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Unable to start the Free switch server
by veeru anji 05 Aug '19

05 Aug '19

Dear Team, I am getting below error message. This is the result when I am trying with Sofia status command freeswitch@CITHYDDKTP-030> Sofia Status Name Type Data State ================================================================================================= external-ipv6 profile sip:mod_sofia@[::1]:5080 RUNNING (0) external profile sip:mod_sofia@183.82.114.188:5080 RUNNING (0) external::example.com gateway sip:joeuser@example.com NOREG ================================================================================================= 2 profiles 0 aliases But below error is coming>>>>>>>>>>>>>> *Error Creating SIP UA for profile: internal-ipv6 (sip:mod_sofia@[::1]:5060;transport=udp,tcp) ATTEMPT 3 (RETRY IN 5 SEC)2019-08-05 16:20:23.012205 [ERR] sofia.c:3238 Error Creating SIP UA for profile: internal-ipv6 (sip:mod_sofia@[::1]:5060;transport=udp,tcp)The likely causes for this are:1) Another application is already listening on the specified address.2) The IP the profile is attempting to bind to is not local to this system.*

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Trying to install Docker Container FS
by Danny Dias 03 Aug '19

03 Aug '19

Hi Community! Having some weird issues following this guide to install my FS Containers: https://hub.docker.com/r/bettervoice/freeswitch-container/ I ran the following command to download and run the image: dadiaspe@srv-lx-containers-fs-01:~$ CID=$(sudo docker run --name freeswitch -p 5060:5060/tcp -p 5060:5060/udp -p 5080:5080/tcp -p 5080:5080/udp -p 8021:8021/tcp -p 7443:7443/tcp -p 60535-65535:60535-65535/udp -v /home/debian/freeswitch/conf:/usr/local/freeswitch/conf bettervoice/freeswitch-container:1.6.6) It did several Pulls and Downloads, and returned the following message at the end: Digest: sha256:e0db31cd71a7c8ef63be9b20c438bc94f29b73dc147b294a9e1d2b6ce17b2a01 Status: Downloaded newer image for bettervoice/freeswitch-container:1.6.6 . . freeswitch is not configured so not starting. Please review /usr/local/doc/freeswitch/README.Debian *tail: unrecognized file system type 0x794c7630 for '/usr/local/freeswitch/log/freeswitch.log'. please report this to bug-coreutils(a)gnu.org <bug-coreutils(a)gnu.org>. reverting to polling* But the container seems to be running cause i can connect, and "docker ps -a" reports the container UP: dadiaspe@srv-lx-containers-fs-01:~$ docker ps -a CONTAINER ID IMAGE COMMAND CREATED STATUS PORTS NAMES d70484c9f2cb bettervoice/freeswitch-container:1.6.6 "/bin/sh -c 'service…" About an hour ago Up 56 minutes 0.0.0.0:5060->5060/tcp, 0.0.0.0:5060->5060/udp, 0.0.0.0:5080->5080/tcp, 0.0.0.0:7443->7443/tcp, 0.0.0.0:5080->5080/udp, 0.0.0.0:8021->8021/tcp, 0.0.0.0:60535-65535->60535-65535/udp, 5066/tcp freeswitch dadiaspe@srv-lx-containers-fs-01:~$ docker exec -it freeswitch /bin/bash root@d70484c9f2cb:/usr/src/freeswitch# FS appears to be running: root@d70484c9f2cb:/usr/src/freeswitch# ps -aux | grep free root 1 0.0 0.0 4444 620 ? Ss 21:07 0:00 /bin/sh -c service snmpd start && service freeswitch start && tail -f /usr/local/freeswitch/log/freeswitch.log root 21 0.0 0.0 4380 800 ? S 21:07 0:00 tail -f /usr/local/freeswitch/log/freeswitch.log root 71 0.0 0.0 8864 884 pts/0 S+ 22:07 0:00 grep --color=auto free But can't connect to fs_cli: root@d70484c9f2cb:/usr/src/freeswitch# /usr/local/freeswitch/bin/fs_cli *[ERROR] fs_cli.c:1659 main() Error Connecting [Socket Connection Error]* Am i doing something wrong with the container? I'm planning to run around 4 FS containers in this Host...any ideas/recommendations highly appreciated! Thank u

