- FreeSWITCH-users - FreeSWITCH Mailing Lists

Rearranging menus within retrieve voicemail app (mod_voicemail_ivr)
by AlexKr 14 Dec '20

14 Dec '20

Hello Everybody, I'm trying to remove some lines/menus from the 'retrieve voicemail' IVR. I'm using mod_voicemail_ivr for that. The issue is that changes to profile in the voicemail_ivr.conf.xml like commenting out any line in <apis> results in error: config === === log === [ERR] config.c:262 Missing api definition for profile 'default' === Commenting out any of menus does not prevent them from playing. Is there any trick that I'm missing with making changes to the mod_voicemail_ivr default config? Thank you in advance for any advice on the matter Alex -- Sent from: http://freeswitch-users.2379917.n2.nabble.com/

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Freeswitch doesn't run on Raspberry Pi 1
by Dušan Dragić 13 Dec '20

13 Dec '20

Hello all, I just installed Freeswitch 1.10.5 (1.10.5~release~6~25569c1631~buster-1~buster+1) from the offical freeswitch rpi repo on an old Raspberry Pi 1B, but trying to run freeswitch I'm getting "Illegal instruction". The OS is latest Raspberry Pi OS Lite, fully updated. If a take the SD card, plug it in my PC, mount it and run it in a chroot with qemu user arm emulation (qemu-arm-static) I can start freeswitch just fine, it only doesn't run on the rpi 1 hardware. I haven't installed any debug symbols so this isn't really useful, but gdb says: Program received signal SIGILL, Illegal instruction. 0xb68db728 in ?? () from /lib/arm-linux-gnueabihf/libcrypto.so.1.1 (gdb) bt full #0 0xb68db728 in ?? () from /lib/arm-linux-gnueabihf/libcrypto.so.1.1 No symbol table info available. #1 0xb68d518c in ?? () from /lib/arm-linux-gnueabihf/libcrypto.so.1.1 No symbol table info available. Backtrace stopped: previous frame identical to this frame (corrupt stack?) Does anyone know if the old rpi 1 is still supported by the packages from the repo or do they only work on newer pi models (armv7 cpus)? Regards, -- Dušan Dragić

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INCOMPATIBLE_DESTINATION sip softphone <-> jssip webrtc
by Henry S 11 Dec '20

11 Dec '20

Hi FS users I am stuck with INCOMPATIBLE reason of "Not Acceptable Here" 488 message. Calling from sipphone to sipphone works, calling from webrtc to webrtc works, calling a conference room from webrtc and sipphone (mixed) works, but not when directly crossing over. Sipphones tried are microsip and linphone I use the "proxy_media=true" for the webrtc calls to work. I have checked the codecs in the SIP bodies from both sides' INVITE message, they seem to have common codecs in different orders (not sure if this matters) Websocket endpoint listens to 127.0.0.1 7443 behind apache ws_tunnel. SIP 5060 listens on public IP Not sure why either side won't cross over. Text messages work fine from webrtc to sipphone. Any ideas?

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SIP dialin not working
by Markus Heller 11 Dec '20

11 Dec '20

Dear list, I am trying to configure SIP dialin. I can call my freeswitch/BBB server. It does execute the "play_and_get_digits", I can enter the PIN number of my conference which I can now see in my BBB GUI, but then I get a HANGUP. fs_clibbb tells me, when the call comes in: After entering the PIN number (DTMF decode works fine): EXECUTE [depth=0] sofia/external/[callernumber](a)sip.[xyz].com transfer(SEND_TO_CONFERENCE XML default) ... switch_ivr.c:2243 ... State Change CS_EXECUTE -> CS_ROUTING switch_ivr.c:2250 Transfer sofia/external/[callernumber](a)sip.[xyz].com to XML[SEND_TO_CONFERENCE@default] ... some more lines that don't look conspicuous... mod_dialplan_xml.c:637 Processing [callernumber] <[callernumber]>- >SEND_TO_CONFERENCE in context default Then there is a Dialplan block that does not terminate properly: Dialplan: sofia/external/[callernumber](a)sip.[xyz].com parsing [default- >unloop] continue=false Dialplan: sofia/external/[callernumber](a)sip.[xyz].com Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/external/[callernumber](a)sip.[xyz].com Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/external/[callernumber](a)sip.[xyz].com parsing [default- >bbb_conferences_ws] continue=false Dialplan: sofia/external/[callernumber](a)sip.[xyz].com Regex (FAIL) [bbb_conferences_ws] ${bbb_authorized}() =~ /true/ break=on-false Dialplan: sofia/external/[callernumber](a)sip.[xyz].com parsing [default- >bbb_conferences] continue=false Dialplan: sofia/external/[callernumber](a)sip.[xyz].com Regex (FAIL) [bbb_conferences] ${bbb_authorized}() =~ /true/ break=on-false Dialplan: sofia/external/[callernumber](a)sip.[xyz].com parsing [default- >bbb_echo_test_direct] continue=false Dialplan: sofia/external/[callernumber](a)sip.[xyz].com Regex (FAIL) [bbb_echo_test_direct] ${bbb_authorized}() =~ /true/ break=on-false Dialplan: sofia/external/[callernumber](a)sip.[xyz].com parsing [default- >ECHO_TO_CONFERENCE] continue=false Dialplan: sofia/external/[callernumber](a)sip.[xyz].com Regex (FAIL) [ECHO_TO_CONFERENCE] ${bbb_from_echo}() =~ /true/ break=on-false Ending up in: 2020-12-11 10:16:01.332329 [INFO] switch_core_state_machine.c:312 No Route, Aborting With a final statement: 2020-12-11 10:16:01.332329 [NOTICE] switch_core_state_machine.c:313 Hangup sofia/external/[callernumber](a)sip.[xyz].com [CS_ROUTING] [NO_ROUTE_DESTINATION] I understand that the Regex checks try to match the extension names in the three files in /opt/freeswitch/conf/dialplan/default. Probably, the call should match with "bbb_conferences", where the Regex fails with "not authorized". This is strange, because I am explicitly using an "authorized" statement in my dialin.xml config file: >>>>> <extension name="check_if_conference_active"> <condition field="${conference ${pin} list}" expression="/sofia/g" /> <condition field="destination_number" expression="^SEND_TO_CONFERENCE$"> <action application="set" data="bbb_authorized=true"/> <action application="transfer" data="${pin} XML default"/> </condition> </extension> <<<<< I think I fail to understand something right in front of me :-) Please help. Thanks a lot in advance! BR Markus

