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freeswitch 1.8 and 1.10 Valet parking BLF problem
by Jason Holden 24 Aug '22

24 Aug '22

All, Trying to monitor parking slots for valet park on Freeswitch. Using Polycoms or Voice Operator Pannal regular extension BLF works fine but I do not see the BLF status for the parking slots if a call is on hold their. Does anyone have any recommendations? Not finding much online about this. Jason Holden Phone: 1-866-836-9198 X405 Direct: 7868009949 www.thinksimplicity.com <http://www.thinksimplicity.com>

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Welcome!
by Brian West 24 Aug '22

24 Aug '22

FreeSWITCHers, We've fixed the reverse DNS for lists.freeswitch.org, our RIR has removed our delegation multiple times, I think we've finally fixed the process that triggered it, please let me know if you experience any issues with this mailing list moving forward and again we apologize for the inconvenience. Thanks, Brian -- Brian West | Co-founder and Developer Need Commercial support? email sales(a)freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&e…> Email: brian(a)freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/> [image: https://www.facebook.com/signalwireinc?src=email] <https://www.facebook.com/freeswitch> [image: https://twitter.com/freeswitch] <https://twitter.com/freeswitch>

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VP8 conference FIR Signal not always sent when frames are dropped
by Josh H 18 Aug '22

18 Aug '22

Hi, I am using Verto to connect to FreeSwitch over webRTC in a conference call. This works fine most of the time however when using an older device that does not support hardware acceleration for the video encoding, it appears that some frames are dropped. The CPU is pegged at 100% utilisation, with 60-80% of that being from the chrome process. This is understandable when using older hardware however when this occurs FreeSwitch fails to send FIR packets to get a new keyframe, leading to the ‘video mute’ image being shown until the next scheduled keyframe (which with VP8 is, by default, around 3000 frames, equivalent to 100 seconds at 30fps). The client seems to be unaware of any issues as it continues to encode frames and send them over the network, leading me to believe it is an issue with FreeSwitch. My current solution is to enable the ‘kf-max-dist’ parameter in ‘vpx.conf.xml’ however this leads to occasional stuttering, I assume this is because it must encode a full frame every 10 seconds. The other solution I have tested is swapping to the H.264 codec (the only other codec supported by webRTC) which makes the issue less noticeable however the quality of the stream drops significantly when no FIR is requested. H.264 isn’t an option for me as some of my target clients are Android devices which do not support H.264 over webRTC. Given that sometimes the session works as expected with FIR packets being sent when needed, leading to a small video stutter before it recovers, I am unsure how to fix it or what logs etc to provide. Any pointers would be appreciated 😊 Thanks Josh

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freeswitch segfault in mod_cdr_pg_csv
by Matthew Grooms 15 Aug '22

