_stack_event() nua(0x7f136c038fd0): event r_invite 503 Service Unavailablenua_session.c:4135 signal_call_state_change() nua(0x7f136c038fd0): call state changed: calling -> initnua_stack.c:271 nua_stack_event() nua(0x7f136c038fd0): event i_state 503 Service Unavailablenua_stack.c:271 nua_stack_event() nua(0x7f136c038fd0): event i_terminated 503 Service Unavailable Wayne -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at freeswitch.org Mon Jun 14 17:44:24 2021 From: mike at freeswitch.org (Mike Jerris) Date: Mon, 14 Jun 2021 11:44:24 -0600 Subject: [Freeswitch-users] channel variable for time left when using sched_hangup ? In-Reply-To: References: Message-ID: <9B3A539D-BAFB-4E3F-A7A7-681CB4FE7723@freeswitch.org> not directly no. you’d need to at the time you set the sched hangup also set a var with the epoch time of the hangup so you could easily calculate it later. > On Jun 14, 2021, at 8:10 AM, Julien Terrasson wrote: > > Hello, > > Does anybody know if there is a way to capture the time remaining before a channel is hanged_up with sched_hangup ? > Is there somewhere a channel variable that can reflect this information ? > > Regards, > > Julien From mike at freeswitch.org Mon Jun 14 17:45:48 2021 From: mike at freeswitch.org (Mike Jerris) Date: Mon, 14 Jun 2021 11:45:48 -0600 Subject: [Freeswitch-users] UDP port Operation not permitted In-Reply-To: References: <527708163.4877841.1623649305719.ref@mail.yahoo.com> Message-ID: <89DDE2FD-024E-4562-85EA-73EF5A6B9D27@freeswitch.org> tport.c:3663 tport_send_fatal() tport_vsend(0x7f133c004e80): Operation not permitted with (s=35 UDP/10.201.21.140:5060) write to network failing. > On Jun 14, 2021, at 11:42 AM, Wayne Chang via FreeSWITCH-users wrote: > > > From: Wayne Chang > Subject: UDP port Operation not permitted > Date: June 13, 2021 at 11:41:45 PM MDT > To: "freeswitch-users at lists.freeswitch.org" > > > Hi, > > I am using the fs_cli command to originate a call to a media server. However I am receiving the 503 Service Unavailable. I have checked the media server listening on port 5060. I am able to use sip client to send the invite. Can you help me understand why when using the originate command it failed? > > Thanks. Here is the log. > > 2021-06-13 22:08:10.682603 [DEBUG] switch_core_state_machine.c:585 (sofia/external/PaymentIVR2-34117) Running State Change CS_CONSUME_MEDIA (Cur 1 Tot 29) > soa_static.c:1028 soa_sdp_mode_set() soa_sdp_mode_set(0x7f1347ffc5c0, (nil), ""): called > soa_static.c:1445 offer_answer_step() soa_static(0x7f133c0412b0, soa_generate_offer): storing local description > soa.c:1268 soa_get_local_sdp() soa_get_local_sdp(static::0x7f133c0412b0, [(nil)], [0x7f1347ffe720], [0x7f1347ffe718]) called > 2021-06-13 22:08:10.682603 [DEBUG] switch_core_state_machine.c:663 (sofia/external/PaymentIVR2-34117) State CONSUME_MEDIA > nta.c:2694 nta_tpn_by_url() nta: selecting scheme sip > 2021-06-13 22:08:10.682603 [DEBUG] switch_core_state_machine.c:663 (sofia/external/PaymentIVR2-34117) State CONSUME_MEDIA going to sleep > tport.c:3285 tport_tsend() tport_tsend(0x7f133c004e80) tpn = UDP/10.201.21.140:5060 > tport.c:4071 tport_resolve() tport_resolve addrinfo = 10.201.21.140:5060 > tport.c:4708 tport_by_addrinfo() tport_by_addrinfo(0x7f133c004e80): not found by name UDP/10.201.21.140:5060 > tport.c:3663 tport_send_fatal() tport_vsend(0x7f133c004e80): Operation not permitted with (s=35 UDP/10.201.21.140:5060) > tport.c:3521 tport_send_msg() tport_vsend returned -1 > nta.c:8542 outgoing_print_tport_error() nta: INVITE (37269853): Operation not permitted (1) with UDP/[10.201.21.140]:5060 > nua_session.c:4135 signal_call_state_change() nua(0x7f136c038fd0): call state changed: init -> calling, sent offer > soa.c:1268 soa_get_local_sdp() soa_get_local_sdp(static::0x7f133c0412b0, [0x7f1347ffe730], [0x7f1347ffe738], [(nil)]) called > nua_stack.c:269 nua_stack_event() nua(0x7f136c038fd0): event i_state INVITE sent > nua_stack.c:359 nua_application_event() nua: nua_application_event: entering > nua_stack.c:271 nua_stack_event() nua(0x7f136c038fd0): event r_invite 503 Service Unavailable > nua_session.c:4135 signal_call_state_change() nua(0x7f136c038fd0): call state changed: calling -> init > nua_stack.c:271 nua_stack_event() nua(0x7f136c038fd0): event i_state 503 Service Unavailable > nua_stack.c:271 nua_stack_event() nua(0x7f136c038fd0): event i_terminated 503 Service Unavailable > -------------- next part -------------- An HTML attachment was scrubbed... URL: From wayne_w_chang at yahoo.com Thu Jun 17 07:42:11 2021 From: wayne_w_chang at yahoo.com (Wayne Chang) Date: Thu, 17 Jun 2021 07:42:11 +0000 (UTC) Subject: [Freeswitch-users] UDP port Operation not permitted In-Reply-To: <89DDE2FD-024E-4562-85EA-73EF5A6B9D27@freeswitch.org> References: <527708163.4877841.1623649305719.ref@mail.yahoo.com> <89DDE2FD-024E-4562-85EA-73EF5A6B9D27@freeswitch.org> Message-ID: <1953716406.789130.1623915731801@mail.yahoo.com> Hi Mike, what could be the issue?  I am sure 10.201.21.140 has port 5060 listening. I tested with SIPp and the target 10.201.21.140 could receive INVITE properly. The issue only happened when running following command from fs_cli. freeswitch at freeswitch> originatesofia/external/PaymentIVR2-34117 at 10.201.21.140 1002 Thanks Wayne On Monday, June 14, 2021, 10:45:50 AM PDT, Mike Jerris wrote: tport.c:3663 tport_send_fatal() tport_vsend(0x7f133c004e80): Operation not permitted with (s=35 UDP/10.201.21.140:5060) write to network failing. On Jun 14, 2021, at 11:42 AM, Wayne Chang via FreeSWITCH-users wrote: From: Wayne Chang Subject: UDP port Operation not permitted Date: June 13, 2021 at 11:41:45 PM MDT To: "freeswitch-users at lists.freeswitch.org" Hi, I am using the fs_cli command to originate a call to a media server. However I am receiving the 503 Service Unavailable. I have checked the media server listening on port 5060. I am able to use sip client to send the invite. Can you help me understand why when using the originate command it failed? Thanks. Here is the log. 2021-06-13 22:08:10.682603 [DEBUG] switch_core_state_machine.c:585 (sofia/external/PaymentIVR2-34117) Running State Change CS_CONSUME_MEDIA (Cur 1 Tot 29)soa_static.c:1028 soa_sdp_mode_set() soa_sdp_mode_set(0x7f1347ffc5c0, (nil), ""): calledsoa_static.c:1445 offer_answer_step() soa_static(0x7f133c0412b0, soa_generate_offer): storing local descriptionsoa.c:1268 soa_get_local_sdp() soa_get_local_sdp(static::0x7f133c0412b0, [(nil)], [0x7f1347ffe720], [0x7f1347ffe718]) called2021-06-13 22:08:10.682603 [DEBUG] switch_core_state_machine.c:663 (sofia/external/PaymentIVR2-34117) State CONSUME_MEDIAnta.c:2694 nta_tpn_by_url() nta: selecting scheme sip2021-06-13 22:08:10.682603 [DEBUG] switch_core_state_machine.c:663 (sofia/external/PaymentIVR2-34117) State CONSUME_MEDIA going to sleeptport.c:3285 tport_tsend() tport_tsend(0x7f133c004e80) tpn = UDP/10.201.21.140:5060tport.c:4071 tport_resolve() tport_resolve addrinfo = 10.201.21.140:5060tport.c:4708 tport_by_addrinfo() tport_by_addrinfo(0x7f133c004e80): not found by name UDP/10.201.21.140:5060tport.c:3663 tport_send_fatal() tport_vsend(0x7f133c004e80): Operation not permitted with (s=35 UDP/10.201.21.140:5060)tport.c:3521 tport_send_msg() tport_vsend returned -1nta.c:8542 outgoing_print_tport_error() nta: INVITE (37269853): Operation not permitted (1) with UDP/[10.201.21.140]:5060nua_session.c:4135 signal_call_state_change() nua(0x7f136c038fd0): call state changed: init -> calling, sent offersoa.c:1268 soa_get_local_sdp() soa_get_local_sdp(static::0x7f133c0412b0, [0x7f1347ffe730], [0x7f1347ffe738], [(nil)]) callednua_stack.c:269 nua_stack_event() nua(0x7f136c038fd0): event i_state INVITE sentnua_stack.c:359 nua_application_event() nua: nua_application_event: enteringnua_stack.c:271 nua_stack_event() nua(0x7f136c038fd0): event r_invite 503 Service Unavailablenua_session.c:4135 signal_call_state_change() nua(0x7f136c038fd0): call state changed: calling -> initnua_stack.c:271 nua_stack_event() nua(0x7f136c038fd0): event i_state 503 Service Unavailablenua_stack.c:271 nua_stack_event() nua(0x7f136c038fd0): event i_terminated 503 Service Unavailable -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Thu Jun 17 