April 2020 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Freeswitch drops calls after 32 seconds
by MikeKulls 20 Apr '20

20 Apr '20

I've installed freeswitch on a Raspberry PI 3 with pretty much bog stock config. The only change I've done is to delete the ipv6 profiles and changed the default password. I've connected a few different clients, 3 soft phones and 2 ATAs. When making calls with some of the clients the calls are dropped after 32 seconds. For other clients the calls will proceed for half an hour no issues. I have absolutely no complexity here, it's all internal phones on the local /24 lan. There are no firewalls, traffic only goes via a switch. I have no external SIP gateway, none of the clients are behind a NAT. I've done extensive googling and everything I find talks about NAT issues and VPNs, multiple subnets etc. I'm not running any of that. Packet capture and more details below. Some of the things I've tried - complete OS and freeswitch reinstall on the PI - disable IPv6 on the PI (made a big improvement, now the remote phone rings instantly instead of long delay) - full OS update before installing freeswitch - tried clients with both UDP and TCP - install freeswitch on CentOS 7. - Disable SIP ALG on router (traffic not going through router anyway) - Tried all devices on a different network switch - Tried client options like "use rport" etc What works: - Cisco SPA3102 ATA - Voiper latest version on Windows What doesn't work - Latest Voiper on android - Older version of voiper on windows - Billion 7404VGP - Android version of Voip by antisip TShark capture of a call dropping. IPs are: 192.168.1.245 Freeswitch 192.168.1.225 Windows PC with older copy of voiper 192.168.1.202 Cisco ATA 1 0.000000000 192.168.1.225 â 192.168.1.245 SIP/SDP 1038 Request: INVITE sip:1004@192.168.1.245;transport=UDP | 2 0.005573031 192.168.1.245 â 192.168.1.225 SIP 917 Status: 407 Proxy Authentication Required | 3 0.011167520 192.168.1.225 â 192.168.1.245 SIP 408 Request: ACK sip:1004@192.168.1.245;transport=UDP | 4 0.016217586 192.168.1.225 â 192.168.1.245 SIP/SDP 1314 Request: INVITE sip:1004@192.168.1.245;transport=UDP | 5 0.037157271 192.168.1.245 â 192.168.1.225 SIP 413 Status: 100 Trying | 6 0.094830064 192.168.1.245 â 192.168.1.202 SIP/SDP 1283 Request: INVITE sip:1004@192.168.1.202:5083;transport=tcp | 7 0.115792718 192.168.1.202 â 192.168.1.245 SIP 365 Status: 100 Trying | 8 0.115932040 192.168.1.245 â 192.168.1.202 TCP 54 5060 â 5083 [ACK] Seq=1230 Ack=312 Win=63784 Len=0 9 0.124784941 192.168.1.202 â 192.168.1.245 SIP 526 Status: 180 Ringing | 10 0.124863118 192.168.1.245 â 192.168.1.202 TCP 54 5060 â 5083 [ACK] Seq=1230 Ack=784 Win=63784 Len=0 11 0.151217241 192.168.1.245 â 192.168.1.225 SIP/SDP 1205 Status: 183 Session Progress | 12 2.263254019 192.168.1.202 â 192.168.1.245 SIP/SDP 901 Status: 200 OK | 13 2.263406934 192.168.1.245 â 192.168.1.202 TCP 54 5060 â 5083 [ACK] Seq=1230 Ack=1631 Win=63784 Len=0 14 2.272446033 192.168.1.245 â 192.168.1.202 SIP 471 Request: ACK sip:1004@192.168.1.202:5083;transport=tcp | 15 2.284211777 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 200 OK | 16 2.469876726 192.168.1.202 â 192.168.1.245 TCP 60 5083 â 5060 [ACK] Seq=1631 Ack=1647 Win=16000 Len=0 17 2.784653094 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 200 OK | 18 3.785916037 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 200 OK | 19 5.787085997 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 200 OK | 20 8.903231371 192.168.1.245 â 192.168.1.225 TCP 54 5060 â 56460 [ACK] Seq=1 Ack=1 Win=501 Len=0 21 8.903781836 192.168.1.225 â 192.168.1.245 TCP 60 [TCP ACKed unseen segment] 56460 â 5060 [ACK] Seq=1 Ack=2 Win=8212 Len=0 22 9.788128841 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 200 OK | 23 13.788284085 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 200 OK | 24 17.788442453 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 200 OK | 25 20.361180947 192.168.1.225 â 192.168.1.245 UDP 60 5060 â 5060 Len=4 26 21.788537538 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 200 OK | 27 25.788690954 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 200 OK | 28 29.789847278 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 200 OK | 29 32.583226442 192.168.1.245 â 192.168.1.202 TCP 54 [TCP Keep-Alive] 5060 â 5083 [ACK] Seq=1646 Ack=1631 Win=63784 Len=0 30 32.584076227 192.168.1.202 â 192.168.1.245 TCP 60 [TCP Keep-Alive ACK] 5083 â 5060 [ACK] Seq=1631 Ack=1647 Win=16000 Len=0 31 33.790019961 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 200 OK | 32 34.284637050 192.168.1.245 â 192.168.1.225 SIP 704 Request: BYE sip:1002@192.168.1.225:5060;transport=UDP | 33 34.327728390 192.168.1.245 â 192.168.1.202 SIP 672 Request: BYE sip:1004@192.168.1.202:5083;transport=tcp | 34 34.344437951 192.168.1.202 â 192.168.1.245 SIP 381 Status: 200 OK | 35 34.393228415 192.168.1.245 â 192.168.1.202 TCP 54 5060 â 5083 [ACK] Seq=2265 Ack=1958 Win=63784 Len=0 36 34.422931003 192.168.1.225 â 192.168.1.245 SIP 466 Status: 200 OK | 37 38.352414375 192.168.1.225 â 192.168.1.245 TCP 60 [TCP ACKed unseen segment] 56460 â 5060 [PSH, ACK] Seq=1 Ack=2 Win=8212 Len=4 38 38.352610259 192.168.1.245 â 192.168.1.225 TCP 56 [TCP Previous segment not captured] 5060 â 56460 [PSH, ACK] Seq=2 Ack=5 Win=501 Len=2 39 38.393526512 192.168.1.225 â 192.168.1.245 TCP 60 [TCP ACKed unseen segment] 56460 â 5060 [ACK] Seq=5 Ack=4 Win=8212 Len=0 -- This email has been checked for viruses by AVG. https://www.avg.com

