November 2021 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Need help to setup stir shaken in Freeswitch
by Pete Kay 04 Aug '22

04 Aug '22

Hi Does anyone know how to configure Freeswitch for stir shaken? Could you give me some info or doc how to do it? Thanks, Pete

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log format change breaking fail2ban? percentage added to logs? how to revert?
by Jelle de Jong 06 May '22

06 May '22

Hello everybody, The log format seem to have changed, how can I get it back to what it used to without the percentage thing? A search on Google failed me to provide an answer... # fs_cli -x version FreeSWITCH Version 1.10.7-release-19-883d2cb662~64bit (-release-19-883d2cb662 64bit) # grep pref_line /etc/fail2ban/filter.d/freeswitch.conf _pref_line = ^%(__prefix_line)s(?:\d+-\d+-\d+ \d+:\d+:\d+\.\d+)? failregex = %(_pref_line)s \[WARNING\] sofia_reg\.c:\d+ SIP auth (failure|challenge) $(REGISTER|INVITE)$ on sofia profile \'[^']+\' for \[[^\]]*\] from ip <HOST>$ %(_pref_line)s \[WARNING\] sofia_reg\.c:\d+ Can't find user \[[^@]+@[^\]]+\] from <HOST>$ 2021-10-28 12:28:55.736515 99.20% [WARNING] sofia_reg.c:1806 SIP auth failure (REGISTER) on sofia profile 'internal' for [30640(a)82.94.245.xxx] from ip 217.138.220.158 2021-05-20 16:44:54.159814 [WARNING] sofia_reg.c:1740 SIP auth failure (REGISTER) on sofia profile 'internal' for [123456(a)82.94.245.xxx] from ip 193.107.216.182 # fgrep percentage /etc/freeswitch/ -R What is the use case of this new percentage in the log files? Kind regards, Jelle de Jong

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uuid_broadcast hang and start after the call hangup
by Dmitry Mordovin 06 May '22

06 May '22

Hello I have LUA script which start a call (new session generated) Code here new_session = freeswitch.Session(destination); if (new_session:ready()) then new_session_disp = new_session:getVariable("endpoint_disposition"); while(new_session:ready() and new_session_disp ~= "ANSWER") do <------>new_session_disp = new_session:getVariable("endpoint_disposition"); end api = freeswitch.API(); api:executeString("bgapi uuid_park " .. tostring(new_session.uuid)); end On answer event I park the call and script finished. I see single active call. Listing below. freeswitch@freeswitch43> show calls uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,presence_id,presence_data,accountcode,callstate,callee_name,callee_num,callee_direction,call_uuid,hostname,sent_callee_name,sent_callee_num,b_uuid,b_direction,b_created,b_created_epoch,b_name,b_state,b_cid_name,b_cid_num,b_ip_addr,b_dest,b_presence_id,b_presence_data,b_accountcode,b_callstate,b_callee_name,b_callee_num,b_callee_direction,b_sent_callee_name,b_sent_callee_num,call_created_epoch 3aa3a274-c23d-11eb-9150-e74dc4cdb20a,outbound,2021-05-31 18:22:59,1622485379,sofia/external/1000,CS_PARK,,0000000000,,1000,,,,ACTIVE,Outbound Call,1000,,3aa3a274-c23d-11eb-9150-e74dc4cdb20a,freeswitch43,,,,,,,,,,,,,,,,,,,,,, 1 total. After all I execute command from cli: Command looks like: uuid_broadcast 3aa3a274-c23d-11eb-9a50-e74dc4cdb20a lua::’next-script.lua' both In logs I see, command was sent to FS success, but not executed! Moveover, after call hangup, FS start execution of command but can’t coz call in terminate state. 2021-05-31 18:23:30.312133 [NOTICE] sofia.c:1079 Hangup sofia/external/1000 [CS_PARK] [NORMAL_CLEARING] 2021-05-31 18:23:30.312133 [DEBUG] switch_ivr.c:625 sofia/external/1000 Command Execute lua(next-script.lua) 2021-05-31 18:23:30.312133 [DEBUG] switch_core_session.c:2668 sofia/external/1000 ZOMBIE EXEC lua(next-script.lua) EXECUTE sofia/external/1000 lua(next-script.lua) What/why is ZOMBIE EXEC ? Could someone help me to find reason of strange behavior FS? Looks like the uuid_broadcast command stay in queueu and wait something… and start after call has hangup.

