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[Freeswitch-dev] Build fails under BOOKWORM
by Andy Newlands 15 Oct '25

15 Oct '25

Hi, I'm not sure what details people would need to help solve the following, so I'll just post the problem, then follow up with whatever additional info is requested. Everything builds find under Bullseye, but I'm getting make errors when trying to build FS v1.10.12 in a Docker container, based on Bookworm, as follows: FINAL DOCKER OUTPUT ----------------------------------- 25 | RUN mv BTL/GW/modules.conf /freeswitch/modules.conf &&\ 26 | ./configure --enable-core-pgsql-support --with-java=/usr/lib/jvm/java-8-openjdk-amd64 --enable-optimization 27 | >>> RUN make -j 10 && make install 28 | RUN git clone https://github.com/xadhoom/mod_bcg729.git &&\ 29 | sed -ie 's/FS_INCLUDES=\/usr\/include\/freeswitch/FS_INCLUDES=\/usr\/local\/freeswitch\/include\/freeswitch/g' /freeswitch/mod_bcg729/Makefile &&\ -------------------- ERROR: failed to solve: process "/bin/sh -c make -j 10 && make install" did not complete successfully: exit code: 2 FINAL MAKE OUTPUT ------------------------------- 48.97 cc1: all warnings being treated as errors 48.98 make[1]: *** [Makefile:2654: src/libfreeswitch_la-switch_stun.lo] Error 1 48.98 make[1]: *** Waiting for unfinished jobs.... 50.15 In function 'get_prefixed_str', 50.15 inlined from 'get_prefixed_str' at src/switch_ivr.c:3896:20, 50.15 inlined from 'switch_ivr_set_user_xml' at src/switch_ivr.c:3943:3: 50.15 src/switch_ivr.c:3912:17: error: 'memcpy' forming offset [13, 1023] is out of the bounds [0, 13] [-Werror=array-bounds] 50.15 3912 | memcpy(buffer + prefix_size, str, buffer_size - prefix_size - 1); 50.15 | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ 50.45 cc1: all warnings being treated as errors 50.46 make[1]: *** [Makefile:2640: src/libfreeswitch_la-switch_ivr.lo] Error 1 61.93 make[1]: Leaving directory '/freeswitch' 61.93 make: *** [Makefile:1438: all] Error 2 I'm guessing this may be due to some kind of stricter make rules in Bookworm, but I've now idea how/where i'd manage these (I thought it was all controlled through makefile) Thanks, in advance. Andy

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[Freeswitch-dev] jitter buffer max drift param
by Julien Chavanton 27 Feb '23

27 Feb '23

Hi, I was looking at the doc: https://developer.signalwire.com/freeswitch/Channel-Variables-Catalog/jitte… And the maxdrift description was not making much sense when looking at the examples. I looked in the source code and it seems it is ignored completely. jitterbuffer_msec=40:500:20 Setting Jitterbuffer to 40ms (2 frames) (25 max frames)

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[Freeswitch-dev] Eror: 488 Not Acceptable Here while connecting two JS-SIP clients
by shwetank singh 06 Oct '22