2 1

dialplan continues with bypass_media
by Keith 03 Aug '19

03 Aug '19

I have a mod_python dialplan and have been experimenting with taking FreeSwitch out of the media path when the call answers. It's working great now, and scheduled playback, hangup works brilliantly! However, I noticed that when I set channel variable bypass_media or bypass_media_after_bridge then the dialplan continues as soon as the media is bypassed. I am doing some things in the dialplan after the bridge, like playing audio depending on the hang-up cause, but also some logging of call timings after the bridge. see: https://github.com/Rhizomatica/rccn/blob/master/rccn/context.py#L258 I was surprised to see that code running when I set bypass_media_after_bridge on the channel, but maybe my logic is wrong here, and I should just return after session.execute('bridge') when originate_disposition is SUCCESS and handle the logging in the hangup hook? All the same, I was wondering if there was any way to not have the bridge return after media is bypassed? I've scoured the docs and had a look through the source code and I think the answer is no, but I thought I'd ask. I'm also interested as to the logic of why then the bridge function does not return on SUCCESS if one is not bypassing media. Thanks! Keith.

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Excessive logging?
by Allan Kristensen 01 Aug '19

01 Aug '19

Hello, I'm using the master branch to test some things and I've noticed that there is a lot of these: 2019-08-01 11:50:07.238469 [NOTICE] switch_rtp.c:2191 Sending RTCP SR (ssrc=3712758684) 2019-08-01 11:50:08.258456 [NOTICE] switch_rtp.c:2191 Sending RTCP SR (ssrc=3712758684) 2019-08-01 11:50:09.278517 [NOTICE] switch_rtp.c:2191 Sending RTCP SR (ssrc=3712758684) 2019-08-01 11:50:10.298459 [NOTICE] switch_rtp.c:2191 Sending RTCP SR (ssrc=3712758684) 2019-08-01 11:50:10.518468 [NOTICE] switch_rtp.c:2191 Sending RTCP SR (ssrc=156472476) 2019-08-01 11:50:11.318554 [NOTICE] switch_rtp.c:2191 Sending RTCP SR (ssrc=3712758684) 2019-08-01 11:50:12.338457 [NOTICE] switch_rtp.c:2191 Sending RTCP SR (ssrc=3712758684) 2019-08-01 11:50:13.358469 [NOTICE] switch_rtp.c:2191 Sending RTCP SR (ssrc=3712758684) 2019-08-01 11:50:14.378473 [NOTICE] switch_rtp.c:2191 Sending RTCP SR (ssrc=3712758684) I can see the log level was changed from SWITCH_LOG_DEBUG1 to SWITCH_LOG_NOTICE in commit https://freeswitch.org/stash/projects/FS/repos/freeswitch/commits/7906a2054… is this on purpose? Thank you... /Allan

2 1

rtmp streams sometimes take 4+ sec to start
by David P 30 Jul '19

30 Jul '19

Can you suggest ways we could investigate slow rtmp streams? We stream them to another machine in the same cluster, and send a POST just before, but the stream often arrives 4-10 seconds after the POST. At minimal load conditions