15 Aug '22

Hey Everyone, I've been seeing freeswitch crash a bit in production recently and finally had a chance to look into the issue. Appears to be coming from mod_cdr_pg_csv ... $ /usr/local/freeswitch/bin/fs_cli -x "version" FreeSWITCH Version 1.10.7-release~64bit ( 64bit) 2022-08-15 12:00:02.033897 64.60% [CRIT] mod_cdr_pg_csv.c:274 INSERT command failed: connection not open Program terminated with signal SIGABRT, Aborted. #0 0x00007f0ec45cfa9f in raise () from /lib64/libc.so.6 [Current thread is 1 (Thread 0x7f0db4adc700 (LWP 1203217))] Missing separate debuginfos, use: ... (gdb) bt #0 0x00007f0ec45cfa9f in raise () from /lib64/libc.so.6 #1 0x00007f0ec45a2e05 in abort () from /lib64/libc.so.6 #2 0x00007f0ec4612037 in __libc_message () from /lib64/libc.so.6 #3 0x00007f0ec461919c in malloc_printerr () from /lib64/libc.so.6 #4 0x00007f0ec461af40 in _int_free () from /lib64/libc.so.6 #5 0x00007f0ec7248936 in freePGconn () from /lib64/libpq.so.5 #6 0x00007f0eba7ec8d0 in insert_cdr ( values=0x7f0dd41cd4a0 "'10.16.84.32','SubjectWell','+16193300768','+16199188365','default','2022-08-15 11:59:32',null,'2022-08-15 12:00:02',30,0,'NO_ANSWER','6fd81787-c290-430d-b67b-ebfafe9f29d5',null,null,'PCMU','PCMU','se"...) at mod_cdr_pg_csv.c:289 #7 my_on_reporting (session=0x7f0df80b1668) at mod_cdr_pg_csv.c:385 #8 0x00007f0ec75221cd in switch_core_session_reporting_state (session=session@entry=0x7f0df80b1668) at src/switch_core_state_machine.c:932 #9 0x00007f0ec7522b68 in switch_core_session_run (session=0x7f0df80b1668) at src/switch_core_state_machine.c:606 #10 0x00007f0ec751d1be in switch_core_session_thread (thread=<optimized out>, obj=0x7f0df80b1668) at src/switch_core_session.c:1736 #11 0x00007f0ec751891b in switch_core_session_thread_pool_worker (thread=0x7f0dc8759bd0, obj=<optimized out>) at src/switch_core_session.c:1800 #12 0x00007f0ec786658c in dummy_worker (opaque=0x7f0dc8759bd0) at threadproc/unix/thread.c:151 #13 0x00007f0ec50cb1cf in start_thread () from /lib64/libpthread.so.0 #14 0x00007f0ec45badd3 in clone () from /lib64/libc.so.6 (gdb) frame 6 #6 0x00007f0eba7ec8d0 in insert_cdr ( values=0x7f0dd41cd4a0 "'10.16.84.32','SubjectWell','+16193300768','+16199188365','default','2022-08-15 11:59:32',null,'2022-08-15 12:00:02',30,0,'NO_ANSWER','6fd81787-c290-430d-b67b-ebfafe9f29d5',null,null,'PCMU','PCMU','se"...) at mod_cdr_pg_csv.c:289 289 PQfinish(globals.db_connection); I'm going to try this in production unless someone else has a better idea on how to fix this ... --- mod_cdr_pg_csv.c.orig 2022-08-15 14:44:56.028404623 -0500 +++ mod_cdr_pg_csv.c 2022-08-15 14:45:55.122482570 -0500 @@ -286,7 +286,9 @@ error: - PQfinish(globals.db_connection); + if (PQstatus(globals.db_connection) == CONNECTION_OK) { + PQfinish(globals.db_connection); + } globals.db_online = 0; switch_mutex_unlock(globals.db_mutex); Thanks, -Matthew

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codec negotiation error
by Brian May 10 Aug '22

10 Aug '22

Hello, I receiving an incoming call, with the G.729 PCMA codec. I try to bridge this call to my VOIP provider. This generates the following SDP packet: === cut === v=0 o=FreeSWITCH 175841631 175841632 IN IP4 59.167.180.194 s=FreeSWITCH c=IN IP4 59.167.180.194 t=0 0 m=audio 29188 RTP/AVP 0 101 8 3 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 === cut === I think this is bad, as PCMA isn't listed in the codecs available. How do I debug this? I am assuming this might be the reason why I am not getting incoming audio (does this make sense?) Regards -- Brian May <brian(a)linuxpenguins.xyz> https://linuxpenguins.xyz/brian/

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launch an external program in dialplan
by Gilles SAUVAIRE 10 Aug '22

10 Aug '22

Hi there, I need inside a dialplan, to launch an external program, and to take the return of this program to use it in the dialplan. I think a LUA script can do it? If anyone has an example script, I'm interested. Or is another technique better? Launching an external program at each call can be very resource-intensive, maybe make a request on a specific web service port, or something else? the idea is to add the user's name in the from. (but this name changes depending on the external context, it must be calculated, I cannot make a simple database query) someone has already done it I guess I'm not the first to have this need... Thank you all... Gilles

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freeswitch 180 ringing retransmission interval update
by Patrick Karton 09 Aug '22

09 Aug '22

when b leg is in ringing state freeswitch retransmits 180 on a leg every 60 seconds even if b leg does not send 180. is there a param to increase that interval ? Thanks.

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problem with Grandstream FXO gateway
by Jason Holden 09 Aug '22

09 Aug '22

All, Any suggestions on the following? We have a Grandstream FXO gateway where a call is sent to and I receive a 180 rinigng followed by a 200ok message but the line plays back an audio busy message. Any ideas how to force a fail over / continueation in the dial plan in this situation? Jason Holden Phone: 1-866-836-9198 X405 Direct: 7868009949 www.thinksimplicity.com <http://www.thinksimplicity.com>

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Spurious DTMF characters when using RFC2833 with SRTP
by Andy Newlands 04 Aug '22