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verto and conference for webrtc audio mcu
by Roberto Della Pasqua 16 Apr '20

16 Apr '20

Hello, after a lot researching about the best viable webrtc audio mcu conference server, I have discovered freeswitch, and seems excellent. Can somebody explain me how to: 1. Setup a mcu audio conference with webrtc clients, dynamically opening and closing rooms (no need of signaling, this will be managed outside) 2. How to compile opus codec for maximum speed, eg. FFIXED instead of float (negligible audio difference, but large scaling) 3. Lib_verto and lib_conference are needed, allright? 4. I like manage the freeswitch with lib_socket, does you have examples perhaps? Thank you for this amazing software and for help with hints for best config webrtc audio only mcu. Kind regards. Roberto Della Pasqua www.dellapasqua.com<http://www.dellapasqua.com>

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Changing RTP ports mid call
by Joli Martinez 16 Apr '20

16 Apr '20

Hello, We have a phone app that ties back to our freeswitch servers. Recently the android phones are getting one-way audio. It works on iPhones and desktop clients. Based on the pcap's and fs_cli I see that the audio ports changed after the sdp negotiations. Not sure why this is happening also see that the codec is set to "BUG Codec PCMU:0" Any Suggestions? 2020-04-14 15:56:07.661344 [DEBUG] switch_ivr_async.c:1500 No silence detection configured; assuming start of speech 2020-04-14 15:56:07.701439 [DEBUG] switch_core_io.c:448 Setting BUG Codec PCMU:0 2020-04-14 15:56:07.701439 [DEBUG] switch_rtp.c:1887 rtcp_stats_init: audio ssrc[1321291818] base_seq[30216] 2020-04-14 15:56:07.881347 [INFO] switch_rtp.c:7231 Auto Changing audio port from 124.123.104.212:10488 to 124.123.104.212:9658 Thanks,

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unable to post
by Joli Martinez 16 Apr '20

16 Apr '20

Hello, I am trying to figure out what the guidelines to post stuff is. When I ask a question recently my questions have not been posted. Am I not asking them in the correct way, or is there a limit to how many questions I can post within a given time frame? If that is the case could I get some feedback as to what I am doing wrong? I am an experienced freeswitch user in certain in areas and I am sometimes able to fix thing on my own, but when I am stuck I would like to be able to post stuff online to get help from the community. Thanks,

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Re: [Freeswitch-users] FreeSwitch for WebRTC to SIP/RTP gateway?
by Giovanni Maruzzelli 15 Apr '20