07:53:09 2021 From: david.villasmil.work at gmail.com (David Villasmil) Date: Thu, 17 Jun 2021 08:53:09 +0100 Subject: [Freeswitch-users] UDP port Operation not permitted In-Reply-To: References: <527708163.4877841.1623649305719.ref@mail.yahoo.com> <89DDE2FD-024E-4562-85EA-73EF5A6B9D27@freeswitch.org> Message-ID: This usually happens because either the IP address in the profile is wrong, or maybe fs is running with the wrong user. Verify the IP address both in the profile and on vars, and try running fs from the cli as root and see if anything changes. On Thu, 17 Jun 2021 at 08:42, Wayne Chang via FreeSWITCH-users < freeswitch-users at lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: Wayne Chang > To: FreeSWITCH Users Help , Mike > Jerris > Cc: > Bcc: > Date: Thu, 17 Jun 2021 07:42:11 +0000 (UTC) > Subject: Re: [Freeswitch-users] UDP port Operation not permitted > Hi Mike, > > what could be the issue? I am sure 10.201.21.140 has port 5060 listening. > I tested with SIPp and the target 10.201.21.140 could receive INVITE > properly. > > The issue only happened when running following command from fs_cli. > > freeswitch at freeswitch> originate sofia/external/ > PaymentIVR2-34117 at 10.201.21.140 1002 > Thanks > > Wayne > > > On Monday, June 14, 2021, 10:45:50 AM PDT, Mike Jerris < > mike at freeswitch.org> wrote: > > > tport.c:3663 tport_send_fatal() tport_vsend(0x7f133c004e80): Operation not > permitted with (s=35 UDP/10.201.21.140:5060) > > write to network failing. > > On Jun 14, 2021, at 11:42 AM, Wayne Chang via FreeSWITCH-users < > freeswitch-users at lists.freeswitch.org> wrote: > > > *From: *Wayne Chang > *Subject: **UDP port Operation not permitted* > *Date: *June 13, 2021 at 11:41:45 PM MDT > *To: *"freeswitch-users at lists.freeswitch.org" < > freeswitch-users at lists.freeswitch.org> > > > Hi, > > I am using the fs_cli command to originate a call to a media server. > However I am receiving the 503 Service Unavailable. I have checked the > media server listening on port 5060. I am able to use sip client to send > the invite. Can you help me understand why when using the originate command > it failed? > > Thanks. Here is the log. > > 2021-06-13 22:08:10.682603 [DEBUG] switch_core_state_machine.c:585 > (sofia/external/PaymentIVR2-34117) Running State Change CS_CONSUME_MEDIA > (Cur 1 Tot 29) > soa_static.c:1028 soa_sdp_mode_set() soa_sdp_mode_set(0x7f1347ffc5c0, > (nil), ""): called > soa_static.c:1445 offer_answer_step() soa_static(0x7f133c0412b0, > soa_generate_offer): storing local description > soa.c:1268 soa_get_local_sdp() soa_get_local_sdp(static::0x7f133c0412b0, > [(nil)], [0x7f1347ffe720], [0x7f1347ffe718]) called > 2021-06-13 22:08:10.682603 [DEBUG] switch_core_state_machine.c:663 > (sofia/external/PaymentIVR2-34117) State CONSUME_MEDIA > nta.c:2694 nta_tpn_by_url() nta: selecting scheme sip > 2021-06-13 22:08:10.682603 [DEBUG] switch_core_state_machine.c:663 > (sofia/external/PaymentIVR2-34117) State CONSUME_MEDIA going to sleep > tport.c:3285 tport_tsend() tport_tsend(0x7f133c004e80) tpn = UDP/ > 10.201.21.140:5060 > tport.c:4071 tport_resolve() tport_resolve addrinfo = 10.201.21.140:5060 > tport.c:4708 tport_by_addrinfo() tport_by_addrinfo(0x7f133c004e80): not > found by name UDP/10.201.21.140:5060 > tport.c:3663 tport_send_fatal() tport_vsend(0x7f133c004e80): Operation not > permitted with (s=35 UDP/10.201.21.140:5060) > tport.c:3521 tport_send_msg() tport_vsend returned -1 > nta.c:8542 outgoing_print_tport_error() nta: INVITE (37269853): Operation > not permitted (1) with UDP/[10.201.21.140]:5060 > nua_session.c:4135 signal_call_state_change() nua(0x7f136c038fd0): call > state changed: init -> calling, sent offer > soa.c:1268 soa_get_local_sdp() soa_get_local_sdp(static::0x7f133c0412b0, > [0x7f1347ffe730], [0x7f1347ffe738], [(nil)]) called > nua_stack.c:269 nua_stack_event() nua(0x7f136c038fd0): event i_state > INVITE sent > nua_stack.c:359 nua_application_event() nua: nua_application_event: > entering > nua_stack.c:271 nua_stack_event() nua(0x7f136c038fd0): event r_invite 503 > Service Unavailable > nua_session.c:4135 signal_call_state_change() nua(0x7f136c038fd0): call > state changed: calling -> init > nua_stack.c:271 nua_stack_event() nua(0x7f136c038fd0): event i_state 503 > Service Unavailable > nua_stack.c:271 nua_stack_event() nua(0x7f136c038fd0): event i_terminated > 503 Service Unavailable > > > > > > ---------- Forwarded message ---------- > From: Wayne Chang via FreeSWITCH-users < > freeswitch-users at lists.freeswitch.org> > To: FreeSWITCH Users Help , Mike > Jerris > Cc: > Bcc: > Date: Thu, 17 Jun 2021 00:42:49 -0700 (PDT) > Subject: Re: [Freeswitch-users] UDP port Operation not permitted > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From igor.potjevlesch at gmail.com Thu Jun 17 08:09:37 2021 From: igor.potjevlesch at gmail.com (Igor Potjevlesh) Date: Thu, 17 Jun 2021 10:09:37 +0200 Subject: [Freeswitch-users] Call drop with 406 after being answered In-Reply-To: References: <001a01d75ea1$d1a3af70$74eb0e50$@gmail.com> <004301d75ee0$cfdc1030$6f943090$@gmail.com> Message-ID: <000a01d76350$1e650770$5b2f1650$@gmail.com> Hi David, On the top of the list yes. I just see that g729 is added and his not supported. But there is at least 1 codec supported. Regards, Igor. De : FreeSWITCH-users De la part de David Villasmil Envoyé : vendredi 11 juin 2021 23:30 À : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] Call drop with 406 after being answered I hadn't registered this: Freeswitch receive on his leg B this in-dialog INVITE. is the codec coming from that re-invite acceptable by fs? " 406 Not Acceptable" usually means FS doesn't support the offered codec. Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 On Fri, Jun 11, 2021 at 5:43 PM Igor Potjevlesh > wrote: Hi David, I got a SIP trace. Do you think of another kind of trace? Something in addition of log level debug? Regards, Igor. De : FreeSWITCH-users > De la part de David Villasmil Envoyé : vendredi 11 juin 2021 12:45 À : FreeSWITCH Users Help > Objet : Re: [Freeswitch-users] Call drop with 406 after being answered Try getting a trace for the call. On Fri, 11 Jun 2021 at 10:12, Igor Potjevlesh > wrote: Hello! I'm facing a problem with a call scenario with an in-dialog INVITE just after the call has been picked up. Freeswitch receive on his leg B this in-dialog INVITE. Not forwarded to leg A. In a SIP trace, I can see 406 Not Acceptable on leg B whereas there is no difference in the Media except "Bandwidth Information" which is "AS:80" versus "AS:82" in the RE-INVITE. I turn on the logs on debug, but nothing specific appears. The first thing: 3f6d8dd1-5415-4a58-a948-551d841cf5dd 2021-06-11 10:34:12.873542 [NOTICE] sofia.c:1012 Hangup sofia/internal/ABC [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 3f6d8dd1-5415-4a58-a948-551d841cf5dd 2021-06-11 10:34:12.873542 [DEBUG] switch_ivr_bridge.c:787 BRIDGE THREAD DONE [sofia/internal/ABC] 3f6d8dd1-5415-4a58-a948-551d841cf5dd 2021-06-11 10:34:12.873542 [DEBUG] switch_core_state_machine.c:653 (sofia/internal/ ABC) State EXCHANGE_MEDIA going to sleep […] 5f1866f7-f6c6-43c9-83bb-07cbb51cc3a5 2021-06-11 10:34:12.893545 [DEBUG] switch_core_session.c:2815 sofia/lega/+FROM at W.X.Y.Z . skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) Is there any reason for this? SDP is not duplicated, but there is no specific reason for replying with 406. Regards, Igor. _____ L'absence de virus dans ce courrier électronique a été vérifiée par le logiciel antivirus Avast. www.avast.