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How to modify 'Allow:' in SIP header
by Adrian Worutowicz 11 Dec '20

11 Dec '20

FS sends : [Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, PRACK, NOTIFY] Is it possible to stop FS sending INFO ? If so, how ? In advance thank you for any tips.

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AWS EC2 setup using sometimes internal, sometimes external IP for host candidates
by Sven Peters 07 Dec '20

07 Dec '20

Hi, I have followed Amazon EC2 <https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2?src=context…> to setup latest Freeswitch on an EC2 instance with an internal and public IP. When I dial out to an external SIP endpoint I can see sometimes the internal IP in the candidates list, sometimes the external IP - same configuration, just retrying the same endpoint. Internal IP: a=candidate:5446536444 1 udp 659136 172.25.36.61 29668 typ host generation External IP: a=candidate:1181434720 1 udp 659136 13.238.218.223 20132 typ host generation 0 How can I debug this and does anybody have an idea why this is inconsistent with the same config when redialing same endpoint? Thanks for your help Sven

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HELP DTLS HANDSHAKE to FAIL chromeV87
by Jazmin Marina Florez Marino 07 Dec '20

07 Dec '20

Hi I have a verto client trying to connect to FS using Chrome V87 but we have DTLS problems. The openssl version we are using on FS side is OpenSSL 1.0.2k-fips FreeSWITCH (Version 1.8.4 64bit) FS LOG - Chrome V87 FAILED CONNECTION 2020-12-03 11:23:06.762291 [ERR] switch_rtp.c:3199 video Handshake failure 1 2020-12-03 11:23:06.762291 [INFO] switch_rtp.c:3200 Changing video DTLS state from HANDSHAKE to FAIL 2020-12-03 11:23:06.832295 [ERR] switch_rtp.c:3199 audio Handshake failure 1 2020-12-03 11:23:06.832295 [INFO] switch_rtp.c:3200 Changing audio DTLS state from HANDSHAKE to FAIL 2020-12-03 11:23:06.852295 [NOTICE] switch_rtp.c:3181 Hangup verto.rtc/3520 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] 2020-12-03 11:23:06.852295 [DEBUG] switch_core_media.c:7470 verto.rtc/3520 Video thread ended 2020-12-03 11:23:06.872305 [INFO] conference_loop.c:1670 Channel leaving conference, cause: DESTINATION_OUT_OF_ORDER 2020-12-03 11:23:06.872305 [DEBUG] switch_core_session.c:2639 verto.rtc/3520 skip receive message [DISPLAY] (channel is hungup already) 2020-12-03 11:23:06.872305 [DEBUG] mod_conference.c:2467 verto.rtc/3520 skip receive message [TRANSFER] (channel is hungup already) 2020-12-03 11:23:06.872305 [DEBUG] switch_core_media.c:12220 verto.rtc/3520 skip receive message [BITRATE_REQ] (channel is hungup already) 2020-12-03 11:23:06.872305 [DEBUG] switch_core_codec.c:248 verto.rtc/3520 Restore previous codec opus:116. 2020-12-03 11:23:06.872305 [DEBUG] switch_core_session.c:2886 verto.rtc/3520 skip receive message [PHONE_EVENT] (channel is hungup already) 2020-12-03 11:23:06.872305 [DEBUG] switch_core_state_machine.c:650 (verto.rtc/3520) State EXECUTE going to sleep 2020-12-03 11:23:06.872305 [DEBUG] switch_core_state_machine.c:584 (verto.rtc/3520) Running State Change CS_HANGUP (Cur 1 Tot 3) 2020-12-03 11:23:06.872305 [DEBUG] switch_core_state_machine.c:847 (verto.rtc/3520) Callstate Change ACTIVE -> HANGUP 2020-12-03 11:23:06.872305 [DEBUG] switch_core_state_machine.c:849 (verto.rtc/3520) State HANGUP 2020-12-03 