15 Apr '20

at the bottom of the mail you find instruction on how to unsubscribe On Wed, Apr 15, 2020 at 5:38 PM Yan Zhang <yan(a)petatrading.com> wrote: > WHY I AM STILL RECEIVING THIS ENDLESS MAIL NOTICE? > > Please remove me. Thanks > > > > On Tuesday, April 14, 2020, 06:48:26 PM EDT, <mickael.bride(a)orange.com> > wrote: > > > Hello all, > > > > Sorry in advance if my email is not clear enough or relevant. > > My need is to have some kind of WebRTC gateway between a web application > and a call server managing SIP/RTP (the webapp is a softphone, audio only). > > This call server is not exposed on Internet, this is mainly why I need a > proxy/gateway. > > > > I am quite a newbie on WebRTC / SIP / RTP. > > From what I have seen on Internet some projects/products are doing that > kind of functionality (Asterisk, Doubango webrtc2sip, Janus, Audio Code …) > , but I am not able to focus on the simplest way to do it. Also some are > commercial products but I would prefer open source, installed on CentOS. > > FreeSwitch seems really great but I wonder if it is not too “big” or > complicated for my needs. > > > > Questions are: > > · Is FreeSwitch relevant to have a proxy/gateway between a WebRTC > web app and a call server not exposed on Internet? > > · Is it quite “simple” to install and configure freeSwitch to > have that functionality? > > · Is there any other simpler project to do this? > > > > Any advice would be really appreciate. > > > > Thank you, > > > > Mickaël > > > > _________________________________________________________________________________________________________________________ > > Ce message et ses pieces jointes peuvent contenir des informations confidentielles ou privilegiees et ne doivent donc > pas etre diffuses, exploites ou copies sans autorisation. Si vous avez recu ce message par erreur, veuillez le signaler > a l'expediteur et le detruire ainsi que les pieces jointes. Les messages electroniques etant susceptibles d'alteration, > Orange decline toute responsabilite si ce message a ete altere, deforme ou falsifie. Merci. > > This message and its attachments may contain confidential or privileged information that may be protected by law; > they should not be distributed, used or copied without authorisation. > If you have received this email in error, please notify the sender and delete this message and its attachments. > As emails may be altered, Orange is not liable for messages that have been modified, changed or falsified. > Thank you. > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales(a)freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com > -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18

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FreeSwitch for WebRTC to SIP/RTP gateway?
by mickael.bride＠orange.com 15 Apr '20

15 Apr '20

Hello all, Sorry in advance if my email is not clear enough or relevant. My need is to have some kind of WebRTC gateway between a web application and a call server managing SIP/RTP (the webapp is a softphone, audio only). This call server is not exposed on Internet, this is mainly why I need a proxy/gateway. I am quite a newbie on WebRTC / SIP / RTP. >From what I have seen on Internet some projects/products are doing that kind of functionality (Asterisk, Doubango webrtc2sip, Janus, Audio Code ...) , but I am not able to focus on the simplest way to do it. Also some are commercial products but I would prefer open source, installed on CentOS. FreeSwitch seems really great but I wonder if it is not too "big" or complicated for my needs. Questions are: · Is FreeSwitch relevant to have a proxy/gateway between a WebRTC web app and a call server not exposed on Internet? · Is it quite "simple" to install and configure freeSwitch to have that functionality? · Is there any other simpler project to do this? Any advice would be really appreciate. Thank you, Mickaël _________________________________________________________________________________________________________________________ Ce message et ses pieces jointes peuvent contenir des informations confidentielles ou privilegiees et ne doivent donc pas etre diffuses, exploites ou copies sans autorisation. Si vous avez recu ce message par erreur, veuillez le signaler a l'expediteur et le detruire ainsi que les pieces jointes. Les messages electroniques etant susceptibles d'alteration, Orange decline toute responsabilite si ce message a ete altere, deforme ou falsifie. Merci. This message and its attachments may contain confidential or privileged information that may be protected by law; they should not be distributed, used or copied without authorisation. If you have received this email in error, please notify the sender and delete this message and its attachments. As emails may be altered, Orange is not liable for messages that have been modified, changed or falsified. Thank you.