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 at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com -- Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 _________________________________________________________________________ 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 at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com -- L'absence de virus dans ce courrier électronique a été vérifiée par le logiciel antivirus Avast. https://www.avast.com/antivirus -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Thu Jun 17 11:06:16 2021 From: david.villasmil.work at gmail.com (David Villasmil) Date: Thu, 17 Jun 2021 12:06:16 +0100 Subject: [Freeswitch-users] Call drop with 406 after being answered In-Reply-To: <000a01d76350$1e650770$5b2f1650$@gmail.com> References: <001a01d75ea1$d1a3af70$74eb0e50$@gmail.com> <004301d75ee0$cfdc1030$6f943090$@gmail.com> <000a01d76350$1e650770$5b2f1650$@gmail.com> Message-ID: Try forcing it to your supported codec On Thu, 17 Jun 2021 at 09:10, Igor Potjevlesh wrote: > Hi David, > > > > On the top of the list yes. I just see that g729 is added and his not > supported. But there is at least 1 codec supported. > > > > Regards, > > > > Igor. > > > > *De :* FreeSWITCH-users *De > la part de* David Villasmil > *Envoyé :* vendredi 11 juin 2021 23:30 > > > *À :* FreeSWITCH Users Help > *Objet :* Re: [Freeswitch-users] Call drop with 406 after being answered > > > > I hadn't registered this: > > > > Freeswitch receive on his leg B this in-dialog INVITE. > > > > > > is the codec coming from that re-invite acceptable by fs? " 406 Not > Acceptable" usually means FS doesn't support the offered codec. > > > Regards, > > > > David Villasmil > > email: david.villasmil.work at gmail.com > > phone: +34669448337 > > > > > > On Fri, Jun 11, 2021 at 5:43 PM Igor Potjevlesh < > igor.potjevlesch at gmail.com> wrote: > > Hi David, > > > > I got a SIP trace. Do you think of another kind of trace? Something in > addition of log level debug? > > > > Regards, > > > > Igor. > > > > *De :* FreeSWITCH-users *De > la part de* David Villasmil > *Envoyé :* vendredi 11 juin 2021 12:45 > *À :* FreeSWITCH Users Help > *Objet :* Re: [Freeswitch-users] Call drop with 406 after being answered > > > > Try getting a trace for the call. > > > > On Fri, 11 Jun 2021 at 10:12, Igor Potjevlesh > wrote: > > Hello! > > > > I'm facing a problem with a call scenario with an in-dialog INVITE just > after the call has been picked up. > > Freeswitch receive on his leg B this in-dialog INVITE. Not forwarded to > leg A. > > In a SIP trace, I can see 406 Not Acceptable on leg B whereas there is no > difference in the Media except "Bandwidth Information" which is "AS:80" > versus "AS:82" in the RE-INVITE. > > I turn on the logs on debug, but nothing specific appears. The first thing: > > 3f6d8dd1-5415-4a58-a948-551d841cf5dd 2021-06-11 10:34:12.873542 [NOTICE] > sofia.c:1012 Hangup sofia/internal/ABC [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] > > 3f6d8dd1-5415-4a58-a948-551d841cf5dd 2021-06-11 10:34:12.873542 [DEBUG] > switch_ivr_bridge.c:787 BRIDGE THREAD DONE [sofia/internal/ABC] > > 3f6d8dd1-5415-4a58-a948-551d841cf5dd 2021-06-11 10:34:12.873542 [DEBUG] > switch_core_state_machine.c:653 (sofia/internal/ ABC) State EXCHANGE_MEDIA > going to sleep > > […] > > 5f1866f7-f6c6-43c9-83bb-07cbb51cc3a5 2021-06-11 10:34:12.893545 [DEBUG] > switch_core_session.c:2815 sofia/lega/+FROM at W.X.Y.Z. skip receive message > [APPLICATION_EXEC_COMPLETE] (channel is hungup already) > > > > Is there any reason for this? SDP is not duplicated, but there is no > specific reason for replying with 406. > > > > Regards, > > > > Igor. > > > ------------------------------ > > [image: Avast logo] > > L'absence de virus dans ce courrier électronique a été vérifiée par le > logiciel antivirus Avast. > www.avast.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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com > > -- > > Regards, > > > > David Villasmil > > email: david.villasmil.work at gmail.com > > phone: +34669448337 > > _________________________________________________________________________ > > 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From janne at hess.ooo Tue Jun 22 14:55:15 2021 From: janne at hess.ooo (=?utf-8?q?Janne_He=C3=9F?=) Date: Tue, 22 Jun 2021 16:55:15 +0200 Subject: [Freeswitch-users] Outgoing calls with a AVM FritzBox 7490 Message-ID: <1lvho4-00BhAL-3q@mx1.helsinki.tools> Hello everyone, I'm kind of lost with setting up FS to connect to my FritzBox 7490. The goal is to use FS with spandsp to send and receive Faxes. Now, most things work in my setup. FS can successfully register as a SIP client and I can call the configured number from my mobile phoneand the call is routed to the FS0 virtual modem. The problem is outgoing calls. Using the ATD command on the virtual modem to call any external phone does not work. Internal FritzBox-configured numbers don't work either. Using Wireshark I found that the FritzBox replies with 488 Not Acceptable Here. Looking around the internet, this seems to be related to the codec configuration. I doubt this is the problem in my case since enabling more codecs doesn't help and incoming calls with the same codec restrictions work. So I'm assuming FS sends an INVITE package that the FritzBox does not like for some reason. I played around with the From field but that just results in the FritzBox not replying at all. Does anyone know what might be going on here or is there someone with a working example? I have attached the INVITE package and the response package. 192.168.0.130 is the FritzBox, 192.168.0.133 is the FS host (with firewall disabled). Thank you in advance and best regards Janne Frame 183405: 1278 bytes on wire (10224 bits), 1278 bytes captured (10224 bits) on interface -, id 0 Ethernet II, Src: RealtekU_ff:ff:ff (52:54:00:ff:ff:ff), Dst: AVMAudio_ff:ff:ff (7c:ff:4d:ff:ff:ff) Internet Protocol Version 4, Src: 192.168.0.133, Dst: 192.128.0.130 User Datagram Protocol, Src Port: 5060, Dst Port: 5060 Session Initiation Protocol (INVITE) Request-Line: INVITE sip:0123456789 at 192.168.0.130:5060 SIP/2.0 Message Header Via: SIP/2.0/UDP 192.168.0.133;rport;branch=z9hG4bKcU2Qc6yc0cZ3p Max-Forwards: 70 From: "FSModem" ;tag=j2K0XQKgXBFmN To: Call-ID: 3deaaab4-4e08-123a-ce81-52540059d216 [Generated Call-ID: 3deaaab4-4e08-123a-ce81-52540059d216] CSeq: 37632110 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.10.6-release.12~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Authorization: Digest username="hylafaxtel", realm="fritz.box", nonce="XXXXX", algorithm=MD5, uri="sip:0123456789 at 192.168.0.130:5060", response="XXXXX" Content-Type: application/sdp Content-Disposition: session Content-Length: 225 X-FS-Support: update_display,send_info Remote-Party-ID: "FSModem" ;party=calling;screen=yes;privacy=off Message Body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): FreeSWITCH 1624344691 1624344692 IN IP4 192.168.0.133 Session Name (s): FreeSWITCH Connection Information (c): IN IP4 192.168.0.133 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 27112 RTP/AVP 102 101 Media Attribute (a): rtpmap:102 L16/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-16 Media Attribute (a): ptime:20 [Generated Call-ID: ba99ae97-4de7-123a-09b4-52540059d216] [Generated Call-ID: 751E5AC1C59CE6B0 at 192.168.0.130] [I removed most Call-IDs for brevity] Frame 183406: 837 bytes on wire (6696 bits), 837 bytes captured (6696 bits) on interface -, id 0 Ethernet II, Src: AVMAudio_ff:ff:ff (7c:ff:4d:ff:ff:ff), Dst: RealtekU_ff:ff:ff (52:54:00:ff:ff:ff) Internet Protocol Version 4, Src: 192.168.0.130, Dst: 192.168.0.133 User Datagram Protocol, Src Port: 5060, Dst Port: 5060 Session Initiation Protocol (488) Status-Line: SIP/2.0 488 Not Acceptable Here Message Header Via: SIP/2.0/UDP 192.168.0.133;rport=5060;branch=z9hG4bKcU2Qc6yc0cZ3p From: "FSModem" ;tag=j2K0XQKgXBFmN To: ;tag=CDE9429F6B0A3BC3 Call-ID: 3deaaab4-4e08-123a-ce81-52540059d216 [Generated Call-ID: 3deaaab4-4e08-123a-ce81-52540059d216] CSeq: 37632110 INVITE Warning: 399 0.0.0.0 "successful but result empty" User-Agent: FRITZ!OS Content-Type: application/sdp Content-Length: 361 Message Body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): user 12041099 12041099 IN IP4 129.143.6.130 Session Name (s): call Connection Information (c): IN IP4 192.168.0.130 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 7080 RTP/AVP 8 0 2 102 100 99 97 101 Media Attribute (a): sendrecv Media Attribute (a): rtpmap:2 G726-32/8000 Media Attribute (a): rtpmap:102 G726-32/8000 Media Attribute (a): rtpmap:100 G726-40/8000 Media Attribute (a): rtpmap:99 G726-24/8000 Media Attribute (a): rtpmap:97 iLBC/8000 Media Attribute (a): fmtp:97 mode=30 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-15 Media Attribute (a): rtcp:7081 [Generated