2 1

voicemail files all 252 bytes, no audio
by Nathan Stratton 29 Jul '19

29 Jul '19

Logs look normal to me, but all my voicemail files are 252 bytes and do not contain audio. It does not matter if I leave one from a user using opus or from PSTN using g.711, they all are the same. Any ideas? 2019-07-26 11:09:41.780232 [DEBUG] sofia.c:7323 Channel sofia/internal/ nathan(a)stage.illumy1.com entering state [completed][200] 2019-07-26 11:09:41.800231 [DEBUG] sofia.c:7323 Channel sofia/internal/ nathan(a)stage.illumy1.com entering state [ready][200] 2019-07-26 11:09:41.840229 [INFO] switch_rtp.c:3251 Changing audio DTLS state from HANDSHAKE to SETUP 2019-07-26 11:09:41.860232 [INFO] switch_rtp.c:3158 audio Fingerprint Verified. 2019-07-26 11:09:41.860232 [INFO] switch_rtp.c:4153 Activating audio Secure RTP SEND 2019-07-26 11:09:41.860232 [DEBUG] switch_core_sqldb.c:2617 Secure Type: srtp:dtls:AES_CM_128_HMAC_SHA1_80 2019-07-26 11:09:41.860232 [INFO] switch_rtp.c:4131 Activating audio Secure RTP RECV 2019-07-26 11:09:41.860232 [INFO] switch_rtp.c:3200 Changing audio DTLS state from SETUP to READY 2019-07-26 11:09:41.860232 [DEBUG] switch_core_sqldb.c:2617 Secure Type: srtp:dtls:AES_CM_128_HMAC_SHA1_80 2019-07-26 11:09:41.860232 [NOTICE] mod_dptools.c:1377 Channel [sofia/internal/nathan(a)stage.illumy1.com] has been answered 2019-07-26 11:09:41.860232 [DEBUG] switch_channel.c:3825 (sofia/internal/ nathan(a)stage.illumy1.com) Callstate Change EARLY -> ACTIVE EXECUTE [depth=0] sofia/internal/nathan(a)stage.illumy1.com sleep(1000) 2019-07-26 11:09:41.860232 [DEBUG] switch_ivr.c:195 Codec Activated L16@48000hz 1 channels 20ms 2019-07-26 11:09:41.880232 [DEBUG] switch_rtp.c:1908 rtcp_stats_init: audio ssrc[3778260292] base_seq[13090] EXECUTE [depth=0] sofia/internal/nathan(a)stage.illumy1.com voicemail(default beth(a)stage.illumy1.com) 2019-07-26 11:09:42.880231 [DEBUG] switch_ivr.c:195 Codec Activated L16@48000hz 1 channels 20ms 2019-07-26 11:09:42.980231 [DEBUG] switch_ivr_play_say.c:70 No language specified - Using [en] 2019-07-26 11:09:42.980231 [DEBUG] switch_ivr_play_say.c:250 Handle play-file:[voicemail/vm-person.wav] (en:en) 2019-07-26 11:09:42.980231 [DEBUG] switch_ivr_play_say.c:1497 Codec Activated L16@48000hz 1 channels 20ms 2019-07-26 11:09:44.340230 [DEBUG] switch_ivr_play_say.c:1941 done playing file /usr/share/freeswitch/sounds/en/us/callie/voicemail/vm-person.wav 2019-07-26 11:09:44.340230 [DEBUG] switch_ivr.c:195 Codec Activated L16@48000hz 1 channels 20ms 2019-07-26 11:09:44.440231 [DEBUG] switch_ivr_play_say.c:250 Handle play-file:[voicemail/vm-not_available.wav] (en:en) 2019-07-26 11:09:44.440231 [DEBUG] switch_ivr_play_say.c:1497 Codec Activated L16@48000hz 1 channels 20ms 2019-07-26 11:09:45.420231 [DEBUG] switch_ivr_play_say.c:1941 done playing file /usr/share/freeswitch/sounds/en/us/callie/voicemail/vm-not_available.wav 2019-07-26 11:09:45.420231 [DEBUG] switch_ivr.c:195 Codec Activated L16@48000hz 1 channels 20ms 2019-07-26 11:09:45.520231 [DEBUG] switch_ivr_play_say.c:70 No language specified - Using [en] 2019-07-26 11:09:45.520231 [DEBUG] switch_ivr_play_say.c:250 Handle play-file:[voicemail/vm-record_message.wav] (en:en) 2019-07-26 11:09:45.520231 [DEBUG] switch_ivr_play_say.c:1497 Codec Activated L16@48000hz 1 channels 20ms 2019-07-26 11:09:50.120230 [DEBUG] switch_ivr_play_say.c:1941 done playing file /usr/share/freeswitch/sounds/en/us/callie/voicemail/vm-record_message.wav 2019-07-26 11:09:50.120230 [DEBUG] switch_ivr.c:195 Codec Activated L16@48000hz 1 channels 20ms 2019-07-26 11:09:52.140233 [DEBUG] switch_ivr_play_say.c:567 Raw Codec Activated, ready to waste resources! 2019-07-26 11:09:52.140233 [DEBUG] switch_ivr_play_say.c:681 Raw Codec Activated 2019-07-26 11:09:52.140233 [DEBUG] switch_core_codec.c:223 sofia/internal/ nathan(a)stage.illumy1.com Push codec L16:100 2019-07-26 11:09:55.700231 [NOTICE] sofia.c:1079 Hangup sofia/internal/ nathan(a)stage.illumy1.com [CS_EXECUTE] [NORMAL_CLEARING] 2019-07-26 11:09:55.760231 [DEBUG] switch_core_codec.c:248 sofia/internal/ nathan(a)stage.illumy1.com Restore previous codec opus:116. 