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macOS Wiki - Wow 10 years, already!
by mario_fs 27 Nov '20

27 Nov '20

For macOS users and anyone interested: Nov 22 2020 is the tenth anniversary of when I put up the first complete install guide for macOS on the wiki <https://freeswitch.org/confluence/display/FREESWITCH/macOS>, boy does that make me feel old. I realized this yesterday while doing my weekly FreeSwitch Development build/test for the macOS wiki. I know some have found it useful since I've seen it referenced on the mailing list, some info like emailing voicemail may be useful to Linux users. I also tried helping with the Linux stuff but that didn't last. Mario G

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Timer D (32 seconds) being triggered after ACK in TCP INVITE Transaction
by João Vitor Arruda 26 Nov '20

26 Nov '20

Hello, I have a scenario where freeswitch bridges a call received with TCP on leg A to a destination with TCP on leg B When the destination rejects the call with 486 (Busy Here) that is sent back to leg A which receives the ACK as expected. So from the leg A perspective we have: INVITE 100 Trying 180 Ringing 486 Busy Here ACK After the ACK freeswitch seems to be starting the Timer D (32 seconds) even for that TCP transaction. This is what the log show when the ACK is received: tport.c:2771 tport_wakeup() tport_wakeup(0x561ffce1c3e0): events IN tport.c:2868 tport_recv_event() tport_recv_event(0x561ffce1c3e0) tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x561ffce1c3e0) msg 0x561ffcf2f700 from (tcp/192.168.100.100:34863) has 394 bytes, veclen = 1 tport.c:3023 tport_deliver() tport_deliver(0x561ffce1c3e0): msg 0x561ffcf2f700 (394 bytes) from tcp/192.168.100.100:34863/sip next=0 nta.c:2877 agent_recv_request() nta: received ACK sip:6111@172.16.100.1;transport=tcp SIP/2.0 (CSeq 1) nta.c:3017 agent_recv_request() nta: ACK (1) is going to INVITE (1) tport.c:2298 tport_set_secondary_timer() tport(0x561ffce1c3e0): reset timer 2020-11-26 17:37:16.622026 [DEBUG] switch_scheduler.c:144 Deleting task 292 switch_ivr_schedule_hangup (372ca1ff-4466-48c5-a5a0-a22612a217ed) then 32 seconds later nta.c:9106 outgoing_timer_dk() nta: timer D fired, terminate INVITE (28652323) tport.c:2298 tport_set_secondary_timer() tport(0x561ffcf30a80): reset timer nta.c:8804 outgoing_reclaim_queued() outgoing_reclaim_all(0, 0, 0x7f52bceb8958) nta.c:8927 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free tport.c:2298 tport_set_secondary_timer() tport(0x561ffce1c3e0): reset timer nta.c:5824 incoming_reclaim_queued() incoming_reclaim_all(0, 0, 0x7f52bceb8958) nta.c:7186 _nta_incoming_timer() nta_incoming_timer: 0/0 resent, 0/0 tout, 0/0 term, 1/1 free nta.c:1289 agent_timer() nta: timer not set Based on the RFC 3261 section-17.1.1.2 ( https://tools.ietf.org/html/rfc3261#section-17.1.1.2) it seems that Timer D should zero seconds for reliable transport (TCP): The ACK MUST be sent to the same address, port, and transport to which the original request was sent. The client transaction SHOULD start timer D when it enters the "Completed" state, with a value of at least 32 seconds for unreliable transports, and a value of zero seconds for reliable transports Timer D reflects the amount of time that the server transaction can remain in the "Completed" state when unreliable transports are used. Should this be considered a BUG?