Call-ID: 596BDCD5EDA427D7 at 192.168.0.130] [Generated Call-ID: ba99ae97-4de7-123a-09b4-52540059d216] [I removed most Call-IDs for brevity] From c.wallace777 at gmail.com Sat Jun 19 17:30:12 2021 From: c.wallace777 at gmail.com (Chris Wallace) Date: Sat, 19 Jun 2021 13:30:12 -0400 Subject: [Freeswitch-users] Copy or Extract Contact Parameters from A-leg to B-leg Message-ID: We are working on implementing STIR/SHAKEN and have run into a snag when passing calls through our FreeSwitch LCR which sits between our call feature server and our SBC's. Our call feature server is passing parameters in the Contact for attestation-info and origination-id. It is intended that our SBC's will extract this data from the Contact and use it to generate an identity header before leaving our network. Ideally this information would be passed via X-Header(s) but unfortunately this is what we have to work with at the moment. We have tried to use both the "sip_invite_contact_params" and "sip_contact_params" channel variables but neither of them seem to get any data from the Contact within the inbound INVITE. We had success testing with "sip_contact_user" and dumping that data to an X-Header but we can't seem to get FreeSwitch to recognize or dump the other parameters. When debugging an active call and dumping the channel variable data for each leg of the call we don't see any of the additional parameters that are in the Contact listed in the channel variables that are dumped for either call leg's UUID. *Contact in INVITE into FreeSwitch* Contact: ;isup-oli=00;attestation-info=1;origination-id=1000;verstat=0 *sip_contact_user Dump to X-Header:* X-ContactUser: 15558675309 *FreeSwitch Logging:* EXECUTE [depth=0] sofia/core/15558675309 at x.x.x.x:5060 export(sip_h_X-ContactUser=15558675309) 2021-06-18 15:24:10.079576 [DEBUG] switch_channel.c:1310 EXPORT (export_vars) [sip_h_X-ContactUser]=[15558675309] EXECUTE [depth=0] sofia/core/15558675309 at x.x.x.x:5060 export(sip_h_X-ContactParam=) 2021-06-18 15:24:10.079576 [DEBUG] switch_channel.c:1310 EXPORT (export_vars) [sip_h_X-ContactParam]=[UNDEF] -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Tue Jun 22 17:02:47 2021 From: david.villasmil.work at gmail.com (David Villasmil) Date: Tue, 22 Jun 2021 18:02:47 +0100 Subject: [Freeswitch-users] Copy or Extract Contact Parameters from A-leg to B-leg In-Reply-To: References: Message-ID: Have you tried just using “sip_h_[whatever]” to read from the A leg and setting it on the bridge as nolocal? On Tue, 22 Jun 2021 at 17:25, Chris Wallace wrote: > We are working on implementing STIR/SHAKEN and have run into a snag when > passing calls through our FreeSwitch LCR which sits between our call > feature server and our SBC's. Our call feature server is passing parameters > in the Contact for attestation-info and origination-id. It is intended that > our SBC's will extract this data from the Contact and use it to generate an > identity header before leaving our network. Ideally this information would > be passed via X-Header(s) but unfortunately this is what we have to work > with at the moment. > > We have tried to use both the "sip_invite_contact_params" and > "sip_contact_params" channel variables but neither of them seem to get any > data from the Contact within the inbound INVITE. We had success testing > with "sip_contact_user" and dumping that data to an X-Header but we can't > seem to get FreeSwitch to recognize or dump the other parameters. > > When debugging an active call and dumping the channel variable data for > each leg of the call we don't see any of the additional parameters that are > in the Contact listed in the channel variables that are dumped for either > call leg's UUID. > > *Contact in INVITE into FreeSwitch* > Contact: ;isup-oli=00;attestation-info=1;origination-id=1000;verstat=0 > > *sip_contact_user Dump to X-Header:* > X-ContactUser: 15558675309 > > *FreeSwitch Logging:* > EXECUTE [depth=0] sofia/core/15558675309 at x.x.x.x:5060 export(sip_h_X-ContactUser=15558675309) > 2021-06-18 15:24:10.079576 [DEBUG] switch_channel.c:1310 EXPORT (export_vars) [sip_h_X-ContactUser]=[15558675309] > EXECUTE [depth=0] sofia/core/15558675309 at x.x.x.x:5060 export(sip_h_X-ContactParam=) > 2021-06-18 15:24:10.079576 [DEBUG] switch_channel.c:1310 EXPORT (export_vars) [sip_h_X-ContactParam]=[UNDEF] > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.com Tue Jun 22 17:17:58 2021 From: brian at freeswitch.com (Brian West) Date: Tue, 22 Jun 2021 12:17:58 -0500 Subject: [Freeswitch-users] Copy or Extract Contact Parameters from A-leg to B-leg In-Reply-To: References: Message-ID: Have you used the info app or the uuid_dump to investigate where it is in the variables? /b On Tue, Jun 22, 2021 at 11:56 AM Chris Wallace wrote: > We are working on implementing STIR/SHAKEN and have run into a snag when > passing calls through our FreeSwitch LCR which sits between our call > feature server and our SBC's. Our call feature server is passing parameters > in the Contact for attestation-info and origination-id. It is intended that > our SBC's will extract this data from the Contact and use it to generate an > identity header before leaving our network. Ideally this information would > be passed via X-Header(s) but unfortunately this is what we have to work > with at the moment. > > We have tried to use both the "sip_invite_contact_params" and > "sip_contact_params" channel variables but neither of them seem to get any > data from the Contact within the inbound INVITE. We had success testing > with "sip_contact_user" and dumping that data to an X-Header but we can't > seem to get FreeSwitch to recognize or dump the other parameters. > > When debugging an active call and dumping the channel variable data for > each leg of the call we don't see any of the additional parameters that are > in the Contact listed in the channel variables that are dumped for either > call leg's UUID. > > *Contact in INVITE into FreeSwitch* > Contact: ;isup-oli=00;attestation-info=1;origination-id=1000;verstat=0 > > *sip_contact_user Dump to X-Header:* > X-ContactUser: 15558675309 > > *FreeSwitch Logging:* > EXECUTE [depth=0] sofia/core/15558675309 at x.x.x.x:5060 export(sip_h_X-ContactUser=15558675309) > 2021-06-18 15:24:10.079576 [DEBUG] switch_channel.c:1310 EXPORT (export_vars) [sip_h_X-ContactUser]=[15558675309] > EXECUTE [depth=0] sofia/core/15558675309 at x.x.x.x:5060 export(sip_h_X-ContactParam=) > 2021-06-18 15:24:10.079576 [DEBUG] switch_channel.c:1310 EXPORT (export_vars) [sip_h_X-ContactParam]=[UNDEF] > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Brian West | Co-founder and Developer Need Commercial support? email sales at freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: brian at freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com [image: https://www.facebook.com/signalwireinc?src=email] [image: https://twitter.com/freeswitch] -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.com Tue Jun 22 17:25:38 2021 From: brian at freeswitch.com (Brian West) Date: Tue, 22 Jun 2021 12:25:38 -0500 Subject: [Freeswitch-users] Copy or Extract Contact Parameters from A-leg to B-leg In-Reply-To: References: Message-ID: There are no X Headers if done correctly, I suspect if you uuid_dump or use the info app, you'll find what you'll need for this task. /b On Tue, Jun 22, 2021 at 12:22 PM David Villasmil < david.villasmil.work at gmail.com> wrote: > Have you tried just using “sip_h_[whatever]” to read from the A leg and > setting it on the bridge as nolocal? > > On Tue, 22 Jun 2021 at 17:25, Chris Wallace > wrote: > >> We are working on implementing STIR/SHAKEN and have run into a snag when >> passing calls through our FreeSwitch LCR which sits between our call >> feature server and our SBC's. Our call feature server is passing parameters >> in the Contact for attestation-info and origination-id. It is intended that >> our SBC's will extract this data from the Contact and use it to generate an >> identity header before leaving our network. Ideally this information would >> be passed via X-Header(s) but unfortunately this is what we have to work >> with at the moment. >> >> We have tried to use both the "sip_invite_contact_params" and >> "sip_contact_params" channel variables but neither of them seem to get any >> data from the Contact within the inbound INVITE. We had success testing >> with "sip_contact_user" and dumping that data to an X-Header but we can't >> seem to get FreeSwitch to recognize or dump the other parameters. >> >> When debugging an active call and dumping the channel variable data for >> each leg of the call we don't see any of the additional parameters that are >> in the Contact listed in the channel variables that are dumped for either >> call leg's UUID. >> >> *Contact in INVITE into FreeSwitch* >> Contact: ;isup-oli=00;attestation-info=1;origination-id=1000;verstat=0 >> >> *sip_contact_user Dump to X-Header:* >> X-ContactUser: 15558675309 >> >> *FreeSwitch Logging:* >> EXECUTE [depth=0] sofia/core/15558675309 at x.x.x.x:5060 export(sip_h_X-ContactUser=15558675309) >> 2021-06-18 15:24:10.079576 [DEBUG] switch_channel.c:1310 EXPORT (export_vars) [sip_h_X-ContactUser]=[15558675309] >> EXECUTE [depth=0] sofia/core/15558675309 at x.x.x.x:5060 export(sip_h_X-ContactParam=) >> 2021-06-18 15:24:10.079576 [DEBUG] switch_channel.c:1310 EXPORT (export_vars) [sip_h_X-ContactParam]=[UNDEF] >> >> _________________________________________________________________________ >> >> 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > -- > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Brian West | Co-founder and Developer Need Commercial support? email sales at freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: brian at freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com [image: https://www.facebook.com/signalwireinc?src=email] [image: https://twitter.com/freeswitch] -------------- next part -------------- An HTML attachment was scrubbed... URL: From denis.papes at shishko.eu Wed Jun 23 06:44:20 2021 From: denis.papes at shishko.eu (Denis Papes) Date: Wed, 23 Jun 2021 07:44:20 +0100 Subject: [Freeswitch-users] Outgoing calls with a AVM FritzBox 7490 In-Reply-To: <1lvho4-00BhAL-3q@mx1.helsinki.tools> References: <1lvho4-00BhAL-3q@mx1.helsinki.tools> Message-ID: <522ed988-f004-e75c-96c7-f9565d40eee0@shishko.eu> > I doubt this is the problem in my case > since enabling more codecs doesn't help and incoming calls with the same > codec restrictions work. But it is codec issue. In your case, FreeSWITCH calls using L16, while FritzBox does not support that FreeSWITCH > Media Attribute (a): rtpmap:102 L16/8000 FritzBox > Media Attribute (a): rtpmap:2 G726-32/8000 > Media Attribute (a): rtpmap:102 G726-32/8000 > Media Attribute (a): rtpmap:100 G726-40/8000 > Media Attribute (a): rtpmap:99 G726-24/8000 > Media Attribute (a): rtpmap:97 iLBC/8000 On 22/06/2021 15:55, Janne Heß wrote: > Hello everyone, > > I'm kind of lost with setting up FS to connect to my FritzBox 7490. > The goal is to use FS with spandsp to send and receive Faxes. > Now, most things work in my setup. FS can successfully register as a SIP client > and I can call the configured number from my mobile phoneand the call is > routed to the FS0 virtual modem. > > The problem is outgoing calls. Using the ATD command on the virtual modem > to call any external phone does not work. Internal FritzBox-configured numbers > don't work either. Using Wireshark I found that the FritzBox replies with > 488 Not Acceptable Here. Looking around the internet, this seems to be > related to the codec configuration. I doubt this is the problem in my case > since enabling more codecs doesn't help and incoming calls with the same > codec restrictions work. > > So I'm assuming FS sends an INVITE package that the FritzBox does not like for some reason. > I played around with the From field but that just results in the FritzBox not replying at all. > Does anyone know what might be going on here or is there someone with a working example? > I have attached the INVITE package and the response package. 192.168.0.130 is the FritzBox, > 192.168.0.133 is the FS host (with firewall disabled). > > Thank you in advance and best regards > Janne > > Frame 183405: 1278 bytes on wire (10224 bits), 1278 bytes captured (10224 bits) on interface -, id 0 > Ethernet II, Src: RealtekU_ff:ff:ff (52:54:00:ff:ff:ff), Dst: AVMAudio_ff:ff:ff (7c:ff:4d:ff:ff:ff) > Internet Protocol Version 4, Src: 192.168.0.133, Dst: 192.128.0.130 > User Datagram Protocol, Src Port: 5060, Dst Port: 5060 > Session Initiation Protocol (INVITE) > Request-Line: INVITE sip:0123456789 at 192.168.0.130:5060 SIP/2.0 > Message Header > Via: SIP/2.0/UDP 192.168.0.133;rport;branch=z9hG4bKcU2Qc6yc0cZ3p > Max-Forwards: 70 > From: "FSModem" ;tag=j2K0XQKgXBFmN > To: > Call-ID: 3deaaab4-4e08-123a-ce81-52540059d216 > [Generated Call-ID: 3deaaab4-4e08-123a-ce81-52540059d216] > CSeq: 37632110 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.10.6-release.12~64bit > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY > Supported: timer, path, replaces > Allow-Events: talk, hold, conference, refer > Authorization: Digest username="hylafaxtel", realm="fritz.box", nonce="XXXXX", algorithm=MD5, uri="sip:0123456789 at 192.168.0.130:5060", response="XXXXX" > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 225 > X-FS-Support: update_display,send_info > Remote-Party-ID: "FSModem" ;party=calling;screen=yes;privacy=off > Message Body > Session Description Protocol > Session Description Protocol Version (v): 0 > Owner/Creator, Session Id (o): FreeSWITCH 1624344691 1624344692 IN IP4 192.168.0.133 > Session Name (s): FreeSWITCH > Connection Information (c): IN IP4 192.168.0.133 > Time Description, active time (t): 0 0 > Media Description, name and address (m): audio 27112 RTP/AVP 102 101 > Media Attribute (a): rtpmap:102 L16/8000 > Media Attribute (a): rtpmap:101 telephone-event/8000 > Media Attribute (a): fmtp:101 0-16 > Media Attribute (a): ptime:20 > [Generated Call-ID: ba99ae97-4de7-123a-09b4-52540059d216] > [Generated Call-ID: 751E5AC1C59CE6B0 at 192.168.0.130] > [I removed most Call-IDs for brevity] > > > Frame 183406: 837 bytes on wire (6696 bits), 837 bytes captured (6696 bits) on interface -, id 0 > Ethernet II, Src: AVMAudio_ff:ff:ff (7c:ff:4d:ff:ff:ff), Dst: RealtekU_ff:ff:ff (52:54:00:ff:ff:ff) > Internet Protocol Version 4, Src: 192.168.0.130, Dst: 192.168.0.133 > User Datagram Protocol, Src Port: 5060, Dst Port: 5060 > Session Initiation Protocol (488) > Status-Line: SIP/2.0 488 Not Acceptable Here > Message Header > Via: SIP/2.0/UDP 192.168.0.133;rport=5060;branch=z9hG4bKcU2Qc6yc0cZ3p > From: "FSModem" ;tag=j2K0XQKgXBFmN > To: ;tag=CDE9429F6B0A3BC3 > Call-ID: 3deaaab4-4e08-123a-ce81-52540059d216 > [Generated Call-ID: 3deaaab4-4e08-123a-ce81-52540059d216] > CSeq: 37632110 INVITE > Warning: 399 0.0.0.0 "successful but result empty" > User-Agent: FRITZ!OS > Content-Type: application/sdp > Content-Length: 361 > Message Body > Session Description Protocol > Session Description Protocol Version (v): 0 > Owner/Creator, Session Id (o): user 12041099 12041099 IN IP4 129.143.6.130 > Session Name (s): call > Connection Information (c): IN IP4 192.168.0.130 > Time Description, active time (t): 0 0 > Media Description, name and address (m): audio 7080 RTP/AVP 8 0 2 102 100 99 97 101 > Media Attribute (a): sendrecv > Media Attribute (a): rtpmap:2 G726-32/8000 > Media Attribute (a): rtpmap:102 G726-32/8000 > Media Attribute (a): rtpmap:100 G726-40/8000 > Media Attribute (a): rtpmap:99 G726-24/8000 > Media Attribute (a): rtpmap:97 iLBC/8000 > Media Attribute (a): fmtp:97 mode=30 > Media Attribute (a): rtpmap:101 telephone-event/8000 > Media Attribute (a): fmtp:101 0-15 > Media Attribute (a): rtcp:7081 > [Generated Call-ID: 596BDCD5EDA427D7 at 192.168.0.130] > [Generated Call-ID: ba99ae97-4de7-123a-09b4-52540059d216] > [I removed most Call-IDs for brevity] > > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- A non-text attachment was scrubbed... Name: OpenPGP_signature Type: application/pgp-signature Size: 840 bytes Desc: OpenPGP digital signature URL: From c.wallace777 at gmail.com Wed Jun 23 16:09:30 2021 From: c.wallace777 at gmail.com (Chris Wallace) Date: Wed, 23 Jun 2021 12:09:30 -0400 Subject: [Freeswitch-users] Copy or Extract Contact Parameters from A-leg to B-leg In-Reply-To: