July 2019 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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

3 8

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

9 14

Call to unregister endpoint with auth=true after endpoint password change
by Jose David Jurado Alonso 25 Jul '19

25 Jul '19

Hi, I have FS configured to only send calls to authenticated endpoints. It works very well in general but there is a case in which it sends the call to unregistered endpoints: 1. Auth the endpoint and "registration" (ok) 2. Call to endpoint and call arrives (ok) 3. Change the endpoint password in FS DB and endpoint "fail registration" 4. Call to endpoint and call arrives (ERROR, call must be ignored) The ERROR is because the REGISTER event know that endpoint "fail registration" and yet send the call anyway ignoring the "auth_calls=true" configuration. If I execute these fs_cli commands between step 3 and 4, the call is not sent (OK for me, expected and desired case): > sofia profile external flush_inbound_reg all > sofia profile external flush_inbound_reg reboot > sofia profile external check_sync I have not tested if all the commands are necessary or not. Does anyone know how it can be done to make it work as desired using configuration without having to execute the commands after every password change in an endpoint? Relevant configuration is: vars.xml:  <X-PRE-PROCESS cmd="set" data="internal_auth_calls=true"/> <X-PRE-PROCESS cmd="set" data="internal_sip_port=5060"/> <X-PRE-PROCESS cmd="set" data="internal_tls_port=5061"/> <X-PRE-PROCESS cmd="set" data="internal_ssl_enable=false"/>  <X-PRE-PROCESS cmd="set" data="external_auth_calls=true"/> <X-PRE-PROCESS cmd="set" data="external_sip_port=5080"/> <X-PRE-PROCESS cmd="set" data="external_tls_port=5081"/> <X-PRE-PROCESS cmd="set" data="external_ssl_enable=false"/> sip_profile "external.xml" (i use it): <param name="manage-presence" value="false"/> <param name="auth-calls" value="true"/> <param name="multiple-registrations" value="true"/> I use a dynamic directory getting SIP accounts from DB and generate the XML (mod_curl) setting "cacheable=0". Thanks! José David Jurado Alonso.

4 4

Missing remote sound in recordings using OPUS codec
by Allan Kristensen 23 Jul '19

23 Jul '19

Hello, I'm missing remote sound when recording call using the Opus codec. The sound works perfectly during the call, but when listing to recording afterwards, remote sound is missing (sound from SIP UA / A-leg). When using PCMA/PCMU codec recording works fine. *To use OPUS I do:* Set 'absolute_codec_string', 'OPUS@8000h@20i@1c' *To start the recordin i do:* <action application="record_session" data="$${RECORDING_DIR}/${uuid}.wav"/> Extra info: The call is being transcoded to PCMA/PCMU to carrier when using Opus codec. Tried both v1.8 branch and master and using Opus lib 1.3-1 Does transcoding cause the problems perhaps? or is it mod_opus? Thank you... /Allan

2 1

Corrupted core.db
by Jai Rangi 23 Jul '19

23 Jul '19

We are getting this error When we do show registration switch_core_sqldb.c:1197 SQL ERR: [select * from registrations where hostname='freeswitch'] database disk image is malformed If there a way to fix this without restarting free-switch. Thank you -Jai

11 20

Call Center sound announcement
by Moshe Rosenberg 23 Jul '19

23 Jul '19

Hi i just came acorss this bug report https://freeswitch.org/jira/browse/FS-10253 while i was searching to fix my issue i see this is a few years old. did anyone find a way around this bug? -- Moshe Rosenberg

3 3

How to configure External Gateway with Multiple NICs
by Vishal Kakkar 22 Jul '19

22 Jul '19

Hi all, My freeswitch server has two NICs eth0 and eth1. eth0 - Connected to internet for public/remote SIP Gateways eth1 - Connected to ISP(non internet) SIP server directly. Now to have all external gateways working together i need to specify which sip-ip, rtp-ip to be used in each gateway. But i can see these settings can be done only in external.xml So if i specify IP of any eth0 or eth1, all external gateways (be public or local ISP) registrations go using VIA tag of that very IP address, which is causing failed registrations. If i give eth0 IP then public SIP gateways work, but not local SIP SIP. If i give eth1 IP then local ISP SIP gateway work, but not public/remote gateways. How can such situations be handled. Is there any way to specify via header/sip,rtp-ip per gateway. Thanks Vishal.

2 1