December 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

3 3

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

Predictive dialer
by Gregor Nanger 28 Dec '21

28 Dec '21

Hi! I want to build a predictive dialer and need some advice on architecture logic. There are a lot of studies on how to calculate predictive calls, but I need most from a programming point of view. When the campaign starts, should I create a schedule to run every n seconds and check current calls, logged agents, statistics and then decide to trigger new calls? Or should I subscribe to call events and on every hangup calculate new calls? Any thoughts? BR, Gregor

2 3

Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 186, Issue 39
by Tony Pemberton 28 Dec '21

28 Dec '21

After many years of tinkering with Freeswitch at home, this is my first posting so I might get things a bit wrong! I also have been trying to install freeswitch 1.10.7 on a 4Gb Raspberry Pi and have come across the same installation problems. I thought I read that only 64 bit builds of Freeswitch were to be supported in Debian Bullseye so I also tried the development version of RaspbianOS 64bit but got just the same dependency problems. Looking at the Repo's, Freeswitch Bullseye binaries are only available for Intel/AMD architectures at present. As a result, I have decided to compile from source using armhf (32bit) version of Raspbian Bullseye. I did have to compile sofia-sip, spandsp and links from git sources to satisfy Freeswitch's bootstrap/configure dependencies. As my domestic system does not access the Signalwire cloud, I have, for the time being, disabled the mod_signalwire in modules.conf. A compilation is in progress as I write this so I'm not yet sure I will get a clean compile. But at least I have got past the dependency issue. Regards, Tony Pemberton On Wed, 22 Dec 2021 at 12:09, <freeswitch-users-request(a)lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > Today's Topics: > > 1. debian repos for Raspberry Pi broken? (Johan Helsingius) > 2. Re: debian repos for Raspberry Pi broken? (Andrey Volk) > 3. Re: debian repos for Raspberry Pi broken? (Johan Helsingius) > > > > ---------- Forwarded message ---------- > From: Johan Helsingius <julf(a)julf.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Tue, 21 Dec 2021 20:59:31 +0100 > Subject: [Freeswitch-users] debian repos for Raspberry Pi broken? > Hi, > > Trying to install freeswitch on a RPi 4 (Debian 11 Bullseye), > my /etc/apt/sources.list.d/freeswitch.list has: > > deb http://files.freeswitch.org/repo/deb/rpi/debian-release/ bullseye main > deb-src http://files.freeswitch.org/repo/deb/rpi/debian-release/ bullseye > main > > but when I try > > apt-get update && apt-get install -y freeswitch-meta-all > > I get > > Unable to locate package freeswitch-meta-all > > apt-cache search freeswitch only gives me: > libsipwitch-dev - secure peer-to-peer SIP VoIP server - development files > libsipwitch1 - secure peer-to-peer SIP VoIP server - shared libraries > libsipwitch1-dbg - secure peer-to-peer SIP VoIP server - debug symbols > sipwitch - secure peer-to-peer VoIP server for the SIP protocol > sipwitch-cgi - secure peer-to-peer SIP VoIP server - CGI XML-RPC interface > > Any suggestions? > > Julf > > > > > > > ---------- Forwarded message ---------- > From: Andrey Volk <andywolk(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Tue, 21 Dec 2021 23:36:21 +0300 > Subject: Re: [Freeswitch-users] debian repos for Raspberry Pi broken? > Let me see what I can do. May take some time. > > вт, 21 дек. 2021 г. в 23:19, Johan Helsingius <julf(a)julf.com>: > >> Hi, >> >> Trying to install freeswitch on a RPi 4 (Debian 11 Bullseye), >> my /etc/apt/sources.list.d/freeswitch.list has: >> >> deb http://files.freeswitch.org/repo/deb/rpi/debian-release/ bullseye >> main >> deb-src http://files.freeswitch.org/repo/deb/rpi/debian-release/ >> bullseye main >> >> but when I try >> >> apt-get update && apt-get install -y freeswitch-meta-all >> >> I get >> >> Unable to locate package freeswitch-meta-all >> >> apt-cache search freeswitch only gives me: >> libsipwitch-dev - secure peer-to-peer SIP VoIP server - development files >> libsipwitch1 - secure peer-to-peer SIP VoIP server - shared libraries >> libsipwitch1-dbg - secure peer-to-peer SIP VoIP server - debug symbols >> sipwitch - secure peer-to-peer VoIP server for the SIP protocol >> sipwitch-cgi - secure peer-to-peer SIP VoIP server - CGI XML-RPC interface >> >> Any suggestions? >> >> Julf >> >> >> _________________________________________________________________________ >> >> 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 > > > > > ---------- Forwarded message ---------- > From: Johan Helsingius <julf(a)julf.com> > To: freeswitch-users(a)lists.freeswitch.org > Cc: > Bcc: > Date: Tue, 21 Dec 2021 22:11:37 +0100 > Subject: Re: [Freeswitch-users] debian repos for Raspberry Pi broken? > On 21-12-2021 21:36, Andrey Volk wrote: > > Let me see what I can do. May take some time. > > Thanks! > > Julf > > > > _________________________________________________________________________ > > 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

1 0

Adding optional SDP Header in the Invite
by Jenish patel 23 Dec '21

23 Dec '21

Hi, I am looking for a way to add the optional sdp header for the bandwidth in the invite sdp (reference *rfc3890*). I see that the video has the option to modify it with the variables rtp_video_max_bandwidth_in rtp_video_max_bandwidth but for the audio I don't see any options or reference. Regards, Jenish

1 0

SIP timer at 900 seconds?
by Avi Marcus 22 Dec '21

22 Dec '21

I see two recent calls from FreeSWITCH that end at 15 minutes: billsec: 904.1, hangup cause: MANDATORY_IE_MISSING billsec: 901.3, hangup cause: MANDATORY_IE_MISSING My system is hanging up the call. Is there some sort of SIP timer? I don't have a SIP trace to see what happened. The remote carrier said: >However I'm unsure of what the issue is you are reporting as the BYE simply came from your side? >This is not something we are familiar with or why your server did or requires this. > Reason: SIP;cause=488;text="No answer to offer". Any suggestions? -Avi Marcus

2 1