2019-07-26 11:09:55.760231 [DEBUG] switch_core_session.c:2905 sofia/internal/nathan(a)stage.illumy1.com skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2019-07-26 11:09:55.760231 [DEBUG] switch_core_state_machine.c:650 (sofia/internal/nathan(a)stage.illumy1.com) State EXECUTE going to sleep 2019-07-26 11:09:55.760231 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/nathan(a)stage.illumy1.com) Running State Change CS_HANGUP (Cur 1 Tot 72) 2019-07-26 11:09:55.760231 [DEBUG] switch_core_state_machine.c:847 (sofia/internal/nathan(a)stage.illumy1.com) Callstate Change ACTIVE -> HANGUP 2019-07-26 11:09:55.760231 [DEBUG] switch_core_state_machine.c:849 (sofia/internal/nathan(a)stage.illumy1.com) State HANGUP 2019-07-26 11:09:55.760231 [DEBUG] mod_sofia.c:460 Channel sofia/internal/ nathan(a)stage.illumy1.com hanging up, cause: NORMAL_CLEARING 2019-07-26 11:09:55.760231 [DEBUG] switch_core_state_machine.c:60 sofia/internal/nathan(a)stage.illumy1.com Standard HANGUP, cause: NORMAL_CLEARING 2019-07-26 11:09:55.760231 [DEBUG] switch_core_state_machine.c:849 (sofia/internal/nathan(a)stage.illumy1.com) State HANGUP going to sleep 2019-07-26 11:09:55.760231 [DEBUG] switch_core_state_machine.c:619 (sofia/internal/nathan(a)stage.illumy1.com) State Change CS_HANGUP -> CS_REPORTING 2019-07-26 11:09:55.760231 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/nathan(a)stage.illumy1.com) Running State Change CS_REPORTING (Cur 1 Tot 72) 2019-07-26 11:09:55.760231 [DEBUG] switch_core_state_machine.c:935 (sofia/internal/nathan(a)stage.illumy1.com) State REPORTING 2019-07-26 11:09:55.760231 [DEBUG] mod_odbc_cdr.c:345 sql INSERT INTO cdr_table_both (CallId, TEST_id) VALUES ('e1b09fde-a805-44ab-8fd1-b1ad8de4933a', 'nathan(a)stage.illumy1.com') 2019-07-26 11:09:55.800244 [DEBUG] mod_odbc_cdr.c:345 sql INSERT INTO cdr_table_a_leg (TelNumberFull, TelNumber, CallId, sip_endpoint_disposition, orig_id, CallerID, CODEC, sip_current_application, directGateway, IPInternal, IP, term_id, ClientId) VALUES ('beth', 'beth', 'e1b09fde-a805-44ab-8fd1-b1ad8de4933a', 'ANSWER', 'e1b09fde-a805-44ab-8fd1-b1ad8de4933a', 'nathan', 'opus', 'voicemail', ' stage.illumy1.com', 'a83a9a5lqgp2.invalid', '50.246.121.173', '3kol74gh1no7rp5ll228', 'e1b09fde-a805-44ab-8fd1-b1ad8de4933a') 2019-07-26 11:09:55.800244 [DEBUG] switch_core_state_machine.c:174 sofia/internal/nathan(a)stage.illumy1.com Standard REPORTING, cause: NORMAL_CLEARING 2019-07-26 11:09:55.800244 [DEBUG] switch_core_state_machine.c:935 (sofia/internal/nathan(a)stage.illumy1.com) State REPORTING going to sleep 2019-07-26 11:09:55.800244 [DEBUG] switch_core_state_machine.c:610 (sofia/internal/nathan(a)stage.illumy1.com) State Change CS_REPORTING -> CS_DESTROY 2019-07-26 11:09:55.800244 [DEBUG] switch_core_session.c:1726 Session 72 (sofia/internal/nathan(a)stage.illumy1.com) Locked, Waiting on external entities 2019-07-26 11:09:55.800244 [NOTICE] switch_core_session.c:1744 Session 72 (sofia/internal/nathan(a)stage.illumy1.com) Ended 2019-07-26 11:09:55.800244 [NOTICE] switch_core_session.c:1748 Close Channel sofia/internal/nathan(a)stage.illumy1.com [CS_DESTROY] 2019-07-26 11:09:55.800244 [DEBUG] switch_core_state_machine.c:738 (sofia/internal/nathan(a)stage.illumy1.com) Running State Change CS_DESTROY (Cur 0 Tot 72) 2019-07-26 11:09:55.800244 [DEBUG] switch_core_state_machine.c:748 (sofia/internal/nathan(a)stage.illumy1.com) State DESTROY 2019-07-26 11:09:55.800244 [DEBUG] mod_sofia.c:365 sofia/internal/ nathan(a)stage.illumy1.com SOFIA DESTROY 2019-07-26 11:09:55.800244 [DEBUG] mod_opus.c:725 Opus decoder stats: Frames[179] PLC[0] FEC[0] 2019-07-26 11:09:55.800244 [DEBUG] mod_opus.c:740 Opus encoder stats: Frames[0] Bytes encoded[0] Encoded length ms[0] Average encoded bitrate bps[0] 2019-07-26 11:09:55.800244 [DEBUG] mod_opus.c:725 Opus decoder stats: Frames[0] PLC[0] FEC[0] 2019-07-26 11:09:55.800244 [DEBUG] mod_opus.c:740 Opus encoder stats: Frames[647] Bytes encoded[74014] Encoded length ms[12940] Average encoded bitrate bps[49342] 2019-07-26 11:09:55.800244 [DEBUG] switch_core_state_machine.c:181 sofia/internal/nathan(a)stage.illumy1.com Standard DESTROY 2019-07-26 11:09:55.800244 [DEBUG] switch_core_state_machine.c:748 (sofia/internal/nathan(a)stage.illumy1.com) State DESTROY going to sleep ><> nathan stratton