References: Message-ID: Yes, we dumped the channel variables using the info application, unfortunately it didn't expose the contact parameters that were in the Contact. We did end up with a work around by enabling the "parse-all-invite-headers" in the sofia config and then that did indeed dump the full Contact to the "sip_i_contact" variable. We were then able to export that as a new X-Header to pass along to the SBC. Kind of messy but it got us working in the meantime. Ideally we would still like to be able to extract that info and place it back in the outbound contact of the b-leg. However, I have seen a couple of older posts that indicate it may be difficult because we are using the bridge application after the LCR lookup and it is difficult to manipulate the Contact because of the outbound configuration. Thoughts? --Chris On Tue, Jun 22, 2021 at 1:46 PM Brian West wrote: > Have you used the info app or the uuid_dump to investigate where it is in > the variables? > > /b > > > On Tue, Jun 22, 2021 at 11:56 AM Chris Wallace > wrote: > >> We are working on implementing STIR/SHAKEN and have run into a snag when >> passing calls through our FreeSwitch LCR which sits between our call >> feature server and our SBC's. Our call feature server is passing parameters >> in the Contact for attestation-info and origination-id. It is intended that >> our SBC's will extract this data from the Contact and use it to generate an >> identity header before leaving our network. Ideally this information would >> be passed via X-Header(s) but unfortunately this is what we have to work >> with at the moment. >> >> We have tried to use both the "sip_invite_contact_params" and >> "sip_contact_params" channel variables but neither of them seem to get any >> data from the Contact within the inbound INVITE. We had success testing >> with "sip_contact_user" and dumping that data to an X-Header but we can't >> seem to get FreeSwitch to recognize or dump the other parameters. >> >> When debugging an active call and dumping the channel variable data for >> each leg of the call we don't see any of the additional parameters that are >> in the Contact listed in the channel variables that are dumped for either >> call leg's UUID. >> >> *Contact in INVITE into FreeSwitch* >> Contact: ;isup-oli=00;attestation-info=1;origination-id=1000;verstat=0 >> >> *sip_contact_user Dump to X-Header:* >> X-ContactUser: 15558675309 >> >> *FreeSwitch Logging:* >> EXECUTE [depth=0] sofia/core/15558675309 at x.x.x.x:5060 export(sip_h_X-ContactUser=15558675309) >> 2021-06-18 15:24:10.079576 [DEBUG] switch_channel.c:1310 EXPORT (export_vars) [sip_h_X-ContactUser]=[15558675309] >> EXECUTE [depth=0] sofia/core/15558675309 at x.x.x.x:5060 export(sip_h_X-ContactParam=) >> 2021-06-18 15:24:10.079576 [DEBUG] switch_channel.c:1310 EXPORT (export_vars) [sip_h_X-ContactParam]=[UNDEF] >> >> _________________________________________________________________________ >> >> 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > > > -- > > Brian West | Co-founder and Developer > > Need Commercial support? email sales at freeswitch.com > > FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 > > > Email: brian at freeswitch.com > > Mobile: 918-424-9378 > > Website: https://www.FreeSWITCH.com > > [image: https://www.facebook.com/signalwireinc?src=email] > [image: > https://twitter.com/freeswitch] > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From dragos at freeswitch.org Thu Jun 24 12:25:28 2021 From: dragos at freeswitch.org (Dragos Oancea) Date: Thu, 24 Jun 2021 15:25:28 +0300 Subject: [Freeswitch-users] Call drop with 406 after being answered In-Reply-To: References: <001a01d75ea1$d1a3af70$74eb0e50$@gmail.com> <004301d75ee0$cfdc1030$6f943090$@gmail.com> <000a01d76350$1e650770$5b2f1650$@gmail.com> Message-ID: note that reply "406 Not Acceptable" is not the same as "488 Not Acceptable Here". Try a "sofia loglevel all 9" to get more information. On Thu, Jun 17, 2021 at 2:07 PM David Villasmil < david.villasmil.work at gmail.com> wrote: > Try forcing it to your supported codec > > On Thu, 17 Jun 2021 at 09:10, Igor Potjevlesh > wrote: > >> Hi David, >> >> >> >> On the top of the list yes. I just see that g729 is added and his not >> supported. But there is at least 1 codec supported. >> >> >> >> Regards, >> >> >> >> Igor. >> >> >> >> *De :* FreeSWITCH-users *De >> la part de* David Villasmil >> *Envoyé :* vendredi 11 juin 2021 23:30 >> >> >> *À :* FreeSWITCH Users Help >> *Objet :* Re: [Freeswitch-users] Call drop with 406 after being answered >> >> >> >> I hadn't registered this: >> >> >> >> Freeswitch receive on his leg B this in-dialog INVITE. >> >> >> >> >> >> is the codec coming from that re-invite acceptable by fs? " 406 Not >> Acceptable" usually means FS doesn't support the offered codec. >> >> >> Regards, >> >> >> >> David Villasmil >> >> email: david.villasmil.work at gmail.com >> >> phone: +34669448337 >> >> >> >> >> >> On Fri, Jun 11, 2021 at 5:43 PM Igor Potjevlesh < >> igor.potjevlesch at gmail.com> wrote: >> >> Hi David, >> >> >> >> I got a SIP trace. Do you think of another kind of trace? Something in >> addition of log level debug? >> >> >> >> Regards, >> >> >> >> Igor. >> >> >> >> *De :* FreeSWITCH-users *De >> la part de* David Villasmil >> *Envoyé :* vendredi 11 juin 2021 12:45 >> *À :* FreeSWITCH Users Help >> *Objet :* Re: [Freeswitch-users] Call drop with 406 after being answered >> >> >> >> Try getting a trace for the call. >> >> >> >> On Fri, 11 Jun 2021 at 10:12, Igor Potjevlesh >> wrote: >> >> Hello! >> >> >> >> I'm facing a problem with a call scenario with an in-dialog INVITE just >> after the call has been picked up. >> >> Freeswitch receive on his leg B this in-dialog INVITE. Not forwarded to >> leg A. >> >> In a SIP trace, I can see 406 Not Acceptable on leg B whereas there is no >> difference in the Media except "Bandwidth Information" which is "AS:80" >> versus "AS:82" in the RE-INVITE. >> >> I turn on the logs on debug, but nothing specific appears. The first >> thing: >> >> 3f6d8dd1-5415-4a58-a948-551d841cf5dd 2021-06-11 10:34:12.873542 [NOTICE] >> sofia.c:1012 Hangup sofia/internal/ABC [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] >> >> 3f6d8dd1-5415-4a58-a948-551d841cf5dd 2021-06-11 10:34:12.873542 [DEBUG] >> switch_ivr_bridge.c:787 BRIDGE THREAD DONE [sofia/internal/ABC] >> >> 3f6d8dd1-5415-4a58-a948-551d841cf5dd 2021-06-11 10:34:12.873542 [DEBUG] >> switch_core_state_machine.c:653 (sofia/internal/ ABC) State EXCHANGE_MEDIA >> going to sleep >> >> […] >> >> 5f1866f7-f6c6-43c9-83bb-07cbb51cc3a5 2021-06-11 10:34:12.893545 [DEBUG] >> switch_core_session.c:2815 sofia/lega/+FROM at W.X.Y.Z. skip receive >> message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) >> >> >> >> Is there any reason for this? SDP is not duplicated, but there is no >> specific reason for replying with 406. >> >> >> >> Regards, >> >> >> >> Igor. >> >> >> ------------------------------ >> >> [image: Avast logo] >> >> L'absence de virus dans ce courrier électronique a été vérifiée par le >> logiciel antivirus Avast. >> www.avast.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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com >> >> -- >> >> Regards, >> >> >> >> David Villasmil >> >> email: david.villasmil.work at gmail.com >> >> phone: +34669448337 >> >> _________________________________________________________________________ >> >> 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 at 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 at 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > -- > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From davidswalkabout at gmail.com Fri Jun 25 03:21:33 2021 From: davidswalkabout at gmail.com (David P) Date: Fri, 25 Jun 2021 15:21:33 +1200 Subject: [Freeswitch-users] "No video stun for a long time!" In-Reply-To: References: Message-ID: We've been using FS 10.5 for a year or so...with verto for Chrome, Firefox, and Safari on all desktop OSs, and we haven't seen a problem like this before. 