April 2019 - FreeSWITCH-users - FreeSWITCH Mailing Lists

New Debian FS vanilla install fails
by Sean Devoy 04 Mar '20

04 Mar '20

HI all, I have a new Debian 8.1 (32 bit) server for testing. I first did an apt-get install curl then per the instructions here: https://freeswitch.org/confluence/display/FREESWITCH/Debian+7 I did this: echo "deb http://files.freeswitch.org/repo/deb/debian/ `lsb_release -sc` main" >> /etc/apt/sources.list.d/freeswitch.list curl http://files.freeswitch.org/repo/deb/debian/freeswitch_archive_g0.pub | apt-key add - apt-get update apt-get install freeswitch-meta-vanilla It runs for less than a minute and dies at: Setting up freeswitch-sysvinit (1.4.19~10~1-1~jessie+1) ... Job for freeswitch.service failed. See 'systemctl status freeswitch.service' and 'journalctl -xn' for details. invoke-rc.d: initscript freeswitch, action "start" failed. dpkg: error processing package freeswitch-sysvinit (--configure): subprocess installed post-installation script returned error exit status 1 Setting up freeswitch-systemd (1.4.19~10~1-1~jessie+1) ... dpkg: dependency problems prevent configuration of freeswitch-init: freeswitch-init depends on freeswitch-sysvinit (= 1.4.19~10~1-1~jessie+1); however: Package freeswitch-sysvinit is not configured yet. dpkg: error processing package freeswitch-init (--configure): dependency problems - leaving unconfigured and ends shortly with: Errors were encountered while processing: freeswitch-sysvinit freeswitch-init E: Sub-process /usr/bin/dpkg returned an error code (1) I ran the cmds it says: systemctl status -l freeswitch.service * freeswitch.service - freeswitch Loaded: loaded (/lib/systemd/system/freeswitch.service; disabled) Active: failed (Result: start-limit) since Wed 2015-06-17 17:11:15 EDT; 18min ago Jun 17 17:11:15 fstest.devoy.us systemd[1]: freeswitch.service: control process exited, code=exited status=1 Jun 17 17:11:15 fstest.devoy.us systemd[1]: Failed to start freeswitch. Jun 17 17:11:15 fstest.devoy.us systemd[1]: Unit freeswitch.service entered failed state. Jun 17 17:11:15 fstest.devoy.us systemd[1]: freeswitch.service start request repeated too quickly, refusing to start. Jun 17 17:11:15 fstest.devoy.us systemd[1]: Failed to start freeswitch. Jun 17 17:11:15 fstest.devoy.us systemd[1]: Unit freeswitch.service entered failed state. journalctl -xn -- Logs begin at Wed 2015-06-17 16:59:45 EDT, end at Wed 2015-06-17 17:17:01 EDT. -- Jun 17 17:11:15 fstest.devoy.us systemd[1]: freeswitch.service: control process exited, code=exited status=1 Jun 17 17:11:15 fstest.devoy.us systemd[1]: Failed to start freeswitch. -- Subject: Unit freeswitch.service has failed -- Defined-By: systemd -- Support: http://lists.freedesktop.org/mailman/listinfo/systemd-devel -- -- Unit freeswitch.service has failed. -- -- The result is failed. Jun 17 17:11:15 fstest.devoy.us systemd[1]: Unit freeswitch.service entered failed state. Jun 17 17:11:15 fstest.devoy.us systemd[1]: freeswitch.service start request repeated too quickly, refusing to start. Jun 17 17:11:15 fstest.devoy.us systemd[1]: Failed to start freeswitch. -- Subject: Unit freeswitch.service has failed -- Defined-By: systemd -- Support: http://lists.freedesktop.org/mailman/listinfo/systemd-devel -- -- Unit freeswitch.service has failed. -- -- The result is failed. Jun 17 17:11:15 fstest.devoy.us systemd[1]: Unit freeswitch.service entered failed state. Jun 17 17:11:54 fstest.devoy.us kernel: perf interrupt took too long (10077 > 10000), lowering kernel.perf_event_max_sample_rate to 12500 Jun 17 17:17:01 fstest.devoy.us CRON[12430]: pam_unix(cron:session): session opened for user root by (uid=0) Jun 17 17:17:01 fstest.devoy.us CRON[12431]: (root) CMD ( cd / && run-parts --report /etc/cron.hourly) Jun 17 17:17:01 fstest.devoy.us CRON[12430]: pam_unix(cron:session): session closed for user root Did I misunderstand that Debian 8 was OK, in fact required at 1.6.n? What am I missing? Sean