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RE - Internal Interface suddenly freezes
by Kakiman Expert 11 Apr '22

11 Apr '22

hello Did you find any solution to this issue ? I am in production with 1.10.5 version and since a week, I have the same issue. The internal profile is out of service and registration is impossible. and we don't have any answer to "sofia status profile internal" , the prompt in fs_cli keeps waiting. the only way to solve is to restart freeswitch thanks regards

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Sending announcements to all callers
by Support from NetworkedAudio LLC 21 Mar '22

21 Mar '22

I have a system where a user dials in, get authenticated and placed into a standard conference. Other local resources get added to the conferences as needed, and that is working well. For safety reasons, we need to play a single incoming PortAudio channel to all dialed in users. If I do an auto call or originate I can add a deaf member to each conference which is a loopback to an eavesdrop. This is messy but works. However, it uses a lot of resources and now my local people have the announcements twice, with delay. What I want is just to send audio out to those dialing in. I have tried sending eavesdropping just to the incoming call but it won’t let me then join a conference through the dialplan - processing stops at the eavesdrop. I see broadcast but can only send tones or saved files. Ideally the incoming external callers would hear the announcements and not the conferences themselves. I’d like to avoid loopback, too. Does anyone have ideas, please?

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Issue with Enterprise originate and 487 "ORIGINATOR_CANCEL".
by Jai Rangi 14 Jan '22

14 Jan '22

All, Running into a small issue with group calling, when the original caller hang up, phones dont show that call as missed call, cause of wrong reason code from freeswitch. Scenario: - Caller -> FS (Groupcall) to 101 and 102 - Phones ring - Caller hangup - Phones should see missed calls, - But free-switch send cancel with Reason: SIP;cause=200;text="Call completed elsewhere", where is should send Reason: SIP;cause=487;text="ORIGINATOR_CANCEL". Here is what my bridge data. bridge(<all_threads=true,absolute_codec_string=PCMU,PCMA,G729,group_confirm_file=playback /usr/local/freeswitch/sound/ivr/1443458342.wav,group_confirm_key=exec>{sip_cid_type=rpid,ignore_early_media=true,originate_timeout=30}[leg_timeout=20 ]user/101@192.168.5.50:_:[ignore_early_media=true,leg_timeout=20 ]user/ 102(a)192.168.5.50) If I send bridge(<all_threads=true,absolute_codec_string=PCMU,PCMA,G729,group_confirm_file=playback /usr/local/freeswitch/sound/ivr/1443458342.wav,group_confirm_key=exec>{sip_cid_type=rpid,ignore_early_media=true,originate_timeout=30}[leg_timeout=20 ]user/101(a)192.168.5.50 Everything works fine. I see some old posts but no resolution. Freeswitch version on Debian 8.9 FreeSWITCH (Version 1.6.16 git 3da6bd0 2017-04-07 16:49:13Z 64bit) Any idea or suggestion for workaround? Jai

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302 handling with enterprise :_: dial.
by Martin Paterson 07 Dec '21

07 Dec '21

I have an issue that I’m struggling to resolve. I’ve tried this out in the vanilla config. If a bridged destination in the default context: <action application="bridge" data="user/1001@${domain_name}”/> returns a 302 Moved Temporarily then the new destination is run through the dialplan in the same default context and everything is good. In the log: [DEBUG] sofia.c:6971 Redirect: Transfering to 1003 [DEBUG] switch_ivr.c:2289 (sofia/internal/1000(a)192.168.1.226) State Change CS_EXECUTE -> CS_ROUTING [NOTICE] switch_ivr.c:2296 Transfer sofia/internal/1000(a)192.168.1.226 to XML[1003@default] However I need to call multiple destinations at once. When separating them with comma, I sometimes get “Only calling the first element in the list in this mode”. Some destinations in my case may have multiple registrations, so that’s not an option. Instead I do enterprise dial with :_: which gets round that: <action application="bridge" data="user/1001@${domain_name}:_:user/1002@${domain_name}"/> but it has the effect that if a destination returns 302 Moved Temporarily (1002 forwards to 1003 here), then it doesn’t run through the dialplan the same context, it goes to the public context and fails because the destination (1003) is an extension in the default context. The log looks like it’s handling the 302 as if it’s a new call: [INFO] sofia.c:10462 sofia/internal/1000(a)192.168.1.226 receiving invite from 192.168.1.226:5060 version: 1.10.7 -release-19-883d2cb662 64bit call-id: ee6cd84a-cca5-123a-d6a1-080027d0c8d3 [INFO] mod_dialplan_xml.c:639 Processing Extension 1000 <1000>->1003 in context public EXECUTE [depth=0] sofia/internal/1000(a)192.168.1.226 deflect(1003) I must be missing something in my understanding here – I don’t understand why the behaviour is different, but more interestingly, is there a way of getting the enterprise dial to process the 302s in the same way as when dialling one destination? Best wishes, Martin. Martin Paterson, Pattersong Music Reduced orchestrations of G&S

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Serving sofia.conf via lua (updated)
by Antony Stone 05 Dec '21