0:30 into a call, the mp4 recording of the session starts showing the FS banner. The user is on Safari 14.0.2, macOS Mojave 10.14.6. The log seems fine until suddenly "No video stun for a long time!" appears repeatedly (see below). The media_timeout should be more than adequate: The only thing we changed recently that I can think of is that we stopped echoing the video back to verto. But testing across many client combinations hasn't revealed any problem with that. The only posts I can find that mention this warning are specific to iPhone, but we've had no problem on iPhone/iPad,macOS in general. Any guesses? The log snippet... 2021-06-24 17:54:08.219324 [INFO] avformat.c:2576 use video codec implementation Video: h264 (libx264), yuv420p(pc, gbr/unknown/unknown), 320x240, q=10-31, 82 kb/s 2021-06-24 17:54:08.219324 [NOTICE] avformat.c:785 video thread start 2021-06-24 17:54:08.219324 [DEBUG] switch_vpx.c:703 VPX VER:v1.8.1 VPX_IMAGE_ABI_VERSION:4 VPX_CODEC_ABI_VERSION:8 2021-06-24 17:54:08.219324 [DEBUG] conference_video.c:3380 Setting up video write codec VP8 at slot 0 group _none_ c82798ee-1825-4952-b0dd-8ed7c6969123 2021-06-24 17:54:08.241930 [NOTICE] switch_core_media.c:15852 Activating write resampler 2021-06-24 17:54:08.299856 [INFO] switch_vpx.c:564 config: vp8 2021-06-24 17:54:08.299856 [NOTICE] switch_vpx.c:599 VPX encoder reset (WxH/BW) from 0x0/0 to 320x240/1024 c82798ee-1825-4952-b0dd-8ed7c6969123 2021-06-24 17:54:08.419309 [DEBUG] conference_member.c:1764 Raw Codec Activation Success L16 at 8000hz 1 channel 20ms c82798ee-1825-4952-b0dd-8ed7c6969123 2021-06-24 17:54:08.419309 [DEBUG] conference_member.c:1811 Raw Codec Activation Success L16 at 48000hz 2 channel 20ms c82798ee-1825-4952-b0dd-8ed7c6969123 2021-06-24 17:54:08.439299 [DEBUG] conference_loop.c:1338 Setup timer soft success interval: 20 samples: 160 from codec PCMU 2021-06-24 17:54:08.679355 [DEBUG] avformat.c:595 colorspace = 0 using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX FMA3 BMI2 AVX2 profile Constrained Baseline, level 4.1 264 - core 155 r2917 0a84d98 - H.264/MPEG-4 AVC codec - Copyleft 2003-2018 - http://www.videolan.org/x264.html - options: cabac=0 ref=1 deblock=0:0:0 analyse=0x1:0x111 me=hex subme=2 psy=1 psy_rd=1.00:0.00 mixed_ref=0 me_range=16 chroma_me=1 trellis=0 8x8dct=0 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=0 threads=3 lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=0 weightp=0 keyint=60 keyint_min=30 scenecut=40 intra_refresh=0 rc_lookahead=10 rc=cbr mbtree=1 bitrate=81 ratetol=1.0 qcomp=0.00 qpmin=0 qpmax=69 qpstep=4 vbv_maxrate=81 vbv_bufsize=81 nal_hrd=none filler=0 ip_ratio=1.40 aq=1:1.00 2021-06-24 17:54:08.679355 [INFO] avformat.c:2576 use video codec implementation Video: h264 (libx264), yuv420p(pc, gbr/unknown/unknown), 320x240, q=-1--1, 81 kb/s 2021-06-24 17:54:08.679355 [NOTICE] avformat.c:785 video thread start c82798ee-1825-4952-b0dd-8ed7c6969123 2021-06-24 17:54:09.559310 [DEBUG] switch_rtp.c:7793 Correct audio ip/port confirmed. 12c6801b-9409-adbc-ba01-15e788df9503 2021-06-24 17:55:02.819310 [WARNING] switch_rtp.c:855 No audio stun for a long time! 12c6801b-9409-adbc-ba01-15e788df9503 2021-06-24 17:55:03.619308 [WARNING] switch_rtp.c:855 No video stun for a long time! 12c6801b-9409-adbc-ba01-15e788df9503 2021-06-24 17:55:34.059362 [WARNING] switch_rtp.c:855 No video stun for a long time! 12c6801b-9409-adbc-ba01-15e788df9503 2021-06-24 17:56:05.099323 [WARNING] switch_rtp.c:855 No video stun for a long time! On Thu, Mar 4, 2021 at 3:46 PM David P wrote: > Hi Allan, > > I don't know if the media_timeout=300 behavior you saw is a bug or not, > but I wanted to add my own observation of weirdness about hangup > cause MEDIA_TIMEOUT... > > I just noticed a conference end abruptly after one leg spoke for 5 > minutes. The logs aren't clear why this happened, but it seems that name="rtp-timeout-sec" value="300"/> in our sip_profiles/internal.xml is > the reason -- it seems that because the *other* leg of the conference > remained silent, the RTP timeout was reached. > > I couldn't find any confluence pages about MEDIA_TIMEOUT by googling. > > On Fri, Jan 8, 2021 at 1:08 AM < > freeswitch-users-request at lists.freeswitch.org> wrote: > >> >> ---------- Forwarded message ---------- >> From: Allan Kristensen >> To: FreeSWITCH Users Help >> Cc: >> Bcc: >> Date: Wed, 6 Jan 2021 19:37:33 +0100 >> Subject: [Freeswitch-users] rtp-timeout-sec VS media_timeout >> We had some problems with "hanging channels" for our webrtc clients (via >> kamailio). >> To solve the problem I tried to use "media_timeout" setting but it didn't >> really work. So I tried the deprecated "rtp-timeout-sec" and this actually >> works fine? >> >> Not working: >> >> >> Working: >> >> >> How to reproduce: Make a call using webrtc and just close browser window, >> after some time freeswitch should close the channel because of missing rtp. >> Instead it just keeps writing: "switch_rtp.c:853 No audio stun for a long >> time!" forever... >> >> Anyway, it works now....just curious why...a typo or bug? >> >> /Allan >> >> Using: >> FreeSWITCH Version 1.10.5-release+git~20200818T185121Z~25569c1631~64bit >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bullehs at gmail.com Sun Jun 27 16:23:53 2021 From: bullehs at gmail.com (HS) Date: Sun, 27 Jun 2021 21:23:53 +0500 Subject: [Freeswitch-users] Amazon EC2 Instances - Freeswitch In-Reply-To: References: Message-ID: Hello all, David - worked with your suggestions for setting up the production environment (separate Opensips + Freeswitch instances). Most seems to work well. Except, in the tcpdump, I see when I call the IVR, the reply is being sent to the correct IP but the port number is 7078. Any thoughts on the problem? And fix? Or more details required? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Mon Jun 28 09:31:24 2021 From: david.villasmil.work at gmail.com (David Villasmil) Date: Mon, 28 Jun 2021 10:31:24 +0100 Subject: [Freeswitch-users] Amazon EC2 Instances - Freeswitch In-Reply-To: References: Message-ID: The client is probably sending from 7078 and that’s why the reply is being sent there? On Sun, 27 Jun 2021 at 17:24, HS wrote: > Hello all, > > David - worked with your suggestions for setting up the production > environment (separate Opensips + Freeswitch instances). Most seems to work > well. Except, in the tcpdump, I see when I call the IVR, the reply is being > sent to the correct IP but the port number is 7078. > > Any thoughts on the problem? And fix? Or more details required? > > Thanks. > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From bullehs at gmail.com Mon Jun 28 12:01:37 2021 From: bullehs at gmail.com (HS) Date: Mon, 28 Jun 2021 17:01:37 +0500 Subject: [Freeswitch-users] Amazon EC2 Instances - Freeswitch In-Reply-To: References: Message-ID: Hi again. Thanks a lot for taking the time to respond. I thought port 7078 must be a default port or something and overlooked giving context. The pcap file shows the call coming from the correct IP address and port. However, the destination port (no matter what the IP) is always 7078. I am setup with one instance running Opensips in Amazon EC2 (using private IP in ACL) etc. Connects fine to Freeswitch on another instance (private IP). Call rings and is answered, but since the port is incorrect - I can't hear any audio. Additionally, I thought RTP would use ports from 16384 - 32768 range? Hope that helps a little. > -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Mon Jun 28 13:12:35 2021 From: david.villasmil.work at gmail.com (David Villasmil) Date: Mon, 28 Jun 2021 14:12:35 +0100 Subject: [Freeswitch-users] Amazon EC2 Instances - Freeswitch In-Reply-To: References: Message-ID: You really need to look at your configuration and understand what’s supposed to happen. Sip signaling is not the same as RTP. I’m guessing you followed some instructions to set it up. It probably instructed you to set the sip port to 7870 at some point and you simply forgot. Check both opensips at freeswitch’s config. If opensips is sending to freeswitch on 7870 it can only be because it is configured to do so in opensips somewhere, be it dispatcher or hard-coded in the config script, and FS is configured yo listen on 7870 which is by no means a standard sip port and wouldn’t be configured on FS by default. So you changed it at some point :) On Mon, 28 Jun 2021 at 13:02, HS wrote: > Hi again. > > Thanks a lot for taking the time to respond. I thought port 7078 must be a > default port or something and overlooked giving context. The pcap file > shows the call coming from the correct IP address and port. However, the > destination port (no matter what the IP) is always 7078. I am setup with > one instance running Opensips in Amazon EC2 (using private IP in ACL) etc. > Connects fine to Freeswitch on another instance (private IP). Call rings > and is answered, but since the port is incorrect - I can't hear any audio. > > Additionally, I thought RTP would use ports from 16384 - 32768 range? > > Hope that helps a little. > >> _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From kunaalz at gmail.com Tue Jun 22 23:17:40 2021 From: kunaalz at gmail.com (Kunal Mittal) Date: Tue, 22 Jun 2021 16:17:40 -0700 Subject: [Freeswitch-users] Issue with BLF and shared line Indications Message-ID: Looking for some help with BLF in Freeswitch. Randomly for few users BLF indicator for parking and shared lines stops working. Rebooting the phone fixes the problems for some time and then again it stops working. We have FS 1.10.6 Approx. 