- FreeSWITCH-users - FreeSWITCH Mailing Lists

Sequential enterprise originate
by Gregor Nanger 03 Dec '17

03 Dec '17

Is it possible to create enterprise originate in sequential order? I want to create gateway failover, but each failover with own cdr post, like enterprise originate, where each thread creates new cdr. Proposed failover with | creates only one cdr. Thanks for any hint Gregor -- Gregor Nanger *CTO* t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia • www.infomedia.si

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Presence BLF problems on FreeSWITCH 1.6.19
by Shaun Stokes 01 Dec '17

01 Dec '17

Does anyone have experience with presence on FreeSWITCH when under load? We've considered offloading presence to a Kamailio proxy but it would great if we could understand the limitations of presence on FreeSWITCH and recommended configuration for best performance under load. We've been experiencing issues with presence (BLF) intermittently not working or being delayed by 5 minutes or more. The BLF lights may show an extension as available when they're on the phone, as ringing when they're available etc. This problem can only be re-produced on a system which has been under load but it doesn't occur immediately unless the system has been running for a couple of weeks. This effects multiple SIP profiles in a multi-tenant (multi domain) environment with roughly 800 registrations, 500 registrations on one SIP profile and 300 on another. The FreeSWITCH DB files use a 512MB RAM disk and we've balanced extensions across multiple SIP profiles using separate (not shared) DB files to distribute load. What we've noticed is the internal DB files (i.e. sofia_reg_internal.db) appear to increase exponentially over time after roughly 22MB we often start to run into problems with presence while the system is under load, restarting FreeSWITCH and flushing the internal DB files clears the problem. Interestingly we didn't have this problem on FreeSWITCH 1.4 We've recently adjusted the following settings on our internal SIP profiles and are continuing to monitor: force-subscription-expires [900] -> [1800] sip-subscription-max-deviation [300] -> [600] max-proceeding [1000] -> [5000] initial-event-threads [2] -> [4] Here is a snippet of our internal SIP profile configuration: nonce-ttl [60] outbound-codec-prefs [G722,PCMA,H264] pass-rfc2833 [true] record-path [/path/freeswitch/recordings] record-template [${domain_name}/archive/${strftime(%Y)}/${strftime(%b)}/${strftime(%d)}/${uuid}.${record_ext}] multiple-registrations [contact] inbound-reg-force-matching-username [true] inbound-reg-in-new-thread [true] initial-event-threads [4] local-network-acl [localnet.auto] manage-presence [true] unregister-on-options-fail [true] tls-version [tlsv1.2] tls-cert-dir [/path/freeswitch/certs] tls-bind-params [transport=tls] tls [true] stun-enabled [false] stun-auto-disable [false] sip-trace [no] sip-subscription-max-deviation [600] sip-port [xxxx] sip-ip [x.x.x.x] sip-force-expires [900] sip-expires-max-deviation [300] sip-capture [no] rtp-timeout-sec [0] rtp-timer-name [soft] rtp-ip [x.x.x.x] rtp-hold-timeout-sec [0] tls-verify-in-subjects [] tls-verify-depth [2] tls-verify-date [true] tls-sip-port [xxxx] tls-passphrase [] tls-only [false] inbound-codec-prefs [G722,PCMA,H264] inbound-codec-negotiation [greedy] log-auth-failures [true] user-agent-string [FreeSWITCH] watchdog-enabled [no] watchdog-event-timeout [30000] watchdog-step-timeout [30000] ext-rtp-ip [x.x.x.x] accept-blind-auth [false] accept-blind-reg [false] aggressive-nat-detection [true] apply-inbound-acl [domains] apply-nat-acl [nat.auto] auth-all-packets [false] auth-calls [true] challenge-realm [auto_from] context [public] debug [0] dialplan [XML] dtmf-duration [2000] dtmf-type [rfc2833] enable-timer [true] ext-sip-ip [x.x.x.x] registration-thread-frequency [240] rfc2833-pt [101] force-subscription-expires [1800] forward-unsolicited-mwi-notify [false] hold-music [local_stream://default] nat-options-ping [true] NDLB-force-rport [safe] max-proceeding [5000] ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________

4 9

FS for only Media processing.
by Sharath Kumar 30 Nov '17

30 Nov '17

Hello, Is it possible to use FS just for RTP/codec processing ? I don't want to use SIP/mod_sofia ? the SIP is terminated on a different box but I would like to decode the RTP/codecs on a box running FS ? Is it too difficult to write a custom module to do this ? is it too tightly coupled with mod_sofia ? Any ideas will be greatly appreciated ? thank you, Shaks.