04 Aug '22

Hi, We have a reproducible problem where spurious DTMF characters are introduced on calls from Freeswitch to a remote (3rd party) IVR, but ONLY when using SRTP (not with RTP) When the spurious character is generated it is: 1. Only ever following a valid/legitimate character (one that was genuinely keyed) 2. Often has a very long duration (see below) 3. Appears to be a random character - often 0x00, but sometimes 0-9/A-F,#,* (and that's when we get problems with the IVR - FS automatically drops the 0x00 values). We have been able to reproduce this from mobile/cell phones and traditional land-lines. I added some code to: switch_rtp.c, in static handle_rfc2833_result_t handle_rfc2833(switch_rtp_t *rtp_session, switch_size_t bytes, int *do_cng), to try to address this. It logs an error (with the character hex value and duration) for DTMF characters with "long" durations (over 10000) and then discards them. As you can see, below, this is a fairly frequent occurrence (although most of the time, the character value is null, so FS drps it, anyway - but sometimes, "real" characters are introduced). 2022-02-28 14:00:53.743665 [NOTICE] switch_vpx.c:599 VPX encoder reset (WxH/BW) from 0x0/0 to 352x288/1024 2022-02-28 14:00:56.103665 [INFO] switch_channel.c:522 RECV DTMF 4:2400 2022-02-28 14:00:56.303664 [ERR] switch_channel.c:557 Ignored invalid DTMF on Call-ID: 3b5d08d3-d898-40b6-ae00-5886713ca6dd, DTMF: 0x0 Duration: 5063 2022-02-28 14:00:56.383661 [INFO] switch_channel.c:522 RECV DTMF 4:2400 2022-02-28 14:00:56.583665 [INFO] switch_channel.c:522 RECV DTMF 5:2400 2022-02-28 14:00:57.063662 [INFO] switch_channel.c:522 RECV DTMF 5:2400 2022-02-28 14:00:57.123670 [INFO] switch_channel.c:522 RECV DTMF 8:2720 2022-02-28 14:00:57.323673 [ERR] switch_rtp.c:686 Discarded long-duration DTMF on Call-ID: 3b5d08d3-d898-40b6-ae00-5886713ca6dd, DTMF: 0x0 Duration: 54088 2022-02-28 14:00:57.603671 [INFO] switch_channel.c:522 RECV DTMF 9:2400 2022-02-28 14:00:57.824102 [INFO] switch_channel.c:522 RECV DTMF 8:2720 2022-02-28 14:00:57.903663 [ERR] switch_rtp.c:686 Discarded long-duration DTMF on Call-ID: 3b5d08d3-d898-40b6-ae00-5886713ca6dd, DTMF: 0x0 Duration: 57002 2022-02-28 14:00:58.183668 [INFO] switch_channel.c:522 RECV DTMF 6:2400 2022-02-28 14:00:58.363666 [ERR] switch_rtp.c:686 Discarded long-duration DTMF on Call-ID: 3b5d08d3-d898-40b6-ae00-5886713ca6dd, DTMF: 0x35 Duration: 57073 2022-02-28 14:00:58.503667 [INFO] switch_channel.c:522 RECV DTMF 9:2400 2022-02-28 14:00:58.643667 [INFO] switch_channel.c:522 RECV DTMF 1:2400 2022-02-28 14:00:58.963663 [INFO] switch_channel.c:522 RECV DTMF 6:2400 2022-02-28 14:00:59.283665 [INFO] switch_channel.c:522 RECV DTMF 1:2400 The first spurious character is 0x00 (which FS will drop anyway) but this could easily be a valid character (and would escape detection because its duration is not too long - it's long but not unreasonable). So, this is an imperfect "fix" as we occasionally see a spurious character with a legitimate value and a "sensible" duration - sufficiently often for users to complain. The last [ERR] entry shows a '5' being introduced but this is dropped because the duration is "crazy" (57073) As mentioned, this does not happen with secure media disabled - only with SRTP. So, I am wondering if the code correctly processes a digit then, sometimes (somehow) incorrectly attempts to decode part of an audio packet as though it were RFC2833. Has anyone else experienced this? Does anyone know what may be causing the problem and what the fix might be. I can provide console output and tcp dumps if required, along with any other details that may help (I'm also comfortable making code changes - to for extra logging/diagnostics etc). Thank you. Kind regards, Andy

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Need help to setup stir shaken in Freeswitch
by Pete Kay 04 Aug '22

04 Aug '22

Hi Does anyone know how to configure Freeswitch for stir shaken? Could you give me some info or doc how to do it? Thanks, Pete

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