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RTP (re)generation issue
by Dmitriy Borisov 29 Jul '19

29 Jul '19

Hello! Need your help! I have next scheme of RTP processing: rtpengine <-> caller freeswitch <-> robot freeswitch. And next scheme of call: 1. Caller make an outgoing call (1) via rtpengine 2. On answer caller make an outgoing call (2) to robot and bridge it with call (1) I have a problem with RTP stream rtpengine -> caller freeswitch -> robot freeswitch I see in a pcap that RTP packets goes from rtpengine to caller freeswitch without significant jitter. But sometimes caller freeswitch make a ~200ms length pause in RTP transmitting. After pause caller freeswitch renew transmitting of RTP packets, and first packet have a marker bit set. As I know from RFC 3551, marker bit means that stream starts after pause of transmitting. This pauses are very harmful for me becuse them break speech recognition process of robot freeswitch side. I use PCMA codec with ptime=20 and no any virtualization of caller freeswitch hardware. Do you have any thinks about reason of this pauses and how can I remove them? -- With best regards Dmitry Borisov

2 3

master branch no incoming audio
by Bipin Patel 28 Jul '19

28 Jul '19

hi, FS master branch has been receiving a lot of commits recently and i noticed since those PVS-Studio patches started coming in it compiles fine but there is no incoming audio on any calls even though there are no errors, tried with normal udp/rtp, tls/srtp and its same for both. Im compiling on VS2019 community on windows 10 64bit with just the SDK upgraded, toolset tried the existing one as well as tried upgrading that but same behavior for both. Is is still some work in progress or some bug? Regards, Bipin

4 13

BLF issue
by Sean Devoy 26 Jul '19

26 Jul '19

Hi, After a lot of emails to China, typically one per day/night due the time difference, Fanvil believes this problem to be with FreeSwitch. I am quite skeptical; my money is on you guys. Here is the scenario: Phone A is registered to ext 101. It also has a BLF for ext 102. Phone B is registered to 102. Almost everything works exactly as expected. When phone B gets or makes a call the BLF works. The thing that fails is ONLY when phone B calls phone A. At the end of the call, phone A's BLF for B stays red. A call in the other direction, does not produce the problem. Fanvil ran network captures and says we (you) do not send a Notify to phone A regarding Phone B's call end. My question is, does FS send the Notify regarding B after a call from B ends? If not, I am betting there is a standard somewhere that explains why it is unnecessary - clearly it should not be needed. Thanks for your time, Sean

3 3

Update DNS SRV records
by Paul Muaddib 26 Jul '19

26 Jul '19

Hi, how can I tell freeswitch to update the *SRV* records or is there a way to force the update without restarting. My provider keeps changing the entries after a few days and if I don't restart freeswitch I can not call out anymore (403) Best regards

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