December 2021 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Problems with Hold, Dynamic Payload, Media Direction
by Fuad Trle 22 Dec '21

22 Dec '21

Hello, I have encountered several problems while configuring and testing FreeSWITCH. This is on a current master branch, but the same issues are present in older versions. The tests are done on a minimal config with these additions: vars: - remove auth - global_codec_prefs=OPUS,PCMU,PCMA,VP8 - rtp_pass_codecs_on_stream_change=true # To be able to toggle video stream profiles: - inbound-late-negotiation (with and w/o inherit codec) dialplan: - Simple dial plan. External to Internal flow. Condition for ext, then bridge call to the other side. Unfortunately, this Sofia options are removed from FS :( Are there alternatives for them (while using rtp_pass_codecs_on_stream_change)? renegotiate-codec-on-hold|true,false renegotiate-codec-on-reinvite|true,false (a) -- (FS) -- (b) *1st: Hold* - Break transcoding. -- INVITE (Opus, PCMA) --> | | -- INVITE (PCMA) --> | <---- OK (PCMA) ---- <------ OK (Opus) -------- | . . | <--- ReINVITE (Hold or video on/off) <---- ReINVITE (PCMA) ---- | x I tried with absolute codec string, inbound/outbound codec prefs... it seems that FS does not keep/honor previous channel state and does not differentiate stream change from stream parameters change (like media flow). *2nd: Dynamic Payload *- It will offer Opus to the other side, because both have it. But FS will renegotiate with new PT number on reinvite. This break stream for some endpoints. -- INVITE (Opus, 111) --> | | -- INVITE (Opus, 102) --> | <---- OK (Opus, 102) ---- <---- OK (Opus, 111) ---- | . . | <-- ReINVITE (Hold or new m=line) <- ReINVITE (Opus, 102) - | x Is there a way to originate call with a codec that has a dynamic payload number, but to choose PT number? Something like setting rfc2833-pt variable for outbound channel before originating call? *3rd Media direction* - FS does not propagate media direction parameter to inbound channel/leg. Inbound side expect media and some endpoints will drop call after awhile. I know that answering with recvonly does not have sense for audio, but for a video it has. -- INVITE (sendrecv) --> | | -- INVITE (sendrecv) --> | <---- OK (recvonly) ---- <---- OK (sendrecv) ---- | For late negotiation this is wrong. Even for early it should be possible to pass reinvite to the other side on media direction change.

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debian repos for Raspberry Pi broken?
by Johan Helsingius 21 Dec '21

21 Dec '21

Hi, Trying to install freeswitch on a RPi 4 (Debian 11 Bullseye), my /etc/apt/sources.list.d/freeswitch.list has: deb http://files.freeswitch.org/repo/deb/rpi/debian-release/ bullseye main deb-src http://files.freeswitch.org/repo/deb/rpi/debian-release/ bullseye main but when I try apt-get update && apt-get install -y freeswitch-meta-all I get Unable to locate package freeswitch-meta-all apt-cache search freeswitch only gives me: libsipwitch-dev - secure peer-to-peer SIP VoIP server - development files libsipwitch1 - secure peer-to-peer SIP VoIP server - shared libraries libsipwitch1-dbg - secure peer-to-peer SIP VoIP server - debug symbols sipwitch - secure peer-to-peer VoIP server for the SIP protocol sipwitch-cgi - secure peer-to-peer SIP VoIP server - CGI XML-RPC interface Any suggestions? Julf

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License Documentation
by Tom Lynn 20 Dec '21

20 Dec '21

Is this page till current? It has not been modified since 2016. Is there a more current version? https://freeswitch.org/confluence/display/FREESWITCH/Licensing Thank you, Tom Lynn

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FEC and bitrates
by David Villasmil 18 Dec '21

