August 2019 - FreeSWITCH-users - FreeSWITCH Mailing Lists

mod call center tier behaviour
by abubacker 04 May '20

04 May '20

I am working in mod_callcenter Before my call center configuration is like this /referene : callcenter.conf.xml/ /<agents> <agent name="1000@default" type="callback" contact="[call_timeout=10]user/1000" status="Available" max-no-answer="0" wrap-up-time="0" reject-delay-time="1" busy-delay-time="1" /> <agent name="1001@default" type="callback" contact="[call_timeout=10]user/1001" status="Available" max-no-answer="0" wrap-up-time="0" reject-delay-time="1" busy-delay-time="1" /> <agent name="1003@default" type="callback" contact="[call_timeout=10]user/1006" status="Available" max-no-answer="0" wrap-up-time="0" reject-delay-time="1" busy-delay-time="1" /> </agents> <tiers>  <tier agent="1000@default" queue="support@default" level="1" position="1"/> <tier agent="1001@default" queue="support@default" level="1" position="2"/> <tier agent="1003@default" queue="support@default" level="1" position="3"/>  <tier agent="1003@default" queue="secondary" level="1" position="3"/> <tier agent="1001@default" queue="secondary" level="1" position="2"/> <tier agent="1000@default" queue="secondary" level="1" position="1"/> </tiers> </configuration> / Tier list : /freeswitch@FMS-Queue> callcenter_config tier list support@default queue|agent|state|level|position support@default|1000@default|Ready|1|1 support@default|1001@default|Ready|1|2 support@default|1003@default|Ready|1|3 +OK/ When I renamed the 1003@default as 1006@default / <agents> <agent name="1000@default" type="callback" contact="[call_timeout=10]user/1000" status="Available" max-no-answer="0" wrap-up-time="0" reject-delay-time="1" busy-delay-time="1" /> <agent name="1001@default" type="callback" contact="[call_timeout=10]user/1001" status="Available" max-no-answer="0" wrap-up-time="0" reject-delay-time="1" busy-delay-time="1" /> <agent name="1006@default" type="callback" contact="[call_timeout=10]user/1006" status="Available" max-no-answer="0" wrap-up-time="0" reject-delay-time="1" busy-delay-time="1" /> </agents> <tiers>  <tier agent="1000@default" queue="support@default" level="1" position="1"/> <tier agent="1001@default" queue="support@default" level="1" position="2"/> <tier agent="1006@default" queue="support@default" level="1" position="3"/>  <tier agent="1006@default" queue="secondary" level="1" position="3"/> <tier agent="1001@default" queue="secondary" level="1" position="2"/> <tier agent="1000@default" queue="secondary" level="1" position="1"/> </tiers>/ After that I have reload the mod_callcenter http://pastebin.freeswitch.org/14810 The 1003@default is not overwritten , but 1006@default has added newly why 1003@default is still exist /freeswitch@FMS-Queue> callcenter_config tier list support@default queue|agent|state|level|position support@default|1000@default|Ready|1|1 support@default|1001@default|Ready|1|2 support@default|1003@default|Ready|1|3 support@default|1006@default|Ready|1|3 +OK/ Is it a bug or Is it a default behaviour ? -- Best regards, N.Abubacker , Associate system engineer , bk systems pvt ltd , Ph : 9144-43902701 Disclaimer:http://www.bksystems.co.in/email-policy

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FYI - new Amazon Web Services TTS option
by Nathan Neulinger 30 Apr '20

30 Apr '20

https://aws.amazon.com/blogs/aws/polly-text-to-speech-in-47-voices-and-24-l… The quality blows away everything I've tried out previously... Highly relevant option for voice prompts. -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

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dynamic music on hold for mod_conference
by Lappe, Adam 19 Mar '20

19 Mar '20

Hi all, i am trying to dynamically set the mod-sound variable in mod_conference, so I can play different music on hold to my conference rooms without having to create multiple profiles. Is there a way to set my moh-sound path while I create the conference? Or can I change the moh-sound file via the event socket in a existing conference? I think this would also lead to success. Thanks in advance, Adam

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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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Help with Lua modules
by Alex Lake 25 Sep '19

25 Sep '19

I'm slightly struggling to work out how to install Lua modules (eg luaxml and luacurl) so that they can be used by Freeswitch. I have managed to install them under Lua5.1, but AIUI, Lua5.2 is the way to go. For lua 5.1, I did: apt-get install -y lua5.1 lua5.1-dev luarocks luarocks install luaxml luarocks install luacurl CURL_INCDIR=/usr/include/x86_64-linux-gnu And that works for commandline lua, but the lua built in to Freeswitch seems to be 5.2. If I try using Luarocks to install luaxml and luacurl, I get problems suggesting that these modules aren't suitable. Should I try to recompile Freeswitch to use Lua5.1? If so, where are the instructions? Thanks!

