- FreeSWITCH-users - FreeSWITCH Mailing Lists

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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Using channel variables as values to time of day attributes in conditions
by Chad Phillips 09 Apr '20

09 Apr '20

Will this dialplan XML condition work in newer versions of FreeSWITCH? <condition wday="${REFORWARDING_DAYS}" hour="${REFORWARDING_HOURS}" break="never"> ${REFORWARDING_DAYS} and ${REFORWARDING_HOURS} are dialplan variables I set prior to calling the extension with that condition. I tried this on an older version of FreeSWITCH and it doesn't seem to work, I end up having to hard code the values for the 'wday' and 'hour' attributes.

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Accessing the Identity header in the dialplan
by Alexandru Tripon 09 Apr '20

09 Apr '20

Hello, I wish to kindly ask how can I access the "Identity" header. I set the following variable to true in my sip profile:`<param name="parse-all-invite-headers" value="true"/>` in order to access all invite header but without luck. The SIP invite header is the following: ` INVITE sip:1002@192.168.59.203 SIP/2.0. Record-Route: <sip:192.168.59.203;lr>. Call-ID: dd566f5f1c0c7f746226fdd374fdb2ae@0:0:0:0:0:0:0:0. CSeq: 1 INVITE. From: "1001" <sip:1001@192.168.59.203>;tag=1bcc4483. To: <sip:1002@192.168.59.203>. Via: SIP/2.0/UDP 192.168.59.203:5060;branch=z9hG4bK2003.33316c51.0. Via: SIP/2.0/UDP 192.168.59.213:5060 ;branch=z9hG4bK-323138-463e2aa4a02196f588595ac028866af3. Max-Forwards: 69. Contact: "1001" <sip:1001@192.168.59.213:5060 ;transport=udp;registering_acc=192_168_59_203>. User-Agent: Jitsi2.11.20200408Linux. Content-Type: application/sdp. Content-Length: 897. Date: Thu, 09 Apr 2020 10:11:24 GMT. Identity: eyJhbGciOiJFUzI1NiIsInBwdCI6InNoYWtlbiIsInR5cCI6InBhc3Nwb3J0IiwieDV1IjoiaHR0cHM6Ly9jZXJ0LmV4YW1wbGUub3JnL3Bhc3Nwb3J0LmNlciJ9.eyJhdHRlc3QiOiJBIiwiZGVzdCI6eyJ0biI6WyI8bnVsbD4iXX0sImlhdCI6MTU4NjQyNzA4NCwib3JpZyI6eyJ0biI6IjxudWxsPiJ9LCJvcmlnaWQiOiIxMjM0NTYifQ.MEQCIH9KqTnrIW6pAlCqrYbHEFoCF3AsS6crh4ptSEzfnOM_AiAjZrZJ2Ur1csuYvJN8NygYEg1N-3RseLSoNHQOEX_Z5w;info=< https://cert.example.org/passport.cer>;ppt="shaken". Test-Header: mytest. ` And the message log I get is : ` variable_sip_i_from: ["1001" <sip:1001@192.168.59.203>;tag=1bcc4483] variable_sip_i_to: [<sip:1002@192.168.59.203>] variable_sip_i_call_id: [dd566f5f1c0c7f746226fdd374fdb2ae@0:0:0:0:0:0:0:0] variable_sip_i_cseq: [1 INVITE] variable_sip_i_max_forwards: [69] variable_sip_i_contact: ["1001" <sip:1001@192.168.59.213:5060 ;transport=udp;registering_acc=192_168_59_203>] variable_sip_i_user_agent: [Jitsi2.11.20200408Linux] variable_sip_i_date: [Thu, 09 Apr 2020 10:11:24 GMT] variable_sip_i_content_type: [application/sdp] variable_sip_i_content_length: [897] variable_sip_i_via: [ARRAY::SIP/2.0/UDP 192.168.59.203:5060;branch=z9hG4bK2003.33316c51.0;received=192.168.58.201|:SIP/2.0/UDP 192.168.59.213:5060;branch=z9hG4bK-323138-463e2aa4a02196f588595ac028866af3] variable_sip_i_record_route: [<sip:192.168.59.203;lr>] variable_sip_i_test_header: [mytest] ` I tested this in latest release( https://github.com/signalwire/freeswitch/releases/tag/v1.10.2) and on master(commit: ae0444e9cbccdee55a80467d605e1e8c3363a36d) and the same thing happened: the Test-Header is present in the log but the Identity header is not. How can I access the "Identity" header in the dialplan? Thanks for your time, Tripon Alexandru

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