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Stumped by mod_portaudio permissions issue
by Mark Haun 02 Feb '20

02 Feb '20

I built freeswitch version 1.7.0+git~20150730T192909Z~ab7f83c654~32bit for my ARM-based home server (an NVIDIA Jetson board), with the intention of using mod_portaudio with a wireless USB headset for my main home phone. It works well when running freeswitch directly as root, but when I configured freeswitch to run at system startup, mod_portaudio could no longer find any audio devices. No problem, I thought, it's a simple permissions issue. I am using the example debian init script from the wiki (this is a Jessie install with SysV init rather than systemd). It runs freeswitch as user freeswitch, group daemon using the -u and -g command-line options. So I added "freeswitch" to the audio group, then used sudo to verify that I could play sound files with "aplay" as the freeswitch user. That worked, but it did not solve the mod_portaudio failure. Thinking there must be an issue with portaudio separate from ALSA, I fetched and built the "pa_devs" utility which enumerates and prints the available audio devices. (This is basically what mod_portaudio is doing on startup.) Running as the freeswitch user, that worked too. Hmmmm. I tried running freeswitch from the command line using sudo rather than the -u and -g options, i.e. $ sudo freeswitch /usr/local/freeswitch/bin/freeswitch -nc and that worked as well! Then I started experimenting with -u and -g and discovered that $ /usr/local/freeswitch/bin/freeswitch -nc -u freeswitch allows mod_portaudio to load correctly, whereas $ /usr/local/freeswitch/bin/freeswitch -nc -u freeswitch -g daemon causes mod_portaudio to fail. Can anyone explain what's happening here? In both cases the process UID, GID, EUID, EGID, FUID, and FGID are the same (freeswitch:daemon) as displayed with "ps xao pid,uid,gid,euid,egid,fuid,fgid". Could this be a bug in freeswitch? There are a couple of threads from 2-3 years ago in the archives, where mod_portaudio permissions issues were never resolved and the solution remained a mystery. For the time being I guess this (not using -g) is a workaround, but I'd sure like to understand what is going on. Mark

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ICE/DTLS handshake
by Mirko Brankovic 25 Oct '19

25 Oct '19

HI, Has anyone experienced DTLS handshake takes 5s to get to SETUP state: > 2017-09-21 11:40:40.021178 [INFO] switch_rtp.c:3515 Changing audio DTLS > state from OFF to HANDSHAKE > 2017-09-21 11:40:45.319457 [INFO] switch_rtp.c:3172 Changing audio DTLS > state from HANDSHAKE to SETUP In network dump i see that answering side is not sending STUN for this 5s and then suddenly answers last 5 STUNs from A side. Has anyone encountered this kind of problem ? I have a pcap if necessary... Thanks, -- Regards, Mirko

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ICTFax version 4.0 , Open source Fax over Ip software
by Tahir Almas 14 Jun '19

14 Jun '19

We are pleased to announce that ICTFax 4.0 has been released developed over freeswitch , ictcore and angular framework. New features include ATA / Extension support and REST API http://ictfax.org/ictfax-4.0-released-with-ata-api-support Regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT

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ESL connection lost to FS
by Neil Patel 07 May '19

