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dtmf mode = none
by bugi＠nicelan.tk 26 Oct '16

26 Oct '16

Hello FS Experts, I need to disable dtmf tones on an active call in FS. My case scenario is: regular call is setup by caller -> FS -> outgoing trunk and dtmf codes should be working fine. After some time a dtmf blocking event will be send by lua script or FS command line. This should block sending dtmf codes for caller and makes them "invisible" on outgoing trunk. Is this possible to do this with FS? I was thinking that maybe I can change dtmf_mode during active call to none, but this doesn't seem to be working. Any ideas? Best wishes, bugi

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SRTP support
by Tom Chen 26 Oct '16

26 Oct '16

Hello, By default, Freeswitch seems always translates incoming SRTP packets to outbound RTP only packets as the INVITE SIP message to Receiver from FS contains only RTP/AVP audio media in SDP section even if the Caller's INVITE has RTP/SAVP audio media only. Caller to FS Session Initiation Protocol (INVITE) . . . . Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): 1001 625 1382 IN IP4 10.113.49.31 Session Attribute (a): rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics Media Description, name and address (m): audio 7078 RTP/SAVP 96 97 98 99 0 8 101 100 102 FS to Receiver Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): FreeSWITCH 1477472443 1477472444 IN IP4 172.17.10.23 Media Description, name and address (m): audio 25206 RTP/AVP 0 8 101 13 Is there a way to let FS send SRTP packets to receiver as the receiver can accept SRTP media only? Tom

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FS6947 - Tuning Opus bandwidth
by Emrah 26 Oct '16

26 Oct '16

Hi list, I re-read FS6947 and don't understand how this problematic was addressed and the issue fixed. The scope is simple. There should be a setting in the dialplan that allows downsampling of Opus for applications that do not require the 48khz / 2 channels framework. I.e.: terminating to the PSTN with Opus to take advantage of low bandwidth and great PLC. There seems to be a lot of confusion around bandwidth in general there. It doesn't matter if the internal clock of the device is always sampling at 48khz / 2ch. There are settings that can facilitate a lower bandwidth consumption for particular use cases, and it seems the reason it is not being implemented in FS is just a matter of being confused about the intent of the 48khz 2ch base. Please revisit this issue. FS should allow tuning of Opus audio / network bandwidth in the dialplan. It would optimize greatly lots of use cases. If I'm calling the PSTN, I'd rather have my client downsample and stream at a lower bandwidth, even if my audio capture would still be at 48khz / 2ch as per the RFC, and save on bandwidth, than transcode the full 48khz spectrum into PCM on my FS and minimize processing power on the client's side. Jira here: https://freeswitch.org/jira/browse/FS-6947 Emrah

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FreeSWITCH and Opus
by Giacomo Vacca 26 Oct '16

26 Oct '16

Hi everybody, we've added a Confluence page which focuses on Opus under several points of view, e.g. configuration, installation, development, debugging, negotiation. You can find it here: https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+And+The+Opu… Please take a look if you're interested and submit your feedback and questions. Thanks, Giacomo

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testing
by Ken Rice 26 Oct '16

26 Oct '16

This is only a test. you can ignore it (altho I know you want to reply to it)

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Drop requests
by Carlos Ruiz Díaz 25 Oct '16

25 Oct '16

Hi list, Is it possible to drop a request that doesn't match an ACL? The idea is not drop the SIP 403 that is replied by default. I know I can do this using a SIP proxy, or an iptables rule, but it would be best if FS offered this possibility too. Thanks, -- Carlos

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bypass_media=true, call terminate 480
by Aztrock 25 Oct '16

25 Oct '16

Hi inbound_bypass_media=true or bypass_media=true A<->Freeswitch<->B Pre-Answer: Ok MEDIA A<->B Answer:Failed, calls terminated inmediatly 2016-10-25 16:03:00.316897 [DEBUG] sofia.c:6965 Channel sofia/default/xxxx entering state [ready][200] 2016-10-25 16:03:00.316897 [NOTICE] sofia.c:7964 Channel [sofia/default/xxxx] has been answered 2016-10-25 16:03:00.316897 [DEBUG] switch_ivr_originate.c:3686 Originate Resulted in Success: [sofia/default/xxxx] 2016-10-25 16:03:00.316897 [DEBUG] switch_channel.c:3770 (sofia/default/xxxx) Callstate Change EARLY -> ACTIVE 2016-10-25 16:03:00.316897 [DEBUG] switch_ivr_bridge.c:1334 (sofia/default/3238780(a)x.x.x.x) State Change CS_EXECUTE -> CS_HIBERNATE 2016-10-25 16:03:00.316897 [DEBUG] switch_ivr_bridge.c:1335 (sofia/default/xxxx) State Change CS_CONSUME_MEDIA -> CS_HIBERNATE 2016-10-25 16:03:00.316897 [DEBUG] switch_core_state_machine.c:584 (sofia/default/xxxx) Running State Change CS_HIBERNATE 2016-10-25 16:03:00.316897 [DEBUG] switch_core_state_machine.c:665 (sofia/default/xxxx) State HIBERNATE 2016-10-25 16:03:00.316897 [DEBUG] mod_sofia.c:180 sofia/default/xxxx SOFIA HIBERNATE 2016-10-25 16:03:00.316897 [DEBUG] switch_core_state_machine.c:420 sofia/default/xxxx Standard HIBERNATE 2016-10-25 16:03:00.316897 [DEBUG] switch_core_state_machine.c:665 (sofia/default/xxxx) State HIBERNATE going to sleep 2016-10-25 16:03:00.316897 [DEBUG] switch_cpp.cpp:1360 FIN: troncal del plan ... <------------ 2016-10-25 16:03:00.316897 [INFO] switch_cpp.cpp:1360 FIN LLAMADA -> Estoy terminando ahora ... <----------- 2016-10-25 16:03:00.316897 [DEBUG] switch_cpp.cpp:1360 Got value : originate_disposition : SUCCESS 2016-10-25 16:03:00.316897 [DEBUG] switch_cpp.cpp:1288 Hangup not custom 2016-10-25 16:03:00.316897 [DEBUG] switch_cpp.cpp:722 CoreSession::hangup -- Un hombre de carácter podrá ser derrotado pero jamás destruido. ERNEST HEMINGWAY.

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Starting moh outbound mod_event_socket
by Valter Nogueira 25 Oct '16

25 Oct '16

I set hold_music var to local_stream://test How could I, after answering a call, start music on hold via mod_event_socket? Thanks Valter

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Dangerous prototype question for Seven Du
by Christoph Russow 24 Oct '16

24 Oct '16

Hi, i've got a question for Seven Du. I was very impressed by your dangerous prototype on ClueCon 2016 and i'm wondering if you would share some more details about it. For example some background about the UI and how you executed the generated code within Freeswitch. Best Regards Christoph Russow -- ________________________________________________________________ EMTEX GmbH Christoph Russow Software Engineer Bischof-Otto-Weg 9 D-91086 Aurachtal Geschäftsführer: Markus Enzinger Tel. +49 9132 7490 0 Sitz der Gesellschaft: 91086 Aurachtal Fax. +49 9132 7490 900 Amtsgericht Fürth: HRB6804 ________________________________________________________________

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Coredump info
by Chris Mandra 24 Oct '16

24 Oct '16

Hi guys - quick question - Do I need to do anything special/specific to enable core dumps generated by default freeswitch? I'm working on a module and the trace info would be helpful for debugging. thanks! (and happy monday!) chris

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