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mod_com_amd
by Greg Doermann 26 Mar '18

26 Mar '18

We have been using waitforresult to try and wait for AMD to come back with something but we are having issues with it hanging (never returning from that) and locking up a license). We have also tried using the amd_execute_on_person command as it doesn't seem to be working. We originally had the script setup to run amd_execute_on_person but that was never hit and we would end up just waiting the full timeout each time before checking the result: (amd.lua) dst_number = session:getVariable("sip_to_user") > initial_wait = session:getVariable("ppamd_initial_wait") > max_wait = session:getVariable("ppamd_max_wait") > if not max_wait then > max_wait = 3000 > end > while (session:ready() and not session:answered()) do > -- Waiting for answer. > session:sleep(500) > -- freeswitch.msleep(500) > end > function on_finished(reason) > if reason == nil then reason = "unspecified" end > session:setVariable("ppamd_reason", reason) > session:execute("transfer", "process_amd XML default") > end > if session:ready() and session:answered() then > freeswitch.consoleLog("INFO", string.format("AMD Enable on %s.\n", > dst_number)) > if initial_wait then > session:sleep(initial_wait) > end > local use_amd = api:executeString("amd_available") > if use_amd == "true" then > -- Be sure to set these variables. They will stop the wait and > immediatly transfer the call if it gets a hit > session:setVariable("amd_execute_on_machine", "transfer > amd_machine_detected XML default") > session:setVariable("amd_execute_on_person", "transfer > amd_person_detected XML default") > session:setVariable("amd_execute_on_unsure", "transfer > amd_unsure_detected XML default") > session:execute("voice_start") > -- Giving some time to AMD to work on the call. > -- If a voicemail is detected, it will auto-transfer from the > variables that are set. > session:sleep(max_wait) > local amd_finished_called = > session:getVariable("amd_finished_called") > if amd_finished_called == 'true' then > freeswitch.consoleLog("INFO", string.format("AMD Finish script > already called!\n")) > else > -- NOTE: Hopefully this never gets called as the module should > auto transfer this > freeswitch.consoleLog("INFO", string.format("AMD Timeout > reached for vmd: %s\n", max_wait)) > session:execute("voice_stop") > local amd_detect = session:getVariable("amd_status") > freeswitch.consoleLog("INFO", string.format("AMD amd_status: > %s\n", amd_detect)) > on_finished(amd_detect) > end > else > freeswitch.consoleLog("WARNING", string.format("AMD not available > for %s\n", dst_number)) > on_finished("unavailable") > end > else > freeswitch.consoleLog("WARNING", string.format("AMD Failed to run > %s\n", dst_number)) > on_finished("failure") > return > end the amd_execute_on_XXX does not seem to do anything... The sleep would always max out and we would always hit the Timeout reached part of the script instead of getting transferred. After that didn't work because we would always wait the sleep time (3 seconds) we scrapped transfers and tried handling the whole thing through a single lua script using waitforresult: DEFAULT_MAX_WAIT = 3000 -- in (ms) > DEFAULT_INITIAL_WAIT = 100 -- in (ms) > HUMAN_EXTENSION = "amd_queue" > UNSURE_EXTENSION = "amd_queue" > api = freeswitch.API() > > dst_number = session:getVariable("sip_to_user") > initial_wait = session:getVariable("ppamd_initial_wait") or > DEFAULT_INITIAL_WAIT > max_wait = session:getVariable("ppamd_max_wait") or DEFAULT_MAX_WAIT > > while (session:ready() and not session:answered()) do > session:sleep(500) > end > > freeswitch.consoleLog("INFO", string.format("AMD Attempting on %s. \n", > dst_number)) > if initial_wait then > session:sleep(initial_wait) > end > > if session:ready() and session:answered() then > local use_amd = api:executeString("amd_available") > if use_amd == "true" then > freeswitch.consoleLog("INFO", string.format("AMD Available Max > Wait: %s, Initial Wait %s. \n", max_wait, initial_wait)) > -- Allow AMD to run and analyze. > freeswitch.consoleLog("INFO", "AMD Voice start. \n") > session:execute("voice_start") > freeswitch.consoleLog("INFO", "AMD Wait for result. \n") > session:execute("waitforresult", string.format("%s", max_wait)) > freeswitch.consoleLog("INFO", "AMD Voice stop. \n") > session:execute("voice_stop") > amd_detect = session:getVariable("amd_status") > freeswitch.consoleLog("INFO", string.format("AMD Result: %s. \n", > amd_detect)) > if amd_detect == "machine" then > freeswitch.consoleLog("INFO", "AMD Detected machine. \n") > session:hangup() > return > elseif amd_detect == "person" then > freeswitch.consoleLog("INFO", "AMD Detected human, transfering > call. \n") > session:execute("transfer", string.format("%s XML default", > HUMAN_EXTENSION)) > return > else > freeswitch.consoleLog("INFO", string.format("AMD Unknown: %s. > \n", amd_detect)) > session:execute("transfer", string.format("%s XML default", > UNSURE_EXTENSION)) > return > end > end > else > freeswitch.consoleLog("INFO", "AMD not enabled. \n") > session:execute("transfer", string.format("%s XML default", > UNSURE_EXTENSION)) > return > end I prefer not having to do all that transferring around but to just run with waitforresult but it never comes back... The dialplan extensions just connect the call to an agent. Once we do the transfer it all works. The only problem is we have to replace the: session:execute("waitforresult", string.format("%s", max_wait)) with a sleep so no matter if the mod_com_amd detects its a person or not we ALWAYS have to wait the full sleep time. This is not really legal as we would be essentially abandoning every single call... Any thoughts on what we may be missing or how we could improve this?