07 May '19

Hi all, I have a fresh server running FS 1.8 from package on Debian 9. It runs a python app connecting to FS using ESL that executes this code per call attempt: c = ESLconnection(str(settings.ESL_HOSTNAME), str(settings.ESL_PORT), str(settings.ESL_PASSWORD)) if not c.connected(): abort() dial_command = "originate …" ev = c.api("bgapi", str(dial_command)) c.disconnect() if ev: result = ev.serialize() logger.debug('API result = {}'.format(result)) When we ran 350K call test, 70K calls got to this code and failed the final if condition, meaning ev was None. No CDR generated. This is totally an unexpected error case for us after tens of millions of calls over last 3 years on our other servers. Any idea why this is happening? The only difference I can see is we’ve not seen this issue on FS 1.6 (built from source), so maybe it’s introduced in 1.8 (run from package)? FYI we've built pyESL from source, which is what's recommended here. Any chance there is an incompatibility between pyESL built from source and FS 1.8 installed from package? Thanks, Neil This question reproduced from Slack <https://signalwire-community.slack.com/archives/CDC8H14PN/p1555508784091900> where I didn't get a response

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Abuse of files.freeswitch.org
by Brian West 06 May '19

06 May '19

FreeSWITCHers, I have found over 2000 IP Addresses that are abusing the files repo, If you happen to be one that has been blocked please email me directly to resolve your access. This isn't about resources, it isn't about us needing mirrors, its blatant abuse. The few that have come forward and resolved their issues had no idea they were doing this, They have various scripts that have gone off the rails and they are investigating and resolving the issue. AKA When Automation Goes Wrong! Thanks, Brian -- Brian West | Co-founder and Developer Need Commercial support? email sales(a)freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&e…> Email: brian(a)freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/> [image: https://www.facebook.com/signalwireinc?src=email] <https://www.facebook.com/freeswitch> [image: https://twitter.com/freeswitch] <https://twitter.com/freeswitch>

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'To' number modifcation
by devang nathwani 06 May '19

06 May '19

Hello Team, I have idea to modify from user, contact user from dialplan variable as well as channel variables but can we modify to number by any variables? The scenario is like this, My switch's device registered as trunk to destination switch, which is having dynamic ip(not having fqdn) So I am routing the DID number from my switch to that device which is registered as trunk to destination switch, but I need that device as bridge only to connect my switch with destination switch and send the call to another destination number. How can I achieve this by any variables? Thanks in advance.

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Building on win 10 using visual studio community edition failse
by Babak Yakhchali 04 May '19

04 May '19

Hi I'm trying to build fs 1.8.5 using vs community edition. Is it possible or I need other editions of vs? build fails with: C:\Program Files (x86)\Microsoft Visual Studio\2017\Community\Common7\IDE\VC\VCTargets\Microsoft.CppCommon.targets(209,5): error MSB6006: "cmd.exe" exited with code -1073741515 Done building project "libvpx.2017.vcxproj" -- FAILED tried with both win 10 sdk 10.0.17763.0 and 8.1 thanks

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Call Drop ( we think only when originated by Mitel phone)
by Sean Devoy 30 Apr '19