18 Dec '21

Hello all, i've been testing FEC and how low we can go with maxavgbiterate <param name="use-vbr" value="1"/> <param name="use-dtx" value="1"/> <param name="complexity" value="10"/> <param name="packet-loss-percent" value="15"/> <param name="asymmetric-sample-rates" value="1"/> <param name="bitrate-negotiation" value="1"/> <param name="keep-fec-enabled" value="1"/> <param name="use-jb-lookahead" value="true"/> <param name="maxaveragebitrate" value="14400"/> <param name="maxplaybackrate" value="8000"/> <param name="sprop-maxcapturerate" value="0"/> <param name="adjust-bitrate" value="1"/> we only see FEC if we set maxavgbiterate to 18000, we see no FEC with 14400 anyone knows what i'm missing? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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weird extra characters
by Joli Martinez 17 Dec '21

17 Dec '21

Hello, I am running freeswitch 1.10.3. I have several DID's on this system. Some work without any issues, but some somehow get some weird characters added to the SIP message. I have already verified that there are no extra characters on the number. It appears to be a certain combination that it is happening to. If I change the last number from 1024 to 1023, nothing get added but on 1024 it does. Any suggestions on why this is happening and/or how to fix/patch this? INVITE sip:933@13.13.13.13 SIP/2.0 Via: SIP/2.0/UDP 12.12.12.12:5080;rport;branch=z9hG4bK3FX45c6Sj2Ftc Max-Forwards: 69 From: "122233331024" <sip:122233331024%E2%80%AC@12.12.12.12 >;tag=7KmB96jZD2e8r To: <sip:933@13.13.13.13> Call-ID: 6d1bafb7-d855-123a-8dbc-129e3056e999 CSeq: 45235314 INVITE Contact: <sip:gw+668aa3cd-1be2-48a1-9fdd-6d85a405029f@12.12.12.12:5080 ;transport=udp;gw=668aa3cd-1be2-48a1-9fdd-6d85a405029f> 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: 297 X-accountcode: 12.12.12.12 X-FS-Support: update_display,send_info Remote-Party-ID: "122233331024" <sip:122233331024ooo@12.12.12.12 >;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1639556579 1639556580 IN IP4 12.12.12.12 s=FreeSWITCH c=IN IP4 12.12.12.12 t=0 0 m=audio 21634 RTP/AVP 9 0 8 101 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20

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Re: [Freeswitch-users] Failed outbound call
by Antony Stone 17 Dec '21

17 Dec '21

On Wednesday 15 December 2021 at 17:54:32, Sean Devoy via FreeSWITCH-users wrote: > Hi, > > I have a very strange problem. My customer has one extension out of 16 > that has an issue dialing outside numbers. She hears it ring 4 or 5 > times, then gets a "busy" signal. She says the recipient's phone does not > ring on their end. (I had another user test this and it is confirmed). I suggest a packet capture of the SIP communications between FreeSwitch and both the extension, and the outbound trunk, would be helpful in this case. Antony. -- Some mistakes are too much fun to make only once. Please reply to the list; please *don't* CC me.

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Originate with failover
by Gregor Nanger 17 Dec '21

17 Dec '21

Guys, I need some advice. I would like to make an originate to provider and if call fails, proceed to second provider. For this, I can use pipe |. But will pipe obey what I set in continue_on_fail? For example: 1. continue_on_fail=NORMAL_TEMPORARY_FAILURE,NO_ROUTE_DESTINATION 2. originate sofia/gateway/provider1|sofia/gateway/provider2 19005551212 XML default Any advice would be appreciated. Br, Gregor

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Problem with socket connection
by Lucas Ducculi 17 Dec '21