06 Oct '22

Hi perfectly agree that there are so many queries to this end and almost all pointers are towards codec negotiation failures. i still have this issue after i see that codec negotiation goes through. i have set up 2 Js-SIP clients. webrtc. i am able to connect and register both clients (can see those in *sofia status profile external reg*) however when i place a call from one client (extension 1000) to another (extension 1001), i receive a "488 Not Acceptable Here" message and the call ends with "Bad Media Description" error and i never get the call on receiver side i am completely at the surface still and have given it a try with all solutions posted to this end attaching the entire log in a file here. please help here are some important bits from the file: INVITE sip INVITE sip:1001@11.22.33.44 SIP/2.0 Via: SIP/2.0/WSS 11.22.33.44;branch=z9hG4bK7948245 Max-Forwards: 69 To: <sip:1001@11.22.33.44> From: <sip:1000@11.22.33.44>;tag=jimkkm04qn Call-ID: aoc0tkcjs47mrj5n1jl4 CSeq: 3752 INVITE X-Test: tester-tested Contact: <sip:1000@11.22.33.44;ob> Content-Type: application/sdp Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY Supported: ice,replaces,outbound User-Agent: JsSIP 3.9.1 Content-Length: 1950 v=0 o=- 8689874841498562319 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE 0 a=extmap-allow-mixed a=msid-semantic: WMS qikHHSycCzJUUpbTJgCfWpaVC44ukpKp5iAX m=audio 59566 UDP/TLS/RTP/SAVPF 111 63 103 104 9 0 8 106 105 13 110 112 113 126 c=IN IP4 192.168.1.64 a=rtcp:9 IN IP4 0.0.0.0 a=candidate:3460887983 1 udp 2122260223 192.168.1.64 59566 typ host generation 0 network-id 1 network-cost 10 a=candidate:2403671225 1 udp 2122194687 172.27.239.86 55376 typ host generation 0 network-id 2 network-cost 50 a=candidate:2160789855 1 tcp 1518280447 192.168.1.64 9 typ host tcptype active generation 0 network-id 1 network-cost 10 a=candidate:3251036233 1 tcp 1518214911 172.27.239.86 9 typ host tcptype active generation 0 network-id 2 network-cost 50 a=ice-ufrag:KQsy a=ice-pwd:oTFySc5ArZ1oxVZ+qb+errt5 a=ice-options:trickle a=fingerprint:sha-256 C9:60:D5:F6:7C:26:A5:2B:A5:1C:C5:23:0A:36:4B:E2:CE:D5:C8:15:8A:AD:35:DB:2C:E3:7D:C4:E8:7C:D3:E2 a=setup:actpass a=mid:0 a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01 a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid a=sendrecv a=msid:qikHHSycCzJUUpbTJgCfWpaVC44ukpKp5iAX 871ec968-71eb-4810-a9ff-dcd4c80705b2 a=rtcp-mux a=rtpmap:111 opus/48000/2 a=rtcp-fb:111 transport-cc a=fmtp:111 minptime=10;useinbandfec=1 a=rtpmap:63 red/48000/2 a=fmtp:63 111/111 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:110 telephone-event/48000 a=rtpmap:112 telephone-event/32000 a=rtpmap:113 telephone-event/16000 a=rtpmap:126 telephone-event/8000 a=ssrc:3961239200 cname:Bb33wRyqhUohexBZ a=ssrc:3961239200 msid:qikHHSycCzJUUpbTJgCfWpaVC44ukpKp5iAX 871ec968-71eb-4810-a9ff-dcd4c80705b2 ================= Remote SDP: 2022-10-06 10:18:15.078122 99.83% [NOTICE] switch_channel.c:1123 New Channel sofia/external/1000(a)11.22.33.44 [] 2022-10-06 10:18:15.078122 99.83% [DEBUG] switch_core_state_machine.c:581 (sofia/external/1000(a)11.22.33.44) Running State Change CS_NEW (Cur 1 Tot 11) 2022-10-06 