November 2021 - FreeSWITCH-users - FreeSWITCH Mailing Lists

SIP over DTLS support
by Telium Technical Support 04 Nov '21

04 Nov '21

Does FreeSwitch support SIP of DTLS? Not clear from the docs (though mentions of webrtc possibly using this)

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Documentation for hold?
by Antony Stone 04 Nov '21

04 Nov '21

Hi. I want to put a channel on hold in my dialplan. https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools tells me that there is a command "hold" to do this, but the link for the documentation is https://wiki.freeswitch.org/index.php?title=Misc._Dialplan_Tools_hold&actio… which does not take me anywhere useful. Can someone point me at the correct page for this command? Thanks, Antony. -- Police have found a cartoonist dead in his house. They say that details are currently sketchy. Please reply to the list; please *don't* CC me.

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Sofia and ODBC
by Antony Stone 04 Nov '21

04 Nov '21

Hi. https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+Databases tells me that "Instead of using the default SQLite database as the internal FreeSWITCH database, you can use any core database type. This works for: mod_db mod_voicemail mod_sofia mod_dingaling mod_cidlookup mod_lcr" Each of those is a link, and if I click on the mod_sofia link I get to the page https://freeswitch.org/confluence/display/FREESWITCH/mod_sofia which tells me nothing about using ODBC. If I scroll further down the first page I started from, the heading "Configuring FreeSWITCH to use databases" has a sub-section "mod_sofia" with a link pointing to https://freeswitch.org/confluence/display/FREESWITCH/Sofia+Configuration+Fi… whose only mention of ODBC is: odbc-dsn If you have ODBC support and a working dsn you can use it instead of SQLite <param name="odbc-dsn" value="dsn:user:pass"/> So, where can find the information telling me _how_ I can put my sofia configuration into ODBC, instead of XML files? Basically I'm looking for the equivalent of Asterisk's "Realtime" configuration option, where SIP connections can be held in a database table instead of a flat file, meaning that (a) it's easier to maintain dynamically, and (b) new entries get used without needing a reload. I'm liking FreeSwitch (after using it for about a month), but still finding it sometimes hard to discover the definitive documentation for what it can do (or how to make it do it). Thanks, Antony. -- This sentence contains exacly three erors. Please reply to the list; please *don't* CC me.

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disable video from being automatically added to sdp
by Jimmy Simpson 03 Nov '21

03 Nov '21

I can not seem to disable and stop Freeswitch from adding the m=video to the SDP: m=video 17076 RTP/AVP 104 105 Is there a channel variable I can set to disable all Video. The extra m=video keeps getting added. I only need audio only. This takes up an extra 3 ports that I do not want to take up. --------- in INVITE - From: udp:1.1.1.1:5060 To: udp:2.2.2.2:5060 ------ where 1.1.1.1 is the freeswitch, and 2.2.2.2 is the proxy INVITE sip:107@testsystem.example-domain.com SIP/2.0 Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKX2j1QgS2cepaK Route: <sip:testsystem.example-domain.com:5060> Max-Forwards: 67 From: "test10first test10last" <sip:106@testsystem.example-domain.com >;tag=yDttpta215tXc To: <sip:107@testsystem.example-domain.com> Call-ID: 6bb53aa6-b0aa-123a-8c82-0050568c1d26 CSeq: 43054544 INVITE Contact: <sip:mod_sofia@1.1.1.1:5060> User-Agent: Freeswitch Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 991 X-Type: audio X-FS-Support: update_display,send_info Remote-Party-ID: "test10first test10last" < sip:106@testsystem.example-domain.com>;party=calling;screen=yes;privacy=off v=0 o=Cisco9 1635189532 1635189533 IN IP4 1.1.1.1 s=Cisco9 c=IN IP4 1.1.1.1 t=0 0 m=audio 27140 RTP/AVP 102 18 103 9 0 8 106 101 a=rtpmap:102 opus/48000/2 a=fmtp:102 useinbandfec=1; maxaveragebitrate=14000; maxplaybackrate=8000; sprop-maxcapturerate=8000; ptime=20; minptime=10; maxptime=40 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:103 opus/48000/2 a=fmtp:103 useinbandfec=1; maxaveragebitrate=14000; maxplaybackrate=8000; sprop-maxcapturerate=8000; ptime=20; minptime=10; maxptime=120 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 telephone-event/48000 a=fmtp:106 0-16 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=video 17076 RTP/AVP 104 105 b=AS:3072 a=rtpmap:104 VP9/90000 a=rtpmap:105 VP8/90000 a=inactive a=rtcp-fb:104 ccm fir a=rtcp-fb:104 ccm tmmbr a=rtcp-fb:104 nack a=rtcp-fb:104 nack pli a=rtcp-fb:105 ccm fir a=rtcp-fb:105 ccm tmmbr a=rtcp-fb:105 nack a=rtcp-fb:105 nack pli

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Question regarding continuous SIP trace with Freeswitch
by Gregor Maier 02 Nov '21