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Teens Direct Routing - contact and record-route headers are missing
by Tulipa Frigo 06 Sep '19

06 Sep '19

Hi all, I'm trying to configure my freeswitch to send and receive calls from Teens Direct Routing . Then, I configured the SIP gateway /usr/local/freeswitch/sip_profiles/extern/gateway_teens.xml : <include> <gateway name="teens"> <param name="realm" value="sip.pstnhub.microsoft.com:5061"/> <param name="register" value="false"/> <param name="register-transport" value="tls"/> <param name="ping" value="25"/> </gateway> </include> The Teens Direct Routing only enable the sip trunk after receive and send SIP OPTIONS with success, in both direction. The OPTIONS FLOW description from Direct Routing Configure page, https://docs.microsoft.com/en-us/microsoftteams/direct-routing-configure: "The SBC sends a OPTIONS messages to Teens Direct Routing and receives a 200OK. When Direct Routing sees incoming OPTIONS, it will start sending outgoing SIP Options messages to the SBC FQDN configured in the Contact header field in the incoming OPTIONS message and the SBC reply with 200OK." The problem is that the freeswitch (SBC) is receiving the response FORBIDDEN , with reason : Q.850;cause=21;text="29175f0d-ebe8-43ea-836f-708b6d4b0d4f;Record-Route and Contact headers are missing" . Folowing is the OPTIONS msg sent from freeswitch to Teens Direct Rounting and the response: ------------------------------------------------------------------------ send 597 bytes to tls/[52.114.76.76]:5061 at 11:10:15.890560: ------------------------------------------------------------------------ OPTIONS sip:sip.pstnhub.microsoft.com:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS 177.53.17.168:5081;rport;branch=z9hG4bKg5DyZ09c7Bv2r Max-Forwards: 70 From: <sip:sip.pstnhub.microsoft.com:5061>;tag=ge55jU7NBK1rr To: <sip:sip.pstnhub.microsoft.com:5061> Call-ID: 3f0f42b7-3a09-1238-5e8a-00155d11a606 CSeq: 8385268 OPTIONS User-Agent: FreeSWITCH-mod_sofia/1.8.6+git~20190606T190249Z~93b4c92e75~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Length: 0 ------------------------------------------------------------------------ recv 545 bytes from tls/[52.114.76.76]:5061 at 11:10:16.070876: ------------------------------------------------------------------------ SIP/2.0 403 Forbidden FROM: <sip:sip.pstnhub.microsoft.com:5061>;tag=ge55jU7NBK1rr TO: <sip:sip.pstnhub.microsoft.com:5061> CSEQ: 8385268 OPTIONS CALL-ID: 3f0f42b7-3a09-1238-5e8a-00155d11a606 VIA: SIP/2.0/TLS 177.53.17.168:5081;branch=z9hG4bKg5DyZ09c7Bv2r;rport REASON: Q.850;cause=21;text="29175f0d-ebe8-43ea-836f-708b6d4b0d4f;Record-Route and Contact headers are missing" CONTENT-LENGTH: 0 ALLOW: INVITE ALLOW: ACK ALLOW: OPTIONS ALLOW: CANCEL ALLOW: BYE ALLOW: NOTIFY SERVER: Microsoft.PSTNHub.SIPProxy v.2019.7.4.9 i.EUNO.2 ------------------------------------------------------------------------ How can I add these headers into OPTIONS msg ? Thanks and Regards Tulipa Frigo

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Inbound Call Issue
by Md,Mehedi Hasan Kabir(Tanim) 04 Sep '19

04 Sep '19

Hi I am new to FreeSWITCH. I am trying to establish outbound and inbound call using Flowroute. Outbound call is working fine. But the inbound call is not working. If I dialed DID number from the phone, it says “Invalid Number”. No log in FreeSWITCH CLI. Did I miss anything or any more configuration needed? Please find my Gateway Configuration and Dialplan for Inbound call below. Gateway Configuration Path:/usr/local/freeswitch/conf/sip_profiles/external/flowroute.xml <include> <gateway name="flowroute"> <param name="username" value=“my_username”/> <param name="password" value=“my_password”/> <param name="proxy" value="sip.flowroute.com"/> <param name="expire-seconds" value="60"/> <param name="register" value="true"/> <param name="register-transport" value="udp"/> <param name="retry-seconds" value="30"/> <param name="caller-id-in-from" value="false"/> <param name="extension-in-contact" value="true"/>  </gateway> </include> Dialplan for Inbound call Path:/usr/local/freeswitch/conf/dialplan/public/00_inbound_did.xml <include> <extension name="public_did"> <condition field="destination_number" expression="^(12408446009)$">  <action application="set" data="domain_name=$${domain}"/>  <action application="transfer" data="2004 XML default"/> </condition> </extension> </include>

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mod_db values
by Sean Devoy 04 Sep '19

04 Sep '19

Hi, I am using the latest Debian distro. I am proficient with SQL. How can I access the data that has been inserted using application="db" data="insert/..."? I am fine to use either a command line login or XML dial plan? If it is by command line I will need some help getting connected/authenticated. I don't even know which database the default distro uses. Thanks, Sean

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