11:23:06.872305 [DEBUG] switch_core_state_machine.c:60 verto.rtc/3520 Standard HANGUP, cause: DESTINATION_OUT_OF_ORDER 2020-12-03 11:23:06.872305 [DEBUG] switch_core_state_machine.c:849 (verto.rtc/3520) State HANGUP going to sleep The weird thing is the connection works ok when the version of chrome is V86. Some weeks ago chrome was updated and we detected this issue. FS LOG - ChromeV86 CONNECTION OK 2020-12-03 11:19:19.622294 [INFO] switch_rtp.c:3206 Changing video DTLS state from HANDSHAKE to SETUP 2020-12-03 11:19:19.672317 [INFO] switch_rtp.c:3113 video Fingerprint Verified. 2020-12-03 11:19:19.672317 [INFO] switch_rtp.c:4104 Activating video Secure RTP SEND 2020-12-03 11:19:19.672317 [INFO] switch_rtp.c:4082 Activating video Secure RTP RECV 2020-12-03 11:19:19.672317 [DEBUG] switch_core_sqldb.c:2617 Secure Type: srtp:dtls:AES_CM_128_HMAC_SHA1_80 2020-12-03 11:19:19.672317 [INFO] switch_rtp.c:3155 Changing video DTLS state from SETUP to READY 2020-12-03 11:19:19.672317 [DEBUG] switch_rtp.c:1890 rtcp_stats_init: video ssrc[2719546543] base_seq[1999] 2020-12-03 11:19:19.672317 [DEBUG] switch_core_sqldb.c:2617 Secure Type: srtp:dtls:AES_CM_128_HMAC_SHA1_80 2020-12-03 11:19:19.702298 [INFO] switch_rtp.c:3206 Changing audio DTLS state from HANDSHAKE to SETUP 2020-12-03 11:19:19.702298 [INFO] switch_rtp.c:3113 audio Fingerprint Verified. 2020-12-03 11:19:19.702298 [INFO] switch_rtp.c:4104 Activating audio Secure RTP SEND 2020-12-03 11:19:19.702298 [INFO] switch_rtp.c:4082 Activating audio Secure RTP RECV 2020-12-03 11:19:19.702298 [DEBUG] switch_core_sqldb.c:2617 Secure Type: srtp:dtls:AES_CM_128_HMAC_SHA1_80 2020-12-03 11:19:19.702298 [INFO] switch_rtp.c:3155 Changing audio DTLS state from SETUP to READY 2020-12-03 11:19:19.702298 [DEBUG] switch_core_sqldb.c:2617 Secure Type: srtp:dtls:AES_CM_128_HMAC_SHA1_80

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freeswitch won't open file
by Joli Martinez 05 Dec '20

05 Dec '20

Hello, I have the following function that works normally on lua, but when I copied and pasted it into my freeswitch lua script, the file wont open. I am trying to put the passwords I use in a json format outside the script, so that I can share the script. I have installed the correct libraries for lunajson to work. Any suggestions? --Reads json file and parses it function returnJson(option) freeswitch.consoleLog("INFO", "Inside function\n") local file = io.open("password.json", "r") if not file then return nil end local jsonString = file:read("*a") freeswitch.consoleLog("INFO", "File is open\n") file:close() jsonResult = lunajson.decode( jsonString ) freeswitch.consoleLog("INFO", "Parsing json\n") return ( jsonResult[option] ) end --SID = returnJson("SID") --Token = returnJson("TOKEN") ------------------------------------------------------------------------------------------------------------ 2020-12-04 14:38:13.637108 [ERR] mod_lua.cpp:203 /usr/share/freeswitch/scripts/test.lua:67: attempt to concatenate global 'SID' (a nil value) stack traceback: /usr/share/freeswitch/scripts/test.lua:67: in main chunk

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please give user list
by Md Ahammad Muin 04 Dec '20

04 Dec '20

hello sir please give me a user list

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Re: [Freeswitch-users] Force Inband DTMF Reception (SOLVED)
by Adrian Worutowicz 03 Dec '20