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sip Allow headers differents between 2 legs
by Mickael Hubert 26 Nov '20

26 Nov '20

Hi all, FS receives on its A-leg this SIP Allow header: Allow: OPTIONS, PUBLISH, INVITE, ACK, BYE, CANCEL But when it sends INVITE to B-leg, it adds others parameters: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY I saw that it's not possible to modify Allow header with FS. Is there a way to copy exactly the same Allow header between 2 legs ? thanks in advance ++

1 0

Help my son, answer a poll on Libra/CryptoCurrencies/Facebook
by Giovanni Maruzzelli 26 Nov '20

26 Nov '20

Hey FreeSWITCHers! Please take 50 seconds to check some radiobuttons on this anonymous poll: https://docs.google.com/forms/d/e/1FAIpQLSdPBoZTm_yCBRJB0lkQxqSa8eJrRAaM9pS… My son Rocco is writing a paper about Libra (FaceBook cryptocurrency) acceptance and privacy concerns. You'll be proud of yourself ! And I'll be very glad/grateful :)) -giovanni -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18

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hangup from LUA script
by Gandalf Corvotempesta 25 Nov '20

25 Nov '20

Hi guys, i'm trying to make a very simple IVR with a lua script. I'm calling the lua script in this way: <action application="play_and_get_digits" data="6 12 10 7000 # say:'Please inser order number.' silence_stream://250 order_number \d+" /> <action application="lua" data="myivr.lua"/> And myivr.lua has something like this: ---------------------------------------------------------------- if session:getVariable("order_number") ~= "1234567" then session:speak("Error"); session:hangup("NO_ROUTE_DESTINATION") end session:speak("connecting to customer. Please wait"); session:setAutoHangup(false); session:execute("bridge", "{ignore_early_media=true,bridge_generate_comfort_noise=true,effective_caller_id=anonymous,hangup_after_bridge=true}sofia/gateway/mygateway/123456789"); -------------------------------------------------------------------- Now, the questions: 1) Biggest issue is that when I call session:hangup() , the call is still running, how can I force hangup ? 2) as this lua script is called after a "play_and_get_digits" (in dialplan), how can I replace the "Error" inside IF with something that will re-execute the dialplan from scratch asking for a new number ? In other words, if numbers are not "1234567", I have to ask for the numbers again. 3) how can I set the bridge to be anonymous on the remote phone ? I don't want the remote user to be able to see the caller number. It would be better if remote user will se MY number and not the origin number (it's a bridge: ORIGIN -> PBX -> REMOTE, the remote will see origin number. I want to be anonymous or PBX number) Tnx in advance.

2 2

SOFIA channel for Outgoing calls
by Vishal Kakkar 25 Nov '20

25 Nov '20

Hi all, Using a SIP trunk with 30 channels. Is there a way in sofia, to ensure all channels are used equally for *outgoing* *calls*? Like a roundrobin scheme to ensure the load is distributed across all 30 channels in a single SIP trunk. Thanks, Vishal.

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mod_event_socket: command to play and collect DTMF
by Dmitry Mordovin 24 Nov '20

24 Nov '20

Hello I try to solve task: play a prompt and collect DTMF digits. Now I use Lua script and session:playAndGetDigits to did it. How can I do it thru mod_event_socket commands? I know uuid_broadcast command can play prompt, but I can’t find command to wait some DTMF digits. Any one can direct me to right simple solution? PS: one idea, it can be done if I can run same Lua script from mod_event_socket

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Trickle ICE support
by Ramesh Kandasamy 24 Nov '20

24 Nov '20

Hi all, Does FS support the Trickle ICE feature yet? If yes, can someone please point me a link on the documentation for the same? Thanks Ramesh

1 0

Sched_api Tasks List
by edson＠inoutglobal.xyz 24 Nov '20

24 Nov '20

Dears, Is it possible to list all scheduled tasks generated by sched_api command? Thanks, Edson

2 1

US local DID provider recommendations
by Chad Phillips 24 Nov '20

24 Nov '20

I'm on the hunt for a secondary DID provider, and would love any recommendations. Requirements: - Local DIDs in the continental US, with good coverage of the various NPA-NXXs - Per-minute billing, not per trunk - Some kind of server/IP/host failover configuration capability, in case my primary switch is down - Instant number provisioning - Supports SIP Redirects (surprisingly, 2 of my 3 current providers do not) Chad

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