10:18:15.078122 99.83% [INFO] sofia.c:10465 sofia/external/ 1000(a)11.22.33.44 receiving invite from 3.7.130.46:54985 version: 1.10.8-dev git 98d2e62 2022-09-09 13:32:35Z 64bit call-id: aoc0tkcjs47mrj5n1jl4 2022-10-06 10:18:15.078122 99.83% [DEBUG] sofia.c:11674 Setting NAT mode based on websockets 2022-10-06 10:18:15.078122 99.83% [DEBUG] sofia.c:7499 Channel sofia/external/1000(a)11.22.33.44 entering state [received][100] 2022-10-06 10:18:15.078122 99.83% [DEBUG] sofia.c:7509 Remote SDP: v=0 o=- 8689874841498562319 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE 0 a=extmap-allow-mixed a=msid-semantic: WMS qikHHSycCzJUUpbTJgCfWpaVC44ukpKp5iAX m=audio 59566 UDP/TLS/RTP/SAVPF 111 63 103 104 9 0 8 106 105 13 110 112 113 126 c=IN IP4 192.168.1.64 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10;useinbandfec=1 a=rtpmap:63 red/48000/2 a=fmtp:63 111/111 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:110 telephone-event/48000 a=rtpmap:112 telephone-event/32000 a=rtpmap:113 telephone-event/16000 a=rtpmap:126 telephone-event/8000 a=rtcp:9 IN IP4 0.0.0.0 a=candidate:3460887983 1 udp 2122260223 192.168.1.64 59566 typ host generation 0 network-id 1 network-cost 10 a=candidate:2403671225 1 udp 2122194687 172.27.239.86 55376 typ host generation 0 network-id 2 network-cost 50 a=candidate:2160789855 1 tcp 1518280447 192.168.1.64 9 typ host tcptype active generation 0 network-id 1 network-cost 10 a=candidate:3251036233 1 tcp 1518214911 172.27.239.86 9 typ host tcptype active generation 0 network-id 2 network-cost 50 a=ice-ufrag:KQsy a=ice-pwd:oTFySc5ArZ1oxVZ+qb+errt5 a=ice-options:trickle a=fingerprint:sha-256 C9:60:D5:F6:7C:26:A5:2B:A5:1C:C5:23:0A:36:4B:E2:CE:D5:C8:15:8A:AD:35:DB:2C:E3:7D:C4:E8:7C:D3:E2 a=setup:actpass a=mid:0 a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01 a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid a=msid:qikHHSycCzJUUpbTJgCfWpaVC44ukpKp5iAX 871ec968-71eb-4810-a9ff-dcd4c80705b2 a=rtcp-mux a=rtcp-fb:111 transport-cc a=ssrc:3961239200 cname:Bb33wRyqhUohexBZ a=ssrc:3961239200 msid:qikHHSycCzJUUpbTJgCfWpaVC44ukpKp5iAX 871ec968-71eb-4810-a9ff-dcd4c80705b2 ============================================= codec negotiation. i have removed OPUS as it was giving me bitrate mismatch *G722, PCMU, PCMA are exact match* 2022-10-06 10:18:25.018122 99.80% [DEBUG] switch_core_media.c:5663 Audio Codec Compare [opus:111:48000:20:0:1]/[G722:9:8000:20:64000:1] 2022-10-06 10:18:25.018122 99.80% [DEBUG] switch_core_media.c:5663 Audio Codec Compare [opus:111:48000:20:0:1]/[PCMU:0:8000:20:64000:1] 2022-10-06 10:18:25.018122 99.80% [DEBUG] switch_core_media.c:5663 Audio Codec Compare [opus:111:48000:20:0:1]/[PCMA:8:8000:20:64000:1] 2022-10-06 10:18:25.018122 99.80% [DEBUG] switch_core_media.c:5663 Audio Codec Compare [red:63:48000:20:0:2]/[G722:9:8000:20:64000:1] 2022-10-06 10:18:25.018122 99.80% [DEBUG] switch_core_media.c:5663 Audio Codec Compare [red:63:48000:20:0:2]/[PCMU:0:8000:20:64000:1] 2022-10-06 10:18:25.018122 99.80% [DEBUG] switch_core_media.c:5663 Audio Codec Compare [red:63:48000:20:0:2]/[PCMA:8:8000:20:64000:1] 2022-10-06 10:18:25.018122 99.80% [DEBUG] switch_core_media.c:5663 Audio Codec Compare [ISAC:103:16000:30:32000:1]/[G722:9:8000:20:64000:1] 2022-10-06 10:18:25.018122 99.80% [DEBUG] switch_core_media.c:5663 Audio Codec Compare [ISAC:103:16000:30:32000:1]/[PCMU:0:8000:20:64000:1] 2022-10-06 10:18:25.018122 99.80% [DEBUG] switch_core_media.c:5663 Audio Codec Compare [ISAC:103:16000:30:32000:1]/[PCMA:8:8000:20:64000:1] 2022-10-06 10:18:25.018122 99.80% [DEBUG] switch_core_media.c:5663 Audio Codec Compare [ISAC:104:32000:30:32000:1]/[G722:9:8000:20:64000:1] 2022-10-06 10:18:25.018122 99.80% [DEBUG] switch_core_media.c:5663 Audio Codec Compare [ISAC:104:32000:30:32000:1]/[PCMU:0:8000:20:64000:1] 2022-10-06 10:18:25.018122 99.80% [DEBUG] switch_core_media.c:5663 Audio Codec Compare [ISAC:104:32000:30:32000:1]/[PCMA:8:8000:20:64000:1] 2022-10-06 10:18:25.018122 99.80% [DEBUG] switch_core_media.c:5663 Audio Codec Compare [G722:9:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2022-10-06 10:18:25.018122 99.80% [DEBUG] switch_core_media.c:5718 Audio Codec Compare [G722:9:8000:20:64000:1] ++++ is saved as a match 2022-10-06 10:18:25.018122 99.80% [DEBUG] switch_core_media.c:5663 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2022-10-06 10:18:25.018122 99.80% [DEBUG] switch_core_media.c:5663 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2022-10-06 10:18:25.018122 99.80% [DEBUG] switch_core_media.c:5663 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2022-10-06 10:18:25.018122 99.80% [DEBUG] switch_core_media.c:5663 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2022-10-06 10:18:25.018122 99.80% [DEBUG] switch_core_media.c:5718 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match 2022-10-06 10:18:25.018122 99.80% [DEBUG] switch_core_media.c:5663 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2022-10-06 10:18:25.018122 99.80% [DEBUG] switch_core_media.c:5663 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2022-10-06 10:18:25.018122 99.80% [DEBUG] switch_core_media.c:5663 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2022-10-06 10:18:25.018122 99.80% [DEBUG] switch_core_media.c:5663 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2022-10-06 10:18:25.018122 99.80% [DEBUG] switch_core_media.c:5718 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match 2022-10-06 10:18:25.018122 99.80% [DEBUG] switch_core_media.c:5663 Audio Codec Compare [CN:105:16000:20:0:1]/[G722:9:8000:20:64000:1] 2022-10-06 10:18:25.018122 99.80% [DEBUG] switch_core_media.c:5663 Audio Codec Compare [CN:105:16000:20:0:1]/[PCMU:0:8000:20:64000:1] 2022-10-06 10:18:25.018122 99.80% [DEBUG] switch_core_media.c:5663 Audio Codec Compare [CN:105:16000:20:0:1]/[PCMA:8:8000:20:64000:1] 2022-10-06 10:18:25.018122 99.80% [DEBUG] switch_core_media.c:5663 Audio Codec Compare [CN:13:8000:20:0:1]/[G722:9:8000:20:64000:1] 2022-10-06 10:18:25.018122 99.80% [DEBUG] switch_core_media.c:5663 Audio Codec Compare [CN:13:8000:20:0:1]/[PCMU:0:8000:20:64000:1] 2022-10-06 10:18:25.018122 99.80% [DEBUG] switch_core_media.c:5663 Audio Codec Compare [CN:13:8000:20:0:1]/[PCMA:8:8000:20:64000:1] ======================================================= SIP to receiver 2022-10-06 10:18:25.018122 99.80% [DEBUG] mod_sofia.c:913 Local SDP