30 Apr '19

FreeSWITCH Version 1.8.5-6-31281a0bf1~64bit (-6-31281a0bf1 64bit) The server is offsite and not NATed, the phones are NATed. My customer got some Mitel 6867i phones. They are working fine 95% of the time. We have narrowed it down we think but this may be anecdotal based on limited data. It appears to only happen when a call is originated from these Mitel phones. Cisco phones at the same site are rock solid. The customer reports "dropped calls". I got a Mitel 6867i and tested. After 44 test calls, I got a dropped call within seconds of being answered. Another instance where user actually gave me enough information to track it down showed the exact same thing - hangup after 7 seconds. Here is a log extract at the point where I answer the test call. sofia/external/4104440002 is the Mitel originating phone. 15:52:04.041308 [NOTICE] sofia.c:8429 Channel [sofia/internal/314] has been answered 15:52:04.041308 [DEBUG] switch_channel.c:3781 (sofia/internal/314) Callstate Change EARLY -> ACTIVE 15:52:04.041308 [DEBUG] switch_core_codec.c:248 sofia/external/4104440002(a)66.241.96.237 Restore previous codec PCMU:0. 15:52:04.041308 [DEBUG] switch_ivr_originate.c:3798 Originate Resulted in Success: [sofia/internal/314] 15:52:04.041308 [DEBUG] switch_channel.c:2047 (sofia/external/4104440002(a)66.241.96.237) Callstate Change RING_WAIT -> ACTIVE 15:52:04.041308 [DEBUG] switch_ivr_bridge.c:1782 (sofia/internal/314) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA 15:52:04.041308 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/314) Running State Change CS_EXCHANGE_MEDIA (Cur 4 Tot 1109) 15:52:04.041308 [DEBUG] switch_core_state_machine.c:653 (sofia/internal/314) State EXCHANGE_MEDIA 15:52:04.041308 [DEBUG] mod_sofia.c:646 SOFIA EXCHANGE_MEDIA 15:52:04.101306 [DEBUG] switch_rtp.c:7550 Correct audio ip/port confirmed. 15:52:07.501320 [NOTICE] sofia.c:1079 Hangup sofia/external/4104440002(a)66.241.96.237 [CS_EXECUTE] [NORMAL_CLEARING] 15:52:07.501320 [DEBUG] switch_ivr_bridge.c:917 BRIDGE THREAD DONE [sofia/external/4104440002(a)66.241.96.237] 15:52:07.521295 [DEBUG] switch_ivr_bridge.c:825 sofia/external/4104440002(a)66.241.96.237 ending bridge by request from write function 15:52:07.521295 [DEBUG] switch_ivr_bridge.c:917 BRIDGE THREAD DONE [sofia/internal/314] 15:52:07.521295 [NOTICE] switch_ivr_bridge.c:1034 Hangup sofia/internal/314 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 15:52:07.521295 [DEBUG] switch_core_state_machine.c:653 (sofia/internal/314) State EXCHANGE_MEDIA going to sleep 15:52:07.521295 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/314) Running State Change CS_HANGUP (Cur 4 Tot 1109) 15:52:07.521295 [DEBUG] switch_core_state_machine.c:847 (sofia/internal/314) Callstate Change ACTIVE -> HANGUP 15:52:07.521295 [DEBUG] switch_ivr_bridge.c:1881 sofia/internal/314 skip receive message [UNBRIDGE] (channel is hungup already) 15:52:07.521295 [DEBUG] switch_ivr_bridge.c:1884 sofia/external/4104440002(a)66.241.96.237 skip receive message [UNBRIDGE] (channel is hungup already) 15:52:07.521295 [DEBUG] switch_core_state_machine.c:849 (sofia/internal/314) State HANGUP 15:52:07.521295 [DEBUG] mod_sofia.c:443 sofia/internal/314 Overriding SIP cause 480 with 200 from the other leg 15:52:07.521295 [DEBUG] mod_sofia.c:449 Channel sofia/internal/314 hanging up, cause: NORMAL_CLEARING 15:52:07.521295 [DEBUG] switch_core_session.c:2887 sofia/external/4104440002(a)66.241.96.237 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 15:52:07.521295 [DEBUG] switch_core_session.c:2887 sofia/external/4104440002(a)66.241.96.237 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) Any ideas why this is happening? Anyone know any secrets about Mitel phones other then use TCP instead of UDP? Any ideas on how to debug this further since it happens so rarely? Thanks, Sean

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execute_on_answer
by Joli Martinez 30 Apr '19

30 Apr '19

Hello, I am trying to find more info on the application "execute_on_answer". I have a script that calls an IVR then dials a users extension. I need to wait for the calling party extension to answer before executing the rest of my script. I don't seem to find much documentation on this application. Is there another way to test if the calling party has been answered? thanks,

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