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Answer ringing phone with uuid_phone_event behind load balancer
by Jose Fco. Irles Durá 13 Apr '20

13 Apr '20

Hi, I'm testing the command "uuid_phone_event <uuid> talk" for answer ringing channels on remote phones. With a standalone configuration this works perfectly, the NOTIFY with the "Event: talk" is sended to the phone. The problem comes when I have a load balancer (for managing registers), the request uri of the NOTIFY is populated with the value of "sofia_contact" function (I think, because is the value that I write for this function via xml_curl). I think that the request URI for this message must be the contact value for this dialog and not the "sofia_contact" value. I'm right? Best regards -- Jose Fco. Irles Durá

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hangup_hook on an originated leg
by David Villasmil 10 Apr '20

10 Apr '20

Hello, So I'm receiving an INVITE that i'm simply use to generate an origination, something like: <extension name="in_out_test"> <condition field="destination_number" expression="^(whatever)$"> <action application="log" data="ERR Starting lua script..."/> <action application="info" data=""/> <action application="set" data="hangup_after_bridge=true"/> <action application="python" data="foo-outbound"/> </condition> </extension> With that in a python script in which i need to set a hangup_hook (I need the originated variables like billsec, etc) like this: # Some headers data gathering, then: session.execute( "set", "session_in_hangup_hook=true" ) session.execute( "export", "nolocal:api_hangup_hook=python cdr" ) call_command = '{cldnum=' + CallStruct['to'] + \ ',ignore_early_media=true,audio1=' + CallStruct['audios'] + \ ',app_campaing_type=' + CallStruct['type'] + \ ',originate_retries=' + CallStruct['retry'] + \ ',app_phone_desc=' + CallStruct['phone_desc'] + \ ',app_gateway=' + CallStruct['gateway'] + \ ',originate_timeout=30' + \ ',app_from=' + CallStruct['to'] + \ ',app_to=' + CallStruct['from'] + \ ',origination_callee_id_number=' + CallStruct['to'] + \ ',origination_caller_id_number=' + CallStruct['from'] + \ '}sofia/gateway/' + CallStruct['gateway'] + '/' + CallStruct['to'] + ' 9999 XML public' But this doesn't seem to be doing anything.. Anyone ever done this before? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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Rejecting calls (for a missed call service)
by Sam van Herwaarden 10 Apr '20

10 Apr '20

Hi, I am working on an IVR system that is supposed to work as a missed call service: users call, it rings for a moment, either they or the system hangs up, and then the system places a call to them. The goal is to avoid charges on their end. I basically have this working but it doesn’t sound natural on the user’s side. It doesn’t sound like the phone is ringing, there are just two beeps before it disconnects. I am using the ‘hangup’ command, I’ve tried the default and using a CALL_REJECTED status. The latter is worse - I get a voice from the telco saying “the number you are trying to reach is unavailable” (not at all what I want). This is in India for what it’s worth. Does anyone have experience setting up such a system and advice on the commands I should be using? Thanks! Kind regards, Sam

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Mod_conference with many participants. Question for experts
by Gilles SAUVAIRE 10 Apr '20

10 Apr '20

Hello, I am making a conference with 100 participants, on an 8-core server, 16 gigabytes of ram dedicated to my freeswitch. 98 participant are in mute mode and only two participant active. 8 times out of 10 the conference does not work. (very poor quality hatched audio) 2 times out of 10 the conference works for a few minutes then becomes unstable. (very poor quality hatched audio) A htop at that time shows that all the cores are used but not more than 30%. There is also plenty of memory available on the server. <profile name = "000"> <param name = "domain" value = "$$ {domain}" /> <param name = "rate" value = "8000" /> <param name = "interval" value = "20" /> <param name = "caller-controls" value = "none" /> <param name = "moderator-controls" value = "none" /> <param name = "moh-sound" value = "silence" /> <param name = "suppress-events" value = "start-talking, stop-talking" /> <! - Energy level required for audio to be sent to the other users -> <param name = "energy-level" value = "100" /> <param name = "comfort-noise" value = "0" /> <param name = "conference-flags" value = "audio-always" /> </profile> I have already found a solution, just put interval at 60. <param name = "interval" value = "60" /> With this the conference with 100 participants and only 2 active works. I have many qustions 1) It's strange, the conference with all the members in mute attribute, shouldn't it normally do nothing? Mute members should be transparent when it comes to mixing, is that just broadcast? ( conference data = 66123@000++flags{endconf|nomoh|mute} ) Did I make a mistake in my profile? Is it normal ? 2) Is there an easier way to broadcast? That is, a call that "talks" to 90 calls? My idea is to make a conference with 2 or 3 participants and to broadcast this conference to all the other participants since he is not speaking. many thanks to you for your advice and ideas. Gilles

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