sofia/external/1000(a)11.22.33.44: v=0 o=FreeSWITCH 1665021629 1665021630 IN IP4 11.22.33.44 s=FreeSWITCH c=IN IP4 11.22.33.44 t=0 0 a=msid-semantic: WMS VUSFGZ4CEjXvMLYzijtMWkwcRx3I8zET m=audio 29876 UDP/AVPF 9 126 a=rtpmap:9 G722/8000 a=rtpmap:126 telephone-event/8000 a=ptime:20 a=sendrecv a=rtcp:29877 IN IP4 11.22.33.44 a=ice-ufrag:0OsMZPPHcIga8KUi a=ice-pwd:XaZYj9vC6oyD4TjUxCxnhcUk a=candidate:0153546995 1 udp 2130706431 11.22.33.44 29876 typ host generation 0 a=candidate:0153546995 2 udp 2130706430 11.22.33.44 29877 typ host generation 0 a=end-of-candidates a=ssrc:3007569695 cname:kmpRbjjGt625HQuP a=ssrc:3007569695 msid:VUSFGZ4CEjXvMLYzijtMWkwcRx3I8zET a0 a=ssrc:3007569695 mslabel:VUSFGZ4CEjXvMLYzijtMWkwcRx3I8zET a=ssrc:3007569695 label:VUSFGZ4CEjXvMLYzijtMWkwcRx3I8zETa0 2022-10-06 10:18:25.018122 99.80% [NOTICE] mod_dptools.c:1408 Channel [sofia/external/1000(a)11.22.33.44] has been answered send 985 bytes to wss/[3.7.130.46]:54985 at 10:18:25.032155: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/WSS 11.22.33.44;branch=z9hG4bK7948245;received=3.7.130.46;rport=54985 From: <sip:1000@11.22.33.44>;tag=jimkkm04qn To: <sip:1001@11.22.33.44>;tag=0jtBBc4ryZ8Zg Call-ID: aoc0tkcjs47mrj5n1jl4 CSeq: 3752 INVITE Contact: <sip:1001@11.22.33.44:5080;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.10.8-dev+git~20220909T133235Z~98d2e6212c~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Session-Expires: 120;refresher=uas Content-Type: application/sdp Content-Disposition: session Content-Length: 196 Remote-Party-ID: "1001" <sip:1001@11.22.33.44 >;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1665021629 1665021630 IN IP4 11.22.33.44 s=FreeSWITCH c=IN IP4 11.22.33.44 t=0 0 a=msid-semantic: WMS VUSFGZ4CEjXvMLYzijtMWkwcRx3I8zET *m=audio 0 UDP/TLS/RTP/SAVPF 19* 2022-10-06 10:18:25.018122 99.80% [DEBUG] switch_channel.c:3950 (sofia/external/1000(a)11.22.33.44) Callstate Change EARLY -> ACTIVE 2022-10-06 10:18:25.018122 99.80% [DEBUG] sofia.c:7499 Channel sofia/external/1000(a)11.22.33.44 entering state [completed][200] EXECUTE [depth=0] sofia/external/1000(a)11.22.33.44 sleep(1000) ============================================== and then the BYE ------------------------------------------------------------------------ ACK sip:1001@11.22.33.44:5080;transport=udp SIP/2.0 Via: SIP/2.0/WSS 11.22.33.44;branch=z9hG4bK8228266 Max-Forwards: 69 To: <sip:1001@11.22.33.44>;tag=0jtBBc4ryZ8Zg From: <sip:1000@11.22.33.44>;tag=jimkkm04qn Call-ID: aoc0tkcjs47mrj5n1jl4 CSeq: 3752 ACK Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY Supported: outbound User-Agent: JsSIP 3.9.1 Content-Length: 0 recv 462 bytes from wss/[3.7.130.46]:54985 at 10:18:26.584127: ------------------------------------------------------------------------ BYE sip:1001@11.22.33.44:5080;transport=udp SIP/2.0 Via: SIP/2.0/WSS 11.22.33.44;branch=z9hG4bK162798 Max-Forwards: 69 To: <sip:1001@11.22.33.44>;tag=0jtBBc4ryZ8Zg From: <sip:1000@11.22.33.44>;tag=jimkkm04qn Call-ID: aoc0tkcjs47mrj5n1jl4 CSeq: 3753 BYE *Reason: SIP ;cause=488; text="Not Acceptable Here"* Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY Supported: outbound User-Agent: JsSIP 3.9.1 Content-Length: 0 -- Shwetank