500 Users Registered (most of them registered remotely from different networks) 1000 Contacts (Multiple registrations) Using Call Parking and Shared lines, most phones subscribe for BLF to monitor the calls on shared line or parked calls. System Details: VM with 12 GB Ram and 8 Core CPU. Using Postgres for Core Data and profiles. Max 40-50 calls. Sometimes I notice high CPU on this server, while on other servers with no shared lines and no BLF, Virtual machines handles 100+ calls and 700 users. Anyone could provide any suggestions/hints to help troubleshoot this. Is this too much for one server to handle? Is BLF too complicated? -------------- next part -------------- An HTML attachment was scrubbed... URL: From rmartinez at redvoiss.net Fri Jun 25 20:42:33 2021 From: rmartinez at redvoiss.net (Ricardo Martinez) Date: Fri, 25 Jun 2021 16:42:33 -0400 Subject: [Freeswitch-users] Run a python script from dialplan. In-Reply-To: 820c3a99ea802e885bd0504383589323@mail.gmail.com References: 820c3a99ea802e885bd0504383589323@mail.gmail.com Message-ID: Hi I’m trying to run a script in python from the freeswitch dialplan. This scripts has calls to a local REDIS DB which I need to query every time a call arrives. So far I was able to compile freeswitch with mod_python, I even can run a test.py script from the fs_cli like this: freeswitch at reverse> python test 2021-06-25 16:36:36.007177 [NOTICE] mod_python.c:213 Invoking py module: test 2021-06-25 16:36:36.027120 [INFO] switch_cpp.cpp:1465 test Hello But when try to add in the test.py the “import redis” line in order to work with my redis db I’m getting : freeswitch at reverse> python test 2021-06-25 16:37:17.847161 [NOTICE] mod_python.c:213 Invoking py module: test 2021-06-25 16:37:17.847161 [ERR] mod_python.c:261 Error reloading module 2021-06-25 16:37:17.847161 [ERR] mod_python.c:165 Python Error by calling script "test": Message: No module named redis Exception: None Traceback (most recent call last) File: "/usr/share/freeswitch/scripts/test.py", line 1, in Maybe I’m not getting correctly the use of pythons scripts?.. Can I use a script that calls another module?... Is possible? Or maybe there is a better way to do this?. Thanks! *Ricardo* -------------- next part -------------- An HTML attachment was scrubbed... URL: From rmartinez at redvoiss.net Mon Jun 28 16:33:19 2021 From: rmartinez at redvoiss.net (Ricardo Martinez) Date: Mon, 28 Jun 2021 12:33:19 -0400 Subject: [Freeswitch-users] Rum python script from dialplan Message-ID: <91e37512dabf43fac91d1e5bb2c40fbd@mail.gmail.com> Hi I’m trying to run a script in python from the freeswitch dialplan. This scripts has calls to a local REDIS DB which I need to query every time a call arrives. So far I was able to compile freeswitch with mod_python, I even can run a test.py script from the fs_cli like this: freeswitch at reverse> python test 2021-06-25 16:36:36.007177 [NOTICE] mod_python.c:213 Invoking py module: test 2021-06-25 16:36:36.027120 [INFO] switch_cpp.cpp:1465 test Hello But when try to add in the test.py the “import redis” line in order to work with my redis db I’m getting : freeswitch at reverse> python test 2021-06-25 16:37:17.847161 [NOTICE] mod_python.c:213 Invoking py module: test 2021-06-25 16:37:17.847161 [ERR] mod_python.c:261 Error reloading module 2021-06-25 16:37:17.847161 [ERR] mod_python.c:165 Python Error by calling script "test": Message: No module named redis Exception: None Traceback (most recent call last) File: "/usr/share/freeswitch/scripts/test.py", line 1, in Maybe I’m not getting correctly the use of pythons scripts?.. Can I use a script that calls another module?... Is possible? Or maybe there is a better way to do this?. Thanks! Ricardo -------------- next part -------------- An HTML attachment was scrubbed... URL: From lists at telefaks.de Tue Jun 29 15:50:57 2021 From: lists at telefaks.de (Peter Steinbach) Date: Tue, 29 Jun 2021 17:50:57 +0200 Subject: [Freeswitch-users] Setting Contact Header in Register Message-ID: Hello, for a german SIP provider, I need to send the contact header in the Register request the following way: Contact: with 012345678 as the registered SIP trunk number Today we send: Contact: which is not accepted by them. The reason is: They filter SIP packets by their phone numbers in the contact header for security reasons. So my question is: Is there a way to set the desired behaviour in the Register request: Contact: -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de From brian at freeswitch.com Tue Jun 29 16:24:16 2021 From: brian at freeswitch.com (Brian West) Date: Tue, 29 Jun 2021 11:24:16 -0500 Subject: [Freeswitch-users] Setting Contact Header in Register In-Reply-To: References: Message-ID: set extension-in-contact in the gateway to prevent that. On Tue, Jun 29, 2021 at 11:14 AM Peter Steinbach wrote: > Hello, > > > for a german SIP provider, I need to send the contact header in the > Register request the following way: > > Contact: > > with 012345678 as the registered SIP trunk number > > Today we send: > > Contact: > > which is not accepted by them. The reason is: They filter SIP packets by > their phone numbers in the contact header for security reasons. > > So my question is: Is there a way to set the desired behaviour in the > Register request: > > Contact: > > > -- > With kind regards > Peter Steinbach > > Telefaks Services GmbH > mailto:lists (att) telefaks.de > Internet: www.telefaks.de > > > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Brian West | Co-founder and Developer Need Commercial support? email sales at freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: brian at freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com [image: https://www.facebook.com/signalwireinc?src=email] [image: https://twitter.com/freeswitch] -------------- next part -------------- An HTML attachment was scrubbed... URL: From dinesh.krishnamurthy at teleapps.com Tue Jun 29 16:40:15 2021 From: dinesh.krishnamurthy at teleapps.com (Dinesh Krishnamurthy) Date: Tue, 29 Jun 2021 22:10:15 +0530 Subject: [Freeswitch-users] webapi - call center module - queue functionality Message-ID: <000001d76d05$7199d510$54cd7f30$@teleapps.com> Hi all, We are trying to do the following with the help of the webapi and wanted to get some help. We are following the URL published in confluence page. https://freeswitch.org/confluence/display/FREESWITCH/mod_xml_rpc 1. We want to add an agent/set of agents to specific queues so they can take calls that come into those queues. We are able to do this via the xml file configuration but we could not find an equivalent API to do this. 2. We are trying to add or modify a user profile but only listing users is available through the API - E.g.: http://192.168.X.X:8080/webapi/list_users 3. We would also need to know if we are able to get the output via JSON, it is difficult to parse text files Any help would be appreciated Thank you, DK -------------- next part -------------- An HTML attachment was scrubbed... URL: