February 2022 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Issue with Enterprise originate and 487 "ORIGINATOR_CANCEL".
by Christian Berger 25 Oct '22

25 Oct '22

Thanks :) Am 14.01.22 um 16:15 schrieb Zvonimir Bužanić via FreeSWITCH-users: > Hi, I've created pull request: > > https://github.com/signalwire/freeswitch/pull/1520 Christian Berger foncloud GmbH & Co KG Hahlweg 2a 36093 Künzell Tel: / Fax: +49 661 968990-99 Email: Christian.Berger(a)foncloud.net Web: www.foncloud.net P.S.: Wussten Sie schon? Unter https://www.foncloud.net/wissen finden Sie zahlreiche Informationen und hilfreiche Artikel rund um unsere Produkte und Services. Registergericht: Amtsgericht Fulda, Persönlich haftende Gesellschafterin der foncloud GmbH&Co.KG: Global Brain Network GmbHGeschäftsführer der Global Brain Network GmbH: Peter Krug Sitz der Gesellschaft: Künzell. Diese E-Mail enthält vertrauliche und/oder rechtlich geschützte Informationen. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrtümlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser Mail ist nicht gestattet. This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorised copying, disclosure or distribution of the material in this e-mail is strictly forbidden.

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Spurious DTMF characters when using RFC2833 with SRTP
by Andy Newlands 04 Aug '22

04 Aug '22

Hi, We have a reproducible problem where spurious DTMF characters are introduced on calls from Freeswitch to a remote (3rd party) IVR, but ONLY when using SRTP (not with RTP) When the spurious character is generated it is: 1. Only ever following a valid/legitimate character (one that was genuinely keyed) 2. Often has a very long duration (see below) 3. Appears to be a random character - often 0x00, but sometimes 0-9/A-F,#,* (and that's when we get problems with the IVR - FS automatically drops the 0x00 values). We have been able to reproduce this from mobile/cell phones and traditional land-lines. I added some code to: switch_rtp.c, in static handle_rfc2833_result_t handle_rfc2833(switch_rtp_t *rtp_session, switch_size_t bytes, int *do_cng), to try to address this. It logs an error (with the character hex value and duration) for DTMF characters with "long" durations (over 10000) and then discards them. As you can see, below, this is a fairly frequent occurrence (although most of the time, the character value is null, so FS drps it, anyway - but sometimes, "real" characters are introduced). 2022-02-28 14:00:53.743665 [NOTICE] switch_vpx.c:599 VPX encoder reset (WxH/BW) from 0x0/0 to 352x288/1024 2022-02-28 14:00:56.103665 [INFO] switch_channel.c:522 RECV DTMF 4:2400 2022-02-28 14:00:56.303664 [ERR] switch_channel.c:557 Ignored invalid DTMF on Call-ID: 3b5d08d3-d898-40b6-ae00-5886713ca6dd, DTMF: 0x0 Duration: 5063 2022-02-28 14:00:56.383661 [INFO] switch_channel.c:522 RECV DTMF 4:2400 2022-02-28 14:00:56.583665 [INFO] switch_channel.c:522 RECV DTMF 5:2400 2022-02-28 14:00:57.063662 [INFO] switch_channel.c:522 RECV DTMF 5:2400 2022-02-28 14:00:57.123670 [INFO] switch_channel.c:522 RECV DTMF 8:2720 2022-02-28 14:00:57.323673 [ERR] switch_rtp.c:686 Discarded long-duration DTMF on Call-ID: 3b5d08d3-d898-40b6-ae00-5886713ca6dd, DTMF: 0x0 Duration: 54088 2022-02-28 14:00:57.603671 [INFO] switch_channel.c:522 RECV DTMF 9:2400 2022-02-28 14:00:57.824102 [INFO] switch_channel.c:522 RECV DTMF 8:2720 2022-02-28 14:00:57.903663 [ERR] switch_rtp.c:686 Discarded long-duration DTMF on Call-ID: 3b5d08d3-d898-40b6-ae00-5886713ca6dd, DTMF: 0x0 Duration: 57002 2022-02-28 14:00:58.183668 [INFO] switch_channel.c:522 RECV DTMF 6:2400 2022-02-28 14:00:58.363666 [ERR] switch_rtp.c:686 Discarded long-duration DTMF on Call-ID: 3b5d08d3-d898-40b6-ae00-5886713ca6dd, DTMF: 0x35 Duration: 57073 2022-02-28 14:00:58.503667 [INFO] switch_channel.c:522 RECV DTMF 9:2400 2022-02-28 14:00:58.643667 [INFO] switch_channel.c:522 RECV DTMF 1:2400 2022-02-28 14:00:58.963663 [INFO] switch_channel.c:522 RECV DTMF 6:2400 2022-02-28 14:00:59.283665 [INFO] switch_channel.c:522 RECV DTMF 1:2400 The first spurious character is 0x00 (which FS will drop anyway) but this could easily be a valid character (and would escape detection because its duration is not too long - it's long but not unreasonable). So, this is an imperfect "fix" as we occasionally see a spurious character with a legitimate value and a "sensible" duration - sufficiently often for users to complain. The last [ERR] entry shows a '5' being introduced but this is dropped because the duration is "crazy" (57073) As mentioned, this does not happen with secure media disabled - only with SRTP. So, I am wondering if the code correctly processes a digit then, sometimes (somehow) incorrectly attempts to decode part of an audio packet as though it were RFC2833. Has anyone else experienced this? Does anyone know what may be causing the problem and what the fix might be. I can provide console output and tcp dumps if required, along with any other details that may help (I'm also comfortable making code changes - to for extra logging/diagnostics etc). Thank you. Kind regards, Andy

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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

Re: [Freeswitch-users] debian repos for Raspberry Pi broken?
by Mohammad-Reza Tazari 10 Jun '22

10 Jun '22

Dear Andrey, any news here? Thanks and kind regards, -- Saied Andrey Volk wrote on 21-Dec-2021 21:36: > > Let me see what I can do. May take some time. > > вт, 21 дек. 2021 г. в 23:19, Johan Helsingius <julf at julf.com <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>>: > > > >* 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/ <http://files.freeswitch.org/repo/deb/rpi/debian-release/> bullseye main *> >* deb-src http://files.freeswitch.org/repo/deb/rpi/debian-release/ <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 *

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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.

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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

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

Copying From header
by Adiseshu Channasamudhram 01 Mar '22

01 Mar '22

Hello FS Team, How can I copy the From header in INVITE from incoming leg to the outgoing leg? Basically i have the carrier send the From header as below [not the clean way] From: <sip:1234567890@1.19.10.3>;isup-oli=00;tag=3418741140365741_c2b08.2.4.1638259348522.0_6724417_21166610 I want to make sure i send out the oli in the outgoing INVITE From header. Note that in the incoming From , the oli is outside the uri Thanks a lot in advance Regards Adises

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Re: [Freeswitch-users] PCAP help and many call failures PLEASE HELP
by mayamatakeshi 24 Feb '22

24 Feb '22

On Thu, Feb 24, 2022 at 12:27 AM Sean Devoy <sdevoy(a)bizfocused.com> wrote: > > > I have debug level FS logs. There are so many lines it is difficult to > follow a single call. > > > Do you have the logs related to the capture file? If yes, then search for the Call-IDs of the troubled call: Leg1: Call-ID: f0544e98-378154a6(a)192.168.2.111 Leg2: Call-ID: 26e3398f-0e97-123b-3b83-16f3d912a996 For example, I did a test call with Call-ID: 090abe39-188c-4f06-b371-e8ffe3d862f8 So I grepped the freeswitch.log file with: [root@lab002 ~]$ grep 090abe39-188c-4f06-b371-e8ffe3d862f8 /usr/local/freeswitch/log/freeswitch.log 4b860aaf-b481-4e26-8886-37321c2fc3f3 2022-02-24 06:49:17.131915 [INFO] sofia.c:10414 sofia/external_in/123412341234(a)test.com receiving invite from 127.0.0.1:5060 version: 1.10.7-dev git dd24113 2021-08-25 17:37:19Z 64bit call-id: 090abe39-188c-4f06-b371-e8ffe3d862f8 The 4b860aaf-b481-4e26-8886-37321c2fc3f3 above is the channel uuid (the unique identifier of the channel in freeswitch). The freeswitch log file will include that channel uuid in all log lines related to that channel. So you can grep your freeswitch.log file for it. If no clue about the problem is found, do the same for the other leg. > This happens are about 50% of ALL calls. > > Here is the dial plan: > > <extension name="ten_or eleven_digiti_dial" > > > <condition field="destination_number" expression=" > ^(?:1|\+1)?([2-9]\d\d[2-9]\d{6})$"> > > <action application="set" data="ringback=${us-ring}"/> > > <action application="set" data=" > effective_caller_id_number=${outbound_caller_id_number}"/> > > <action application="set" data="ani=${outbound_caller_id_number}"/> > > <action application="set" data=" > origination_caller_id_number=${outbound_caller_id_number}"/> > > <action application="set" data="effective_caller_id_name=REDACTED"/> > > <action application="set" data= > sip_from_uri=${outbound_caller_id_number}@ > <sip_from_uri=$%7boutbound_caller_id_number%7d(a)64.136.173.30>REDACTED/> > > <action application="set" data=" > sip_from_user=${outbound_caller_id_number}"/> > > <action application="set" data="sip_from_host=REDACTED"/> > > <action application="set" data="sip_h_X-CID=${sip_call_id}"/> > > > > <action application="set" data="continue_on_fail=true"/> > > <action application="set" data="hangup_after_bridge=true"/> > > <action application="set" data="sip_h_X-CID=${sip_call_id}"/> > > <action application="bridge_export" data=" > hold_music=/etc/freeswitch/recordings/REDACTED-moh.wav"/> > > <action application="bridge" data= > {sip_from_uri=${outbound_caller_id_number}@REDACTED,sip_from_user=${outbound_caller_id_number},sip_from_host=REDACTED}sofia/gateway/voip-innovations-outbound/1$1 > <%7bsip_from_uri=$%7boutbound_caller_id_number%7d@64.136.173.30,sip_from_user=$%7boutbound_caller_id_number%7d,sip_from_host=64.136.173.30%7dsofia/gateway/voip-innovations-outbound/1$1> > /> > > </condition> > > </extension> > > > OK. So there is nothing setting an answer timeout (so it would default to 60 seconds). The variable controlling it would be this one: https://freeswitch.org/confluence/display/FREESWITCH/call_timeout

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