September 2018 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Freeswitch failed to initiate outbound call using SIPs + SRTP (SRTP unprotect )
by Chhorm Chhatra 22 Oct '18

22 Oct '18

Hello, Currently, I faced a problem regarding SRTP outbound call to user (Leg B). The scenario is like this, - We set up our own root CA to an IP address (e.g 192.168.0.13) - We create a server certificate for freeswitch at 192.168.0.13 - Linphone is used as SIP client and is configured to trust our root CA by default. - Linphone A is configured to register to Freeswitch vis TLS + SRTP. (One leg call to server has both SIPs and SRTP – completely secure) - Linphone B is registered to Freeswitch via TLS + SRTP, and waiting for Linphone A to call to. (One leg call to server, e.g. 9196 (echo test), is completely secure with SRTP + SIPs) - Unfortunately, if A call to B, only A leg has SIPs + SRTP, but Leg B is not encrypted with SRTP and SIPs at all. This causes *SRTP unprotect failed with code 7 (auth check failed)**.* + Dialplan Configuration <action application="set" data="rtp_secure_media=true"/> <action application="export" data="rtp_secure_media=true"/> The dial-string is <action application="bridge" data="user/${dialed_extension}@${domain_name}"/> + Directory Configruation: <param name="dial-string" value="{rtp_secure_media=${ regex(${sofia_contact(${dialed_user}@${dialed_domain}) }|transport=tls)},presence_id=${dialed_user}@${dialed_ domain}}${sofia_contact(${dialed_user}@${dialed_domain})}" /> My question is that, is there any configuration left that I have to set up in order to let freeswitch initiate an outbound call to Leg B correctly with SRTP and SIPs (tls)? Any help would be really appreciated. Thank you so much. Best Regard,

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Multiple concurrent Verto calls in browser
by Michael Avers 05 Oct '18

05 Oct '18

Hello, I'm using Verto Communicator as a base for building a simple app to make/receive calls - and it works great. I'm trying to add the ability to manage multiple concurrent calls. My first thought is to create a new Verto service (in Angular) that maintains call state and reference to currently active call and otherwise keeps all calls in an array. Then to switch calls it'd place the currently active one on hold and then unhold any of the other calls in the array. Does this make sense? Any tips on handling multiple calls with Verto? Do I need a separate websocket connection for each? (they're all using the same proxy and credentials). Thanks! Mike

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Video Upgrade/Downgrade and Transcoding
by Carsten Bock 01 Oct '18

01 Oct '18

Hi, I am using FreeSwitch Release: FreeSWITCH (Version 1.9.0 -n20180903T123919Z-1~jessie+1git 5dd4451 2018-08-31 19:05:39Z 64bit) is ready I have the following, rather simple scenario, where I can't figure out, how to configure it properly. I use FreeSwitch as Transcoding B2B-UA/SBC. All the logic is done on a Kamailio in Front of FreeSwitch. Thus my dialplan is rather simple: <context name="public"> <extension name="other"> <condition> <action application="bridge" data="sofia/internal/${destination_number}@${loadbalancer}"/> </condition> </extension> </context> In order to provide transcoding, I have set the following in my internal.xml profile: <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="disable-transcoding" value="false"/> <param name="inherit_codec" value="false"/>  <param name="inbound-codec-negotiation" value="generous"/>  <param name="inbound-late-negotiation" value="true"/> In vars.xml, I have the following: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G722,PCMU,PCMA"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G722,PCMU,PCMA"/> <X-PRE-PROCESS cmd="set" data="media_mix_inbound_outbound_codecs=true"/> <X-PRE-PROCESS cmd="set" data="rtp_pass_codecs_on_stream_change=true"/> <X-PRE-PROCESS cmd="set" data="sip_renegotiate_codec_on_reinvite=true"/> This works like a charm for all Voice-Calls, the B-Party gets an offer including all the codecs from the list, e.g. A-Party supports only G722, B-Party supports only PCMU, so FreeSwitch does the transcoding. However, it leads to issues, when it comes to Video. As soon as I add H264 and VP8 to the codecs list (global_codec_pref / outbound_codec_prefs), the call to my B-Party includes the offer for Video as well, even if the A-Party did not offer Video initially. Is there an easy way to solve this? What are the right settings for such scenario? I tried the following in my dialplan:  <condition field="${switch_r_sdp}" expression="/(.*)(m=video \d+)(.*)/s" break="never"> <anti-action application="export" data="nolocal:absolute_codec_string=G722,PCMA,PCMU"/> </condition> This would remove the Video from the initial invite, if the A-Party did not offer Video initially. However, if I then enable Video at a later stage (Re-INVITE with m=video), FreeSwitch will not forward the new stream to the B-Party and then instead just send a re-INVITE with Audio only to the B-Party. Where did I go wrong? Any ideas? Thanks, Carsten -- Carsten Bock CEO (Geschäftsführer) ng-voice GmbH Millerntorplatz 1 20359 Hamburg / Germany http://www.ng-voice.com mailto:carsten@ng-voice.com Office +49 40 5247593-40 Fax +49 40 5247593-99 Sitz der Gesellschaft: Hamburg Registergericht: Amtsgericht Hamburg, HRB 120189 Geschäftsführer: Carsten Bock Ust-ID: DE279344284 Hier finden Sie unsere handelsrechtlichen Pflichtangaben: http://www.ng-voice.com/imprint/

