October 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

5 8

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.

3 2

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

11 18

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?

2 5

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

8 12

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

5 13

Bridge to other FS server has no audio until DTMF
by Avi Marcus 14 Nov '21

14 Nov '21

I started a new thread in case anyone muted it... it wasn't simply a network issue. It seems the bridging occurs and dialplan processes, but no media flows - until DTMF from the A-leg. Call flow: PSTN (via carrier) to freeswitch A -> media and IVR -> freeswitch B. Calls directly from carrier to Freeswitch B are fine. Calls from a different carrier to Freeswitch A -> media and IVR -> Freeswitch B are also fine. So it sounds like a carrier/unique SIP/RTP issue, but since FS is in the media path, it's an FS issue... I actually mcguyvered this right now with a queue_dtmf before the bridge, to force the audio stream to update. Here's the log on freeswitch B: EXECUTE [depth=0] sofia/external/972581234567(a)172.123.123.123 log(DEBUG class chosen: 1234567) 2021-10-07 09:16:24.343175 [DEBUG] mod_dptools.c:1879 class chosen: 1234567 EXECUTE [depth=0] sofia/external/972581234567(a)172.123.123.123 javascript(conference/lookupAndJoinConference.js 1234567) EXECUTE [depth=0] sofia/external/972581234567(a)172.123.123.123 playback(class/hold-wait-teacher.wav) 2021-10-07 09:16:24.363379 [DEBUG] sofia.c:7406 Channel sofia/external/ 972581234567(a)172.123.123.123 entering state [completed][200] 2021-10-07 09:16:24.363379 [DEBUG] sofia.c:7406 Channel sofia/external/ 972581234567(a)172.123.123.123 entering state [ready][200] 2021-10-07 09:16:24.363379 [DEBUG ] switch_ivr_play_say.c:1486 Codec Activated L16@8000hz 1 channels 20ms 2021-10-07 09:16:34.903283 [DEBUG ] switch_rtp.c:7793 Correct audio ip/port confirmed. 2021-10-07 09:16:34.923190 [DEBUG] switch_rtp.c:8038 RTP RECV DTMF 3:2080 2021-10-07 09:16:34.923190 [INFO] switch_channel.c:522 RECV DTMF 3:2080 2021-10-07 09:16:34.923190 [DEBUG] mod_dptools.c:2389 Digit 3 2021-10-07 09:16:37.143169 [DEBUG ] switch_ivr_play_say.c:1931 done playing file /usr/share/freeswitch/sounds/en/us/matthew/class/hold-wait-teacher.wav You can see a 10 second gap between call ready 200 and correct audio/ip and file done playing (it's a 2 second file), and this doesn't happen automatically, only when I choose to press something. Any ideas as to the root cause of this? -Avi Marcus ---------- Forwarded message --------- From: Avi Marcus <avi(a)avimarcus.net> Date: Wed, Oct 6, 2021 at 3:32 PM Subject: Bridge to other FS server has no audio ??? To: FreeSWITCH Users Help <FreeSWITCH-users(a)lists.freeswitch.org> Any ideas on why a call doesn't have media? It used to work, but I think my upstream changed his SDP again. - FreeSWITCH Server A - call comes in and bypass_media bridges to FS server B. Media works. - FreeSWITCH Server A - call comes in and bridges to FS server B (not on bypass). Media works. - FreeSWITCH Server A - call comes in, gets answered, then bridges to FS server B. Call looks OK, but no media is flowing (I don't hear anything, PCAPs just have SIP, and there isn't 80kbps network traffic). All the same codecs are set in the json cdrs (PCMU). FS server B is to join a conference if that matters. I was assuming it had to do with codecs, but setting absolute_codec_string to PCMU doesn't make any difference in the logs - it's already always PCMU. I have NO clue what further could cause this other than codecs, which seem to be fine. Any ideas please? -Avi Marcus

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Developers / packagers here?
by Antony Stone 05 Nov '21

05 Nov '21

Hi. Can anyone tell me why https://freeswitch.org/confluence/display/FREESWITCH/mod_odbc_query is not built as part of the standard Debian packages available from https://files.freeswitch.org/repo/deb/ ? Thanks, Antony. (PS: This message sent at 15:21 UTC 2021-10-29, just wondering when it will appear on the list.) -- 90% of networking problems are routing problems. 9 of the remaining 10% are routing problems in the other direction. The remaining 1% might be something else, but check the routing anyway. Please reply to the list; please *don't* CC me.

6 11

SIP over DTLS support
by Telium Technical Support 04 Nov '21

04 Nov '21

Does FreeSwitch support SIP of DTLS? Not clear from the docs (though mentions of webrtc possibly using this)

4 4