05 Dec '21

Hi. Sending again with a bit of clarification - my original message hasn't hit the list yet, so I'm hoping people can reply to just this one... I am trying to get Freeswitch to get its sofia configuation from lua instead of static XML files. https://freeswitch.org/confluence/display/FREESWITCH/mod_xml_curl tells me for example that "Section: configuration - This is used to send back configuration files such as sofia.conf." Yes, I know that that page refers to mod_xml_curl and not mod_xml_lua but I'm pretty certain that the principle is the same, and https://freeswitch.org/confluence/display/FREESWITCH/Serving+Configuration+… seems to confirm this, although is it much thinner on examples. I do not that this page also says "there is no Lua page yet, and Lua-related docs are scattered all over the place..." :( So, anyway, I'm pretty sure what I want to do should be possible, but nowhere have I so far been able to find a fully working example telling me how to do it. My understanding so far is that: 1. I need to create a lua script which generates the *complete* XML as would be served from /etc/freeswitch/autoload_configs/sofia.conf.xml and all its included sub-directories/files 2. This XML as output from the script should be enclosed within tags: <document type="freeswitch/xml"> <section name="configuration"> ... XML goes here ... </section> </document> 3. I should place this lua script in freeswitch's scripting directory, which on my (Debian) machine is /usr/share/freeswitch/scripts 4. I need to enable this script to be run by including in /etc/freeswitch/autoload_configs/lua.conf.xml: <param name="xml-handler-script" value="sofia.conf.lua"/> <param name="xml-handler-bindings" value="configuration"/> The first thing which is not clear to me is what I should do about /etc/freeswitch/autoload_configs/sofia.conf.xml - if I leave it as it is, it appears to be used in the normal way, and my lua script is ignored - if I delete it (or rename it to sofia.conf.xml.noload for example) then sofia doesn't get loaded *at all* and when I go into fs_cli I do not even have a "sofia" command - if I "touch" the filename so that /etc/reeswitch/autoload_configs/sofia.conf.xml exists but is empty, I get the same result egain - the "sofia" command does not even exist in the cli So, can anyone point me at how to join these things up correctly so that I can serve sofia configuration settings from a lua script? I've been told here that: On Friday 05 November 2021 at 12:28:55, David Villasmil wrote: > In FS there’s the embedded lua. You can do almost anything with it. I'm struggling to find the documentation telling me how, though :( Thanks in advance, Antony. -- "It would appear we have reached the limits of what it is possible to achieve with computer technology, although one should be careful with such statements; they tend to sound pretty silly in five years." - John von Neumann (1949) Please reply to the list; please *don't* CC me.

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Re: [Freeswitch-users] WebRTC calls one way with custom sip messages UUI
by kaleem rehman 03 Dec '21

03 Dec '21

Hi Kaiduan, thanks for looking into it. Verto looks cool. we have no restriction as to what to use. main item is to attach data to call so sip client at end user can strip and show data to agent. no need for user to login to enter credentials, we want simple "*call us"* type button which generates a call. to make it safe from attacks as server will be on cloud, we would like some sort of safety measure, either a login and pwd, which gets passed to freeswitch to verify its genuine call from a webpage or app. or some hidden message within the generate call command so freeswitch can verify and drop any calls which arent from right source so answering party doesnt get too many junk calls from random bots who discover port is open on Cloud FS. Regards, K ---------- Forwarded message ---------- From: kaiduan xie <kaiduanx(a)yahoo.ca> To: "freeswitch-users(a)lists.freeswitch.org" <freeswitch -users(a)lists.freeswitch.org> Cc: Bcc: Date: Fri, 26 Nov 2021 02:57:01 +0000 (UTC) Subject: Re: [Freeswitch-users] WebRTC calls one way with custom sip messages UUI You can use JSON based VERTO protocol instead of SIP to make things easier. Does the user have to login in FS? /Kaiduan On Thursday, November 25, 2021, 03:30:52 p.m. EST, kaleem rehman < k4kaleem(a)gmail.com> wrote: Salaam Ehtasham, we are looking to use JSSIP or SIPJS, we are flexible and can look into SIPML if for any reason we have to. Regards, Kaleem ---------- Forwarded message ---------- From: Ehtasham Ul-Haq <ehtasham.malik(a)expertflow.com> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Cc: Ahmed Hasan <ahmad.hasan(a)expertflow.com> Bcc: Date: Thu, 25 Nov 2021 15:28:31 +0500 Subject: Re: [Freeswitch-users] WebRTC calls one way with custom sip messages UUI Hi Which Library you are using to start a call from Website ? Malik Ehtasham, CTI Product Manager / Technical Lead (Mr. ) WWW:[image: domain2.png].expertflow.com <http://www.expertflow.com/> FB: [image: FB-f-Logo__blue_29.png]/Expertflow <https://www.facebook.com/Expertflow> LinkedIn: [image: linkedIn.png] /company/expertflow <https://www.linkedin.com/company/expertflow> Youtube: [image: YouTube-social-square_red_128px.png]/user/expertflow <https://www.youtube.com/user/expertflow> Twitter: [image: twitter.JPG] /Expertflow <https://twitter.com/Expertflow> 361 Model Town Lahore Pakistan ; Mobile +92 3347815664; email, Cisco Spark and Google Talk: ehtasham.malik(a)expertflow.com <andreas.stuber(a)expertflow.com>; Skype: <http://andreas.stuber.expertflow.com/>*shani.awan3* On Thu, Nov 25, 2021 at 4:16 AM kaleem rehman <k4kaleem(a)gmail.com> wrote: Hi All, our requirement is simple, we will have CALL US button on website when they click, we want a call generated to our FS Server via WebRTC (no need for calls from FS to Users, it will be one way only from User to Server. With call we want to send additional data like URL of page they on, login if they are logged in. we can get data like URL and userlogin but want to sent it with SIP call as SIP Message (Probably as USER to USER Information) so we can pull at other end. any ideas of achieving this Thanks, Kaleem _________________________________________________________________________ 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 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 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 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com