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RTP problems when using "soft" timer
by Antonio Modesto 01 Oct '18

01 Oct '18

Hi everyone, I have had some problems lately with poor call quality when using FXO channels (mod_khomp) talking to sip phones. Using wireshark I noticed that the RTP stream sent from FS to my SIP device had some packets with wrong timestamps. After changing the rtp timer from soft to posix, the problem was solved. Do you guys know if there is any side effect in using the posix timer? *I am running FS 1.6.20 on a XenServer VM with Debian (Kernel 3.16.0). I can't upgrade to FS 1.8 because Khomp still does not support it. -- Antônio Modesto

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Verto Error
by Social Boh 29 Sep '18

29 Sep '18

Hello list, I can't trace and solve a problem with Verto: the error is: Client Connect from 1.2.3.4:61079 accepted [DEBUG] mod_verto.c:1992 1.2.3.4:61079 Starting client thread. [DEBUG] mod_verto.c:1839 1.2.3.4:61079 *WS SETUP FAILED* [DEBUG] mod_verto.c:2019 1.2.3.4:61079 Ending client thread. [DEBUG] mod_verto.c:2027 1.2.3.4:61079 Thread ended I found similar threads without solution. Let's Encrypt Certificate is for two sub domains and then: *cat fullchain.pem privkey.pem > wss.pem* Verto Configuration standard. *verto debug=10, fsctl loglevel 7, console loglevel 7* but no more messages on FS console. Thank you. Regards -- --- I'm SoCIaL, MayBe

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Encrypted video calls demo
by Joel Serrano 29 Sep '18

29 Sep '18

Hi All, I friend recently asked me about a PoC he is setting up and he needs encrypted video calls..blablabla. Long story short, I suggested to give FS + Verto a try blablabla.. but I wasn't sure if the "easy getting started" *today* would be debian8+fs16 or debian9+fs18... So my question is, for a really quick demo of FS+Verto would you go with 1.6 and debian8 or 1.8 and debian9 (for a first-time-fs-user)? I personally would test with v1.8 but not sure if the installation and can still be a little more tricky than v1.6 on debian8...? Opinions?? :D Thanks, Joel.

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TLS 1.3 support
by Sharath Kumar 28 Sep '18

28 Sep '18

Is there any plans to support TLS 1.3 in Freeswitch ? thank you, Shaks

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Unable to transfer to automated extension
by Tim Jones 27 Sep '18

27 Sep '18

(Apologies if this comes through twice on the list, but I can't see the original in my Inbox from list or my Sent items !) I've got a FS system where normal transfers (i.e. handset to handset) work fine. However, in an effort to combat telepests without raising blood pressure amongst people, I've created a special pest extension. The idea being you just transfer the pests to a pre-recorded announcement. However, all that happens is that you the caller gets cut-off. So I guess I'm missing an important config line or two ? <condition field="destination_number" expression="^196$" break="on-true">  <action application="log" data="TELEPEST TIME ! /> <action application="answer"/> <action application="set" data="playback_terminators=none"/> <action application="playback" data="/etc/freeswitch/its_audio/spammer_pre"/> <action application="playback" data="/etc/freeswitch/its_audio/spammer_main"/> <action application="hangup" data="INCOMING_CALL_BARRED"/> </condition>

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Unable to transfer to automated extension
by Tim Jones 27 Sep '18

27 Sep '18

Hi, Normal transfers to other extensions work fine (e.g. from one handset to another). However, we're getting tired of telepests and I've created a special extension. The idea is it saves you getting annoyed and shouting, you just transfer to the pest extension. However when attempting that transfer, all that happens is that it cuts the caller off. It doesn't matter if we try blind or normal transfers. Evidently I'm missing a config line or two ? <condition field="destination_number" expression="^196$" break="on-true">  <action application="log" data="TELEPEST TIME ! /> <action application="answer"/> <action application="set" data="playback_terminators=none"/> <action application="playback" data="/etc/freeswitch/its_audio/spammer_pre"/> <action application="playback" data="/etc/freeswitch/its_audio/spammer_main"/> <action application="hangup" data="INCOMING_CALL_BARRED"/> </condition>

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Fwd: Delivery Status Notification (Failure)
by Tiago Galvão Gomes de Souza 27 Sep '18

27 Sep '18

someone can help? I'm trying to install freeswitch on centos 7 , follow the instructions : https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7+and+RHEL+7#Ce…. .. when a use a command "make" to start installation a receveid error like this image below: [image: freeswitch.JPG] -- Atenciosamente, Tiago Galvão Gomes de Souza. -- Atenciosamente, Tiago Galvão Gomes de Souza.

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