03 Dec '20

Just for the records : In order to receive DTMS in-band for incoming calls, FS must not send in SDP : a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 This is achieved by : In vars.xml : <X-PRE-PROCESS cmd="set" data="rtp_liberal_dtmf=false"/> In sip_profiles\internal.xml and/or external.xml (Settings): Add line : <param name="dtmf-type" value="none"/> Remove line (if exists) : <param name="rfc2833-pt" value="101"/> Big thanks to @seven1240 ! De : FreeSWITCH-users <freeswitch-users-bounces(a)lists.freeswitch.org> De la part de Piotr Gregor Envoyé : lundi 30 novembre 2020 16:58 À : FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Objet : Re: [Freeswitch-users] Force Inband DTMF Reception Cześć Adrian, Tak, to ma sens. Spróbuj może ustawić flagę dtmf_type=none <action application="set" data="dtmf_type=none"/> To powinno sprawić, że w SDP ANSWER nie będzie a=rtpmap:101 telephone-event/8000, tylko nie jestem pewien czy detekcja in-band DTMF będzie włączona. pozdrawiam, Piotr On Mon, 30 Nov 2020 at 15:34, Adrian Worutowicz <adrian.worutowicz(a)esifrance.net<mailto:adrian.worutowicz@esifrance.net>> wrote: Witaj Piotrze, Co za niespodzianka! 😊 Otóż faktycznie INVITE zawiera: v=0 o=BroadWorks 210259115 1 IN IP4 xxx.xxx.xxx.xxx s=- c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=audio 62810 RTP/AVP 8 18 101 a=fmtp:18 annexb=no a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=ptime:20 Zaś SIP/2.0 200 OK zawiera: v=0 o=FreeSWITCH 1606729402 1606729403 IN IP4 xxx.xxx.xxx.xxx s=FreeSWITCH c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=audio 20616 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 Gateway to ORANGE, także mam tu niewielkie pole manewru. Powiedzieli mi jednak, że jeśli usunę a=rtpmap:101 telephone-event/8000, to będzie inband – co jest możliwe, bo w tym połączeniu proponują kodek G729 a ja im na to PCMA no i transmisja jest w PCMA (G711A). Także pozostaje pytanie, jak usunąć tę linię z sekcji SDP wysyłanej przez FS w 200 OK. Serdecznie pozdrawiam, Adrian. De : FreeSWITCH-users <freeswitch-users-bounces(a)lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>> De la part de Piotr Gregor Envoyé : lundi 30 novembre 2020 11:58 À : FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>> Objet : Re: [Freeswitch-users] Force Inband DTMF Reception Cześć Adrian, Tryb w którym FS będzie używał DTMF zależy od trybu który jest oferowany w SDP INVITE. Zazwyczaj klient jest konfigurowalny, np. w Bria: Preferences -> Calls -> DTMF: Send in-band. Tak więc upewnij się, że to nie klient żąda RFC 2833. pozdrawiam serdecznie, Piotr [https://drive.google.com/a/dataandsignal.com/uc?id=1lUdzdLjEycxXcGhDz-bKqDX…] Piotr Gregor Software Engineer M: (+44) 07483 866 525 L: (+44) 01256 597 470 www: dataandsignal.com<http://dataandsignal.com> On Thu, Nov 26, 2020 at 6:03 PM Adrian Worutowicz <adrian.worutowicz(a)esifrance.net<mailto:adrian.worutowicz@esifrance.net>> wrote: I tried in the SIP profile: <param name="dtmf-type" value="none"/> And in dialplan default.xml / <extension name I put <action application="set" data="dtmf_type=none"/> and also I tried <action application="bridge" data="{dtmf_type=none} … None of those worked. De : FreeSWITCH-users <freeswitch-users-bounces(a)lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>> De la part de Dragos Oancea Envoyé : jeudi 26 novembre 2020 18:10 À : FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>> Objet : Re: [Freeswitch-users] Force Inband DTMF Reception How about if you set channel variable "dtmf_type" to value "none" ? On Thu, Nov 26, 2020 at 5:59 PM Adrian Worutowicz <adrian.worutowicz(a)esifrance.net<mailto:adrian.worutowicz@esifrance.net>> wrote: Hello, For incoming calls, I want to receive DTMF inband. FS receives INVITE and at certain stage sends 200 OK with a SDP section, which always contains : a=rtpmap:101 telephone-event/8000 (what means ‘DTMF in RTP metadata’). Indeed the DTMFs come via RTP. I tried various FS configuration options to remove that line, without success. How to do it? How to tell a gateway to send DTMFs inband ? In advance thank you very, very much, Adrian. _________________________________________________________________________ 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<mailto:sales@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<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com _________________________________________________________________________ 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<mailto:sales@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<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com _________________________________________________________________________ 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<mailto:sales@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<mailto:FreeSWITCH-users@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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