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Presense, FS as client
by Sergey Safarov 25 Mar '18

25 Mar '18

Hello I want subscribe to BLF events of several extension on remote FS server. I configured gateway with subscription as it described "Presence <https://wiki.freeswitch.org/wiki/Presence#Use_FreeSWITCH_as_a_client>" page. My FS servers is subscribed to extension same as gateway username. Is possible to define "To" header in subscription other than gateway username?

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Subscribing to ESL Events 'dynamically' per channel
by Joshua Gigg 21 Mar '18

21 Mar '18

Hi, We're using the inbound ESL socket to subscribe to certain events. However, we receive these events for every call. Is it possible to dynamically subscribe to events per call via the API? We're hoping to add/remove certain events at certain points during our call flows to avoid unnecessary noise in the events received.

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Change caller ID
by Caio Assis 21 Mar '18

21 Mar '18

Hello. I'm trying to set a different caller id, but it's not working. It show '00000' no matter what parameter or variable I change. I've tried to change effective_caller_id_name/number directly on the sip account parameters, I've tried to change it in the dialplan, but nothing seems to work. Thanks in advance.

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183 early media" followed by 180 Ringing From Mobile Carriers
by Jonathan Hunter 21 Mar '18

21 Mar '18

Hi Guys, Sorry to bring this up again but I see it when performing interopt testing with mobile carriers. A number of them due to ISUP interworking, have gateways that generate a 183 with SDP, then a 180ringing. I understand from ready for example this jira case FreeSWITCHFS-3859 SOFIA "183 early media" followed by "180 Ringing" where this is done by design. I just wondered if this position had changed and if there is a configuration parameter/option that can allow any early media message to be sent back to A leg by freeswitch, in particular a 183 with SDP, then a subsequent 180 ringing, as carriers tend to want to see both. Many thanks Jon

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Ghost Ringing from Verto Client when connection interrupted
by José Lopes 20 Mar '18

20 Mar '18

Hello, I am using Freeswitch as WebRTC Gateway with mod_verto and mod_sofia. When some verto client suspend its laptop, the verto contact was keept on verto status (fs_cli -x 'verto status') . When verto client suspend its laptop, the connection is interrupted, the verto client didn't close the connection. When some SIP User make a call to that verto client, the call is ringing (It receives SIP Ringing). But the call never arrive to verto client. There is any way to avoid the SIP Ringing to SIP User, since this cause the feeling that it is ringing on verto client? Thanks in advance. Best Regards, Jose Lopes

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sched_api conference call with flags
by Yuriy Gorlichenko 19 Mar '18

19 Mar '18

Hi all I trying to call conference via sched_api but can't find any way to send flags to the conference ${sched_api +1 none conference $1-${domain} play file_string://${namefile} } so ${sched_api +1 none conference $1-${domain}+${flags} play file_string://${namefile} } cant find conference with searched name ${sched_api +1 none conference $1-${domain} ${flags} play file_string://${namefile}} Says that ${flags} is can not be used here So is here any way to call conference with flags?

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Transcoding Performance OPUS <-> G722
by Alexander Haugg 15 Mar '18

15 Mar '18

Hi, at the moment we are testing the Freewitch performance. The Freeswitch run on Windows Server 2012 R2 on a Hyper V The virtual maschine have an Intel Xeon 2,2 GHz CPU wiith 8 cores ( = max CPU performance 800%) Here some results: - 5000 SIP Registrations (reregister every 2 minutes) - 500 parallel calls (1000 channels) SRTP Opus without transcoding - Freeswitch CPU performance – 67% of possible 800% - That is a very good result! Now the transcoding part SRTP OPUS <-> RTP G711 - 500 SIP Registrations - 25 parallel calls (50 channels) SRTP OPUS <-> RTP G711 - Freeswitch CPU performance – 126% of possible 800% - 50 parallel calls (100 channels) SRTP OPUS <-> RTP G711 - Freeswitch CPU performance – 280% of possible 800% What is your experience? Are the transconding results ok, or to much?