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Re: [Freeswitch-dev] FreeSWITCH-dev Digest, Vol 167, Issue 3
by varun pratapsingh 29 Sep '22

29 Sep '22

Hi Brian, I am not doing billing. But I need to set session variable in lua script called from api_hangup_hook. I am able to access the session object and channel variable in lua script but could not set the channel variable from there. I can see that when CHANNEL_DESTROY comes it does not come with last set value from lua script. Thanks Varun On Tue, Sep 27, 2022, 5:30 PM <freeswitch-dev-request(a)lists.freeswitch.org> wrote: > Send FreeSWITCH-dev mailing list submissions to > freeswitch-dev(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > or, via email, send a message with subject or body 'help' to > freeswitch-dev-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-dev-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-dev digest..." > Today's Topics: > > 1. Cannot update a Channel Variable inside the api_hangup_hook > lua script (varun pratapsingh) > 2. Re: Cannot update a Channel Variable inside the > api_hangup_hook lua script (Brian West) > > > > ---------- Forwarded message ---------- > From: varun pratapsingh <varunps2003(a)gmail.com> > To: freeswitch-dev(a)lists.freeswitch.org > Cc: > Bcc: > Date: Mon, 26 Sep 2022 11:36:56 +0530 > Subject: [Freeswitch-dev] Cannot update a Channel Variable inside the > api_hangup_hook lua script > Hi Freeswitch Experts, > > I am using session_in_hangup_hook = true and api_hangup_hook to call my > lua script where I want to change the value of my channel variable through > session:setVariable("custom_channel_variable", "new_value"). But I can see > that I cannot do that. It is not getting changed. I am able to access the > channel variables through session:getVariable("custom_channel_variable") > but I cannot set them after the call is hung up. > > How can I do that? If I cannot do that then what is the reason? I do not > see any reason yet in the Freeswitch documentation. > > Can anyone enlighten me over this? > > Thanks and Regards > Varun Singh > > > > ---------- Forwarded message ---------- > From: Brian West <brian(a)freeswitch.com> > To: freeswitch-dev(a)lists.freeswitch.org > Cc: > Bcc: > Date: Mon, 26 Sep 2022 08:56:14 -0500 > Subject: Re: [Freeswitch-dev] Cannot update a Channel Variable inside the > api_hangup_hook lua script > Are you doing billing? > > On Mon, Sep 26, 2022 at 8:48 AM varun pratapsingh <varunps2003(a)gmail.com> > wrote: > >> Hi Freeswitch Experts, >> >> I am using session_in_hangup_hook = true and api_hangup_hook to call my >> lua script where I want to change the value of my channel variable through >> session:setVariable("custom_channel_variable", "new_value"). But I can see >> that I cannot do that. It is not getting changed. I am able to access the >> channel variables through session:getVariable("custom_channel_variable") >> but I cannot set them after the call is hung up. >> >> How can I do that? If I cannot do that then what is the reason? I do not >> see any reason yet in the Freeswitch documentation. >> >> Can anyone enlighten me over this? >> >> Thanks and Regards >> Varun Singh >> _________________________________________________________________________ >> >> 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-dev mailing list >> FreeSWITCH-dev(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> https://freeswitch.com > > > > -- > > 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> > _________________________________________________________________________ > > 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-dev mailing list > FreeSWITCH-dev(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > https://freeswitch.com

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[Freeswitch-dev] Cannot update a Channel Variable inside the api_hangup_hook lua script
by varun pratapsingh 26 Sep '22

26 Sep '22

Hi Freeswitch Experts, I am using session_in_hangup_hook = true and api_hangup_hook to call my lua script where I want to change the value of my channel variable through session:setVariable("custom_channel_variable", "new_value"). But I can see that I cannot do that. It is not getting changed. I am able to access the channel variables through session:getVariable("custom_channel_variable") but I cannot set them after the call is hung up. How can I do that? If I cannot do that then what is the reason? I do not see any reason yet in the Freeswitch documentation. Can anyone enlighten me over this? Thanks and Regards Varun Singh

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[Freeswitch-dev] Compiling mod_av cannot find definitions for enum AVLockOp, AV_LOCK_DESTROY
by Дилян Палаузов 10 Sep '22