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Overriding hangup cause in CDR logs
by Rahman Duran 02 Dec '21

02 Dec '21

Hi, I am using announcements for fail hungup causes like busy, no_answer etc. Here is my dial plan <extension name="local_accounts_freeswitch"> <condition regex="all"> <regex field="destination_number" expression="^[0-9]{4}$"/> <regex field="${local_user_exists}" expression="^true$|^TRUE$|^True$"/> <regex field="${santral_kontrol}" expression="^freeswitch$"/> <action application="set" inline="true" data="call_pickup_group=${user_data(${destination_number}@${domain_name} var call_pickup_group)}"/> <action application="hash" data="insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}"/> <action application="hash" data="insert/${domain_name}-group_pickup_last_uuid/${call_pickup_group}/${uuid}"/> <action application="hash" data="insert/${domain_name}-pickup_last_uuid/${destination_number}/${uuid}"/> <action application="log" data="ep_codec_string -- ${ep_codec_string}" /> <action application="set" data="inherit_codec=true"/> <action application="export" data="nolocal:absolute_codec_string=${ep_codec_string}"/> <action application="set_profile_var" data="callee_id_name=${user_data(${destination_number}@${domain_name} var effective_caller_id_name)}"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="call_timeout=30"/> <action application="set" data="execute_on_answer=sched_hangup +21600 alloted_timeout" /> <action application="bridge" data="{origination_callee_id_name=${user_data(${destination_number}@${domain_name} var effective_caller_id_name)}}user/${destination_number}@${domain_name}"/> <action application="execute_extension" data="hata-${originate_disposition} XML hata_anonslari"/> </condition> </extension> And here is announcements context that I handle hangup causes: <include>  <context name="hata_anonslari">  <extension name="error-causes-user_busy"> <condition field="destination_number" expression="^hata-USER_BUSY$"> <action application="export" data="cdr_hata_anonsu=USER_BUSY"/> <action application="pre_answer"/> <action application="playback" data="$${anons_dosya_yolu}/user_busy.wav"/> <action application="sleep" data="1000"/> <action application="hangup" data="${originate_disposition}"/> </condition> </extension> <extension name="error-causes-no_answer"> <condition field="destination_number" expression="^hata-NO_ANSWER$"> <action application="export" data="cdr_hata_anonsu=NO_ANSWER"/> <action application="pre_answer"/> <action application="playback" data="$${anons_dosya_yolu}/no_answer.wav"/> <action application="sleep" data="1000"/> <action application="hangup" data="${originate_disposition}"/> </condition> </extension> <extension name="error-causes-user_not_registered"> <condition field="destination_number" expression="^hata-USER_NOT_REGISTERED$"> <action application="export" data="cdr_hata_anonsu=USER_NOT_REGISTERED"/> <action application="pre_answer"/> <action application="playback" data="$${anons_dosya_yolu}/user_not_registered.wav"/> <action application="sleep" data="1000"/> <action application="hangup" data="${originate_disposition}"/> </condition> </extension> <extension name="error-causes-any"> <condition field="destination_number" expression="^hata-(.*)$"> <action application="export" data="cdr_hata_anonsu=${originate_disposition}"/> <action application="hangup" data="${originate_disposition}"/> </condition> </extension>  </context> </include> The problem is if the calling leg does not listen the announcement to the end and hangup, CDR logs shows "Originator Cancel" as hangup cause. As I already know the real hangup cause, how can I override the CDR hangup cause with the real one? For now I am setting another variable (cdr_hata_anonsu) and added it to CDR logs, but if possible I want to fix this with a real solution. Regards, Rahman Duran

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