December 2020 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Can't test for empty variable
by David Villasmil 16 Dec '20

16 Dec '20

Hello all, I'm trying to test the play_and_get_digits variable from here: <action application="play_and_get_digits" data="1 1 1 3000 # '' silence_stream://250 dtmf_get \d 3000"/> Like this: <condition field="${dtmf_get}" expression="^$"> But it's always false... How can I test for an empty (null?) variable? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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Copy incoming audio from participants for ASR
by R G 15 Dec '20

15 Dec '20

Hello everyone, i have a technical question about freeswitch and would like to know if my idea is technically possible. I've got a project at the university and my goal is to improve automatic speech recognition for bigbluebutton. In the summer semester a group of students of my university made a subtitling plugin[1] for bigbluebutton. This plugin adds a participant into the meeting and grabs his audiostream to send it to the kaldi ASR Software. When more than one person speaks at a time the detection decreases. My idea is to copy the incoming freeswitch audiostreams from every participant and send them directly as a copy to the ASR (kaldi can handle multiple audiostreams at once). In the Confluence-Wiki i searched for something in this direction but only found things like Dialplan to ring multiple telephones at once. I hope my idea is formulated understandably. Kind regards, Robert [1] https://github.com/3wille/bbb-kaldi-connector

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NORMAL_TEMPORARY_FAILURE Suddenly
by Don Hawkins 14 Dec '20

14 Dec '20

Hi everyone! Suddenly all outbound calls (ALL of our providers) are returning NORMAL_TEMPORARY_FAILURE on our FS server. I've been working on this for roughly four hours so I wanted to reach out. It does seem that at least with one provider the calls are getting to their servers since they show as "Failed" in their management portal. We've changed nothing, suddenly origination just stopped working. sofia/gateway/*Provider*/12145551212%20&park() Any input is appreciated, I know it's something simple, just need to be guided in the right direction. Thanks! Don

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Re: [Freeswitch-users] NORMAL_TEMPORARY_FAILURE Suddenly
by Don Hawkins 14 Dec '20

14 Dec '20

Kaiduan, That's the termination provider's IP address.

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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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