10 Sep '22

Hello, in the past I could compile mod_av, but at some momemnt, it got not compilable. I have Fedora 36 system. There I can choose, whether to install as packages ffmpg or libav. I tried with both, it does not work. But I think that ffmpeg(-devel) is the right one, as https://libav.org/news/ shows no recent activity. The compiler cannot find the definitions of enum AVLockOp, AV_LOCK_CREATE, AV_LOCK_OBTAIN. See the error log below. The error message is below. mod/applications/mod_av/Makefile suggests installing libavformat-dev and libswscale-dev to build mod_av. Installing them does not help. At https://github.com/signalwire/freeswitch/pull/1626 I proposed an improvement to the fs_cli/mod_sofia autocompletion. Will it see any progress towards upstream? Greetings Дилян ---- making all mod_av make[4]: Entering directory '/git/freeswitch/freeswitch/src/mod/applications/mod_av' CC libavmod_la-mod_av.lo mod_av.c:52:38: error: ISO C forbids forward references to 'enum' types [-Werror=pedantic] 52 | int mod_av_lockmgr_cb(void **m, enum AVLockOp op) | ^~~~~~~~ mod_av.c:52:38: error: 'enum AVLockOp' declared inside parameter list will not be visible outside of this definition or declaration [-Werror] mod_av.c:52:47: error: parameter 2 ('op') has incomplete type 52 | int mod_av_lockmgr_cb(void **m, enum AVLockOp op) | ~~~~~~~~~~~~~~^~ mod_av.c: In function 'mod_av_lockmgr_cb': mod_av.c:62:22: error: 'AV_LOCK_CREATE' undeclared (first use in this function); did you mean 'AV_LOG_TRACE'? 62 | case AV_LOCK_CREATE: | ^~~~~~~~~~~~~~ | AV_LOG_TRACE mod_av.c:62:22: note: each undeclared identifier is reported only once for each function it appears in mod_av.c:72:22: error: 'AV_LOCK_OBTAIN' undeclared (first use in this function) 72 | case AV_LOCK_OBTAIN: | ^~~~~~~~~~~~~~ mod_av.c:79:22: error: 'AV_LOCK_RELEASE' undeclared (first use in this function) 79 | case AV_LOCK_RELEASE: | ^~~~~~~~~~~~~~~ mod_av.c:86:22: error: 'AV_LOCK_DESTROY' undeclared (first use in this function) 86 | case AV_LOCK_DESTROY: | ^~~~~~~~~~~~~~~ cc1: all warnings being treated as errors make[4]: *** [Makefile:1015: libavmod_la-mod_av.lo] Error 1 make[4]: Leaving directory '/git/freeswitch/freeswitch/src/mod/applications/mod_av' make[3]: *** [Makefile:727: mod_av-all] Error 1

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[Freeswitch-dev] Unit Tests Failing for FS PR Request (don't know why)
by Andy Newlands 04 Aug '22

04 Aug '22

Hi, I fixed a bug in switch_rtp.c and submitted a PR: https://github.com/signalwire/freeswitch/pull/1742 The unit tests are failing in an unrelated area of code (as far as I can tell), and I can't figure out how to resolve this (I can't get them to complete their run on my local, even with a fresh repo clone). The results are here: https://public-artifacts.signalwire.cloud/drone/signalwire/freeswitch/595/l… I believe the failed tests are these: uac_digest_leak_udp uac_digest_leak_tcp Is anyone able to confirm I've read the results correctly? Can anyone advise why these may have failed and how I can fix it (my changes appear to be unrelated, in switch_rtp.c) Thanks Kind regards, Andy

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[Freeswitch-dev] DOCS not available (503)
by Anton VG 28 May '22

28 May '22

Hi, Any problems with documentation? No docs seems available example https://freeswitch.org/confluence/display/FREESWITCH/mod_commands 503 service unavailable Contact team(a)signalwire.com or call +1.650.382.0000 <+16503820000>

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Re: [Freeswitch-dev] Is 3PCC broken in FreeSWITCH Version 1.10.6?
by KageDS 20 Sep '21

