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Call Group Assistance
by Lee Barnard 04 Oct '18

04 Oct '18

Hi Guys, Having an issue with a call group. Issue: Have 3 users 1. 1 External 2. 2 Internal Call rings to all 3 endpoints, however, when rejecting call on the external entity it clears call on all legs. If call is rejected on either internal entities continues to ring on all other entities. Can anyone give a pointer as to what to try to get the functionality the same as the internals across the external entities. NOTE: You can see some artefacts of things tried in the dial-string. Configuration is built from LUA script and appears in debug as below: 2018-10-03 12:32:52.430641 [NOTICE] switch_cpp.cpp:1365 Call Group Request 2018-10-03 12:32:52.430641 [NOTICE] switch_cpp.cpp:1365 Extension: 2000 and Domain: domain.co.uk 2018-10-03 12:32:52.430641 [NOTICE] switch_cpp.cpp:1365 Query Result: 2018-10-03 12:32:52.430641 [NOTICE] switch_cpp.cpp:1365 Extension: 2001 and Domain: domain.co.uk 2018-10-03 12:32:52.430641 [NOTICE] switch_cpp.cpp:1365 Query Result: 123456789 2018-10-03 12:32:52.430641 [NOTICE] switch_cpp.cpp:1365 Action Huntgroup Returns: <?xml version="1.0" encoding="UTF-8" standalone="no"?> <document type="freeswitch/xml"> <section name="directory"> <domain name="domain.co.uk"> <groups> <group name="groupname"> <users> <user id="123456789"> <params> <param name="dial-string" value="{hangup_after_bridge=true,fail_on_single_reject=false}sofia/internal/123456789(a)domain.co.uk"/> </params> <variables> <variable name="user_context" value="default"/> </variables> </user> <user id="2000" type="pointer"/> <user id="2001" type="pointer"/> </users> </group> </groups> </domain> </section> </document> 2018-10-03 12:32:52.430641 [DEBUG] freeswitch_lua.cpp:382 DBH handle 0xc6bb90 released. 2ffa1d17-2ae3-4920-8f48-23798644ed25 EXECUTE sofia/internal/987654321@IPADDRESS bridge([hangup_after_bridge=true,fail_on_single_reject=false]sofia/internal/123456789@domain.co.uk,user/2000@domain.co.uk,user/2001@domain.co.uk<mailto:sofia/internal/123456789@domain.co.uk,user/2000@domain.co.uk,user/2001@domain.co.uk>) Happy to share any additional information required – please be gentle, I am fairly newish to freeswitch. Thanks in advance Lee Kind regards, Lee Barnard Voice Systems Engineer T: 01202612000 F: 01202612111 E: Lee.barnard(a)voip-unlimited.net Online ordering now live! Visit: portal.voip-unlimited.net/ordering<https://portal.voip-unlimited.net/ordering> or speak to your Account Manager today. ________________________________ [VoiP Unlimited]<http://www.voip-unlimited.net> [Force India] <http://www.forceindiaf1.com/partners/voip-unlimited> [ITSPA] <https://www.voip-unlimited.net/voip-unlimited-named-best-business-itsp-fina…> [Facebook]<https://www.facebook.com/pages/VoIP-Unlimited/286623001407365>[cid:image8dda1d.PNG@aaf8f1d3.4fabb9b6]<https://plus.google.com/114885723213491901544>[cid:imagec180ba.PNG@3cf76013.479eeb80]<https://www.linkedin.com/company/voip-unlimited>[cid:image418cf6.PNG@78887a23.4285e855]<https://twitter.com/voipunlimited>[cid:image3d111d.PNG@9a6f9dd3.4dac0962]<https://www.youtube.com/watch?v=TmD1Eg3uCwY> [VU Guard DDoS Mitigation]<http://www.voip-unlimited.net/services/channel-ddos> 6 Albany Business Park, Cabot Lane Poole, Dorset, UK, BH17 7BX. This email including any attachments is intended only for the addressee named above and it may contain confidential or privileged information. If you are not the intended recipient please notify the sender and note that the contents must not be disclosed to anyone else. Voip Unlimited will not be liable for any error in transmission. You should carry out your own virus checks before opening any attachments. Opinions, conclusions and other information that do not relate to the official business of Voip Unlimited are neither given nor endorsed by it.

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npa_nxx_company_ocn database?
by Avi Marcus 03 Oct '18

03 Oct '18

mod_lcr mentions using npa_nxx_company_ocn to determine inter/intra-state or inter/intra-lata calls. Is there an updated database somewhere? I have npa_nxx_company_ocn from someone from years ago. Thanks! -Avi Marcus 1-718-989-9485 (USA) 02-372-1570 (Israel) 020-3298-2875 (UK)

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keep-alive RE-INVITE dropped calls
by Bipin Patel 03 Oct '18

03 Oct '18

hi, im suffering call drops from carrier in roughly 15mins and i reported this to the carrier and they replied with the below: If the duration of the call was a multiple of 15 minutes, please make sure that you can properly respond to the keep-alive RE-INVITE that we sends every 15 minutes. i have no clue on how this works in FS so can any1 point me the right direction on how to sole this, we use FS to route calls from our switch to carrier -- Regards, Bipin

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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

2 1

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

1 1

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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