1 0

Multi tenant / user 1 in domain A can not reach user 2 in domain B
by Thorsten Göllner 15 Mar '18

15 Mar '18

Hi, I setup freeswitch with fusionpbx in a multi tenant evironment. Works fine. But currently I can not call "cross domain". So user 1 in domain A can not reach user 2 in domain B. When calling my log show the following: 2018-03-14 12:37:39.580028 [NOTICE] switch_channel.c:1104 New Channel sofia/internal/1@domain-A [19bbf41c-277c-11e8-81fa-4dba4b7d93a7] 2018-03-14 12:37:39.580028 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/1@domain-A) Running State Change CS_NEW (Cur 1 Tot 84) 2018-03-14 12:37:39.580028 [DEBUG] sofia.c:9873 sofia/internal/1@domain-A receiving invite from 1.2.3.4:54948 version: 1.6.19 64bit 2018-03-14 12:37:39.580028 [DEBUG] sofia.c:10044 IP 1.2.3.4 Rejected by acl "domains". Falling back to Digest auth. 2018-03-14 12:37:39.580028 [WARNING] sofia_reg.c:1792 SIP auth challenge (INVITE) on sofia profile 'internal' for [2@domain-B] from ip 1.2.3.4 2018-03-14 12:37:39.580028 [DEBUG] switch_core_state_machine.c:603 (sofia/internal/1@domain-A) State NEW 2018-03-14 12:37:39.580028 [DEBUG] sofia.c:2334 detaching session 19bbf41c-277c-11e8-81fa-4dba4b7d93a7 2018-03-14 12:37:39.619980 [DEBUG] sofia.c:2442 Re-attaching to session 19bbf41c-277c-11e8-81fa-4dba4b7d93a7 2018-03-14 12:37:39.619980 [DEBUG] sofia.c:9873 sofia/internal/1@domain-A receiving invite from 1.2.3.4:54948 version: 1.6.19 64bit 2018-03-14 12:37:39.619980 [DEBUG] sofia.c:10044 IP 1.2.3.4 Rejected by acl "domains". Falling back to Digest auth. 2018-03-14 12:37:39.639961 [DEBUG] freeswitch_lua.cpp:365 DBH handle 0x7fdb641160c0 Connected. 2018-03-14 12:37:39.639961 [DEBUG] freeswitch_lua.cpp:382 DBH handle 0x7fdb641160c0 released. 2018-03-14 12:37:39.639961 [WARNING] sofia_reg.c:2906 Can't find user [12(a)telequest.qcall.cloud] from 1.2.3.4 You must define a domain called 'telequest.qcall.cloud' in your directory and add a user with the id="12" attribute and you must configure your device to use the proper domain in it's authentication credentials. 2018-03-14 12:37:39.639961 [WARNING] sofia_reg.c:1737 SIP auth failure (INVITE) on sofia profile 'internal' for [2@domain-B] from ip 1.2.3.4 Is there a special setting or configuration I missed? Is it a ACL issue? Thanks in advance, Thorsten

2 1

Verto - audio only between FS and client
by Rick Jarvis 14 Mar '18

14 Mar '18

What is the best way to remove all call control abilities from the web client, and have just audio between FS and browser? Let me explain what I mean and why: My app server (running on Node) controls all call handling by ESL to FreeSWITCH. As the service is multi-tenant I need to consider security, but all the multi-tenant stuff is on the app server, and I’m trying to avoid the complexities of having multiple domains etc on FS as I would then need to build a way of syncing this with the app server. So, ideally, the browser would connect to FS (via Verto, as I think it is Verto that handles all the clever re-connection stuff?), and would share its call uuid with the app server, which would then handle all the call transfer / conferencing etc using ESL to FS. The sticking point is that I want to make sure that a user can’t instruct Verto to transfer the call to a guessed conference name. At the moment, I have a rather elaborate system of using a randomly generated auth code, which the app server keeps in Redis, and adds to the name of the conference. But the complexities of this are proving problematic in my code. I hope this makes sense, I definitely consider myself to still be very new to webrtc/verto etc, despite having been playing with it for many months now! Hoping for some ideas or guidance. Thanks R

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