20 Sep '21

So to answer my own question, yes it does seem to be broken, the following patch fixes it but I'm not sure if this is the right thing to do: diff --git a/src/mod/endpoints/mod_sofia/sofia.c b/src/mod/endpoints/mod_sofia/sofia.c index b76a804..29f233d 100644 --- a/src/mod/endpoints/mod_sofia/sofia.c +++ b/src/mod/endpoints/mod_sofia/sofia.c @@ -6532,15 +6532,20 @@ static void sofia_handle_sip_r_invite(switch_core_session_t *session, int status switch_channel_clear_flag(channel, CF_REQ_MEDIA); if (status < 200) { - if (switch_core_session_get_partner(session, &other_session) == SWITCH_STATUS_SUCCESS) { - if(switch_core_session_compare(session, other_session)) { - private_object_t *other_tech_pvt = switch_core_session_get_private(other_session); - if (sofia_test_flag(other_tech_pvt, TFLAG_3PCC)) { - sofia_set_flag_locked(tech_pvt, TFLAG_SKIP_EARLY); - } - } - switch_core_session_rwunlock(other_session); - } +// +// **AlanE** +// The following code sets TFLAG_SKIP_EARLY for the b-leg of a call with a 3PCC on the a-leg +// this then later on causes the call to fail with no write codec error +// Just commenting the code out seems to fix the issue +// if (switch_core_session_get_partner(session, &other_session) == SWITCH_STATUS_SUCCESS) { +// if(switch_core_session_compare(session, other_session)) { +// private_object_t *other_tech_pvt = switch_core_session_get_private(other_session); +// if (sofia_test_flag(other_tech_pvt, TFLAG_3PCC)) { +// sofia_set_flag_locked(tech_pvt, TFLAG_SKIP_EARLY); +// } +// } +// switch_core_session_rwunlock(other_session); +// } if (sofia_test_flag(tech_pvt, TFLAG_SKIP_EARLY)) { return;

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[Freeswitch-dev] Is 3PCC broken in FreeSWITCH Version 1.10.6?
by KageDS 02 Sep '21

02 Sep '21

Hello list, I've just upgraded my Freeswitch from 1.4.6 to 1.10.6 and now audio calls that use 3PCC are no longer working. I get this error: [NOTICE] sofia.c:7419 Pre-Answer sofia/sipinterface_1/xxxxxx(a)sip.commpeak.com! [DEBUG] switch_channel.c:3565 (sofia/sipinterface_1/xxxxxx(a)sip.commpeak.com) Callstate Change DOWN -> EARLY [DEBUG] switch_ivr_originate.c:3883 Originate Resulted in Success: [sofia/sipinterface_1/xxxxxx(a)sip.commpeak.com] [DEBUG] switch_channel.c:2130 (sofia/umainterface_1/yyyyyy(a)sip.zzzz.com) Callstate Change RING_WAIT -> ACTIVE [DEBUG] switch_ivr_bridge.c:1793 (sofia/sipinterface_1/xxxxxx(a)sip.commpeak.com) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA [DEBUG] switch_core_state_machine.c:585 (sofia/sipinterface_1/xxxxxx(a)sip.commpeak.com) Running State Change CS_EXCHANGE_MEDIA (Cur 2 Tot 210) [DEBUG] switch_core_state_machine.c:654 (sofia/sipinterface_1/xxxxxx(a)sip.commpeak.com) State EXCHANGE_MEDIA [DEBUG] mod_sofia.c:656 SOFIA EXCHANGE_MEDIA [ERR] switch_core_media.c:15697 sofia/sipinterface_1/xxxxxx(a)sip.commpeak.com has no write codec. [NOTICE] switch_core_media.c:15698 Hangup sofia/sipinterface_1/xxxxxx(a)sip.commpeak.com [CS_EXCHANGE_MEDIA] [INCOMPATIBLE_DESTINATION] On the a_leg the SDP is provided in the usual 3PCC way: Client FS Carrier INVITE (no SDP) -----> <-------- 200 OK (SDP) ACK (SDP) -----------> INVITE (SDP) -----> <-------- 183 (SDP) CANCEL -----------> [Cause: INCOMPATIBLE_DESTINATION] <---------------- BYE If anyone has any insight to share I would be very grateful. Thanks Alan

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