02 Nov '21

Hello! I'm currently struggling how to continuously trace all SIP packages from and to Freeswitch. Tcpdump or sngrep solely or similar tools aren't a solution, because SIP is encrypted (TLS). Asterisk provides an internal interface which logs all (or only restricted hosts e.g.) SIP packages to pcap (after decryption or before encryption of course). I found https://freeswitch.org/confluence/display/FREESWITCH/Packet+Capture - but there is no solution which handles the case "registering as client to a providers SIP server" (there is no local private key involved - the provider is the owner of the private key). Voipmonitor is pretty oversized for the planned use case (home server running on an APU 4 device (https://pcengines.ch/apu4d4.htm) on a sd card). Is there a similar and practical way how Freeswitch can log SIP packages to pcap? Thanks Gregor

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Growing latency in conferences
by Christian Berger 02 Nov '21

02 Nov '21

Hi, we have recently come across a problem with high and growing latency during conferences. In order to make sure it has nothing to do with the telephone terminals, we have traced the calls directly at the FreeSwitches. What we see in this experiment is that latency caused by FreeSwitch seems to grow consistently at roughly 200ms per hour. It seems like this is a problem caused by a Jitter Buffer. Does the Conferencing Module use the Jitter Buffer for Bridging or does it use its own buffer. If it uses its own buffer, how can we configure it? Regards, Christian Christian Berger foncloud GmbH & Co KG Hahlweg 2a 36093 Künzell Tel: / Fax: +49 661 968990-99 Email: Christian.Berger(a)foncloud.net Web: www.foncloud.net P.S.: Wussten Sie schon? Unter https://www.foncloud.net/wissen finden Sie zahlreiche Informationen und hilfreiche Artikel rund um unsere Produkte und Services. Registergericht: Amtsgericht Fulda, Persönlich haftende Gesellschafterin der foncloud GmbH&Co.KG: Global Brain Network GmbHGeschäftsführer der Global Brain Network GmbH: Peter Krug Sitz der Gesellschaft: Künzell. Diese E-Mail enthält vertrauliche und/oder rechtlich geschützte Informationen. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrtümlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser Mail ist nicht gestattet. This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorised copying, disclosure or distribution of the material in this e-mail is strictly forbidden.

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Re: [Freeswitch-users] Codec problem, Freeswitch does not regard absolute_codec_string?
by Rahman Duran 01 Nov '21

01 Nov '21

Hi, I am struggling tuning Freeswitch for late codec negotiation. Here is local profile settings: <param name="inbound-late-negotiation" value="true"/> <param name="inbound-codec-prefs" value="G722,PCMA"/> <param name="outbound-codec-prefs" value="G722,PCMA"/> <param name="inbound-codec-negotiation" value="greedy"/> As you see I enabled late-negotiation and set codecs prefs to allow G722 and PCMA. My goal is to use G722 if both sides support it and fall back to PCMA if G722 is not an option for A leg or B leg. Here is the local dial plan config where I bridge calls: <extension name="local_accounts_freeswitch"> <condition regex="all"> <regex field="destination_number" expression="^[0-9]{4}$"/> <regex field="${local_user_exists}" expression="^true$|^TRUE$|^True$"/> <regex field="${santral_kontrol}" expression="^freeswitch$"/> <action application="set" inline="true" data="call_pickup_group=${user_data(${destination_number}@${domain_name} var call_pickup_group)}"/> <action application="hash" data="insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}"/> <action application="hash" data="insert/${domain_name}-group_pickup_last_uuid/${call_pickup_group}/${uuid}"/> <action application="hash" data="insert/${domain_name}-pickup_last_uuid/${destination_number}/${uuid}"/> <action application="log" data="ep_codec_string -- ${ep_codec_string}" /> <action application="set" data="inherit_codec=true"/> <action application="export" data="nolocal:absolute_codec_string='${ep_codec_string}'"/> <action application="set_profile_var" data="callee_id_name=${user_data(${destination_number}@${domain_name} var effective_caller_id_name)}"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="call_timeout=30"/> <action application="set" data="execute_on_answer=sched_hangup +21600 alloted_timeout" /> <action application="bridge" data="{origination_callee_id_name=${user_data(${destination_number}@${domain_name} var effective_caller_id_name)}}user/${destination_number}@${domain_name}"/> <action application="execute_extension" data="hata-${originate_disposition} XML hata_anonslari"/> </condition> </extension> So I export absolute_codec_string to B leg using ep_codec_string. When I test a call from 2730->2110, the call fails with "incompetible_destination". Call logs here: https://pastebin.freeswitch.org/view/b2978e79 The 2730 is a Cisco phone with "G722,PCMU,PCMA,G729,L16,iLBC" codec support. The 2110 is a Samsung phone that only supports "PCMA,PCMU,G729" codecs. As you see cisco invites with all codecs it supports and freeswitch sets absolute_codec_string with "G722,PCMA" as expected from the config. But as you see in the call logs Freeswitch only sends "G722" in SDP to Samsung phone so Samsung rejects the call. Why does Freeswitch only send G722? How can I fix this? What am I doing wrong? P.S. using debian packages: FreeSWITCH Version 1.10.6-release-18-1ff9d0a60e~64bit (-release-18-1ff9d0a60e 64bit) Regards, Rahman Duran

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