17 Dec '21

Hi, I am using FreeSWITCH (version 1.10.2-release-14-f7bdd3845a ~ 64bit), the system handles 30-70 simultaneous calls. On the other hand, we have an IVR system that uses the node-esl library to connect to FreeSWITCH. The problem we have is that some calls are dropped at different stages of the IVR menu, and in the FS log we see the following. In this case, a connection with the IVR was not established: 2021-12-15 13:31:43.332652 [DEBUG] sofia.c:7301 Channel sofia/external/ XXXXXXXXXX(a)sip.telnyx.com entering state [early][183] 2021-12-15 13:31:43.332652 [DEBUG] switch_core_media_bug.c:970 Attaching BUG to sofia/external/XXXXXXXXXX(a)sip.telnyx.com EXECUTE [depth=0] sofia/external/XXXXXXXXXX(a)sip.telnyx.com set_audio_level(read 0) EXECUTE [depth=0] sofia/external/XXXXXXXXXX(a)sip.telnyx.com socket( 127.0.0.1:8085 async full) 2021-12-15 13:31:43.332652 [NOTICE] mod_event_socket.c:450 Trying host: 127.0.0.1:8085 2021-12-15 13:31:43.332652 [ERR] mod_event_socket.c:484 Socket Error: Connection refused 2021-12-15 13:31:43.332652 [ERR] mod_event_socket.c:488 Socket Error! 2021-12-15 13:31:43.332652 [NOTICE] switch_core_state_machine.c:386 sofia/external/XXXXXXXXXX(a)sip.telnyx.com has executed the last dialplan instruction, hanging up. 2021-12-15 13:31:43.332652 [NOTICE] switch_core_state_machine.c:388 Hangup sofia/external/XXXXXXXXXX(a)sip.telnyx.com [CS_EXECUTE] [NORMAL_CLEARING] In this other example I don't see an error with the socket in the log, but the call cuts out just after reproducing this silence: 2021-12-14 21:20:08.732663 [DEBUG] switch_ivr_play_say.c:1933 done playing file silence_stream://250 2021-12-14 21:20:08.732663 [NOTICE] switch_core_state_machine.c:386 sofia/external/XXXXXXXXXX(a)sip.telnyx.com has executed the last dialplan instruction, hanging up. 2021-12-14 21:20:08.732663 [NOTICE] switch_core_state_machine.c:388 Hangup sofia/external/XXXXXXXXXX(a)sip.telnyx.com [CS_EXECUTE] [NORMAL_CLEARING] Other example: 2021-12-15 13:08:10.452655 [DEBUG] switch_ivr.c:632 sofia/external/ XXXXXXXXXX(a)sip.telnyx.com Command Execute [depth=1] play_and_get_digits(0 1 1 0 1234567890 phrase:greeting silence_stream://250 dialed_digits) EXECUTE [depth=1] sofia/external/XXXXXXXXXX(a)sip.telnyx.com play_and_get_digits(0 1 1 0 1234567890 phrase:greeting silence_stream://250 dialed_digits) 2021-12-15 13:08:10.492654 [DEBUG] switch_ivr.c:632 sofia/external/ XXXXXXXXXX(a)sip.telnyx.com Command Execute [depth=2] sleep(3000) EXECUTE [depth=2] sofia/external/XXXXXXXXXX(a)sip.telnyx.com sleep(3000) 2021-12-15 13:08:10.532671 [DEBUG] sofia.c:7301 Channel sofia/external/ XXXXXXXXXX(a)sip.telnyx.com entering state [ready][200] 2021-12-15 13:08:30.092667 [NOTICE] switch_core_state_machine.c:386 sofia/external/XXXXXXXXXX(a)sip.telnyx.com has executed the last dialplan instruction, hanging up. 2021-12-15 13:08:30.092667 [NOTICE] switch_core_state_machine.c:388 Hangup sofia/external/XXXXXXXXXX(a)sip.telnyx.com [CS_EXECUTE] [NORMAL_CLEARING] My impression is that the problem is related to the socket connection between the IVR and FreeSWITCH. The two things are installed on the same server (Linux Debian 10), any idea what could cause this problem?

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Re: [Freeswitch-users] Outlook mail to FS List???
by Normando Hall 15 Dec '21

15 Dec '21

The same for me from thunderbird. El 15/12/2021 a las 13:41, Sean Devoy via FreeSWITCH-users escribió: > _________________________________________________________________________ > > 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

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Outlook mail to FS List???
by Sean Devoy 15 Dec '21

15 Dec '21

I am embarrassed that I can't seem to send plain text when submitting to this group. Any tips on Outlook settings? I have selected "Plain Text" as the format. It still appears as an attachment in the list. Regards, Sean Devoy

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