7 16

Endpoint dev question
by Adrian Worutowicz 30 Nov '17

30 Nov '17

Hello, I have developed an endpoint which I call X1. Its objective is sending audio to / receiving audio from an external system. I originate a call, bridging extension 7550 with X1: bgapi originate {origination_caller_id_number=00000002}sofia/external/7550(a)192.168.1.200 <mailto:%7borigination_caller_id_number=00000002%7dsofia/external/7550@192.1 68.1.200> &bridge(X1/00000002) Extension 7550 is ringing and I answer it. [...] 91407aa0-792a-4f10-bc4e-40d143dfc2e2 2017-11-27 14:44:42.971030 [DEBUG] switch_core_state_machine.c:643 (X1/83688055) State ROUTING going to sleep 91407aa0-792a-4f10-bc4e-40d143dfc2e2 2017-11-27 14:44:42.971030 [DEBUG] switch_core_state_machine.c:584 (X1/83688055) Running State Change CS_CONSUME_MEDIA (Cur 2 Tot 4) 91407aa0-792a-4f10-bc4e-40d143dfc2e2 2017-11-27 14:44:42.971030 [DEBUG] switch_core_state_machine.c:662 (X1/83688055) State CONSUME_MEDIA 91407aa0-792a-4f10-bc4e-40d143dfc2e2 2017-11-27 14:44:42.971030 [DEBUG] switch_core_state_machine.c:662 (X1/83688055) State CONSUME_MEDIA going to sleep 2017-11-27 14:44:43.001033 [DEBUG] switch_rtp.c:7229 Correct audio ip/port confirmed. Here the audio sent to FreeSwitch is perfectly reproduced at the ext. 7550, while one received is more or less broken (there are many missing frames). I try to execute: switch_channel_t *channel = switch_core_session_get_channel(session); switch_channel_answer(channel); ... but it has no effect at all. Exactly after 60s FreeSwitch hangs up. 91407aa0-792a-4f10-bc4e-40d143dfc2e2 2017-11-27 14:45:42.000933 [NOTICE] switch_ivr_originate.c:3605 Hangup X1/83688055 [CS_CONSUME_MEDIA] [NO_ANSWER] What do I do wrong? May be I should originate a call to the extension 7550, then originate a call to my endpoint X1 and then bridge both? I'll appreciate very much any hint. Adrian.

1 0

Call quality troubleshooting best practices
by Jacob Greene 30 Nov '17

30 Nov '17

Can someone point me to any resources for troubleshooting call quality issues using freeswitch CDR or logs, or whatever tool I may be missing? We are having a hard time time pin pointing where intermentent call quality issues source from, and running a packet capture for long periods of time on our server and the customer firewall seems clunky. I was wondering what the pros are doing when users report issue with call quality. Thanks in advance!!

4 3

Freeswitch - SIP profiles not working
by Achintha 29 Nov '17

29 Nov '17

Dear All, We configured freeswitch (1.6.19-36-7a77e0b~64bit) on debian 8.9. it handled around 7000 calls per day. We use the following modules. mod_xml_curl : Dynamic Dialplans (from rest Service) mod_json_cdr : for CDR (from rest Service) and one module developed by our self (mod_ards) Server spec are 4-cors and 8GB ram. Memory usage details: used : 7.7G cached :7.2G free :156M On peak hours system handles more than 65 concurrent calls and at that time the freeswitch service used CPU over than 110%. Since the cached memory usage is high we had to clear the cache memory. From the moment the cached memory was cleared outbound calls were not recorded for more than 30 mins yet the dial plan contains the path to record the outbound clips. Anyways after 6 days sip profiles were not working. In freeswith log it shows that sofia is deleted [2017-11-29 03:42:31.794428 [NOTICE] switch_loadable_module.c:950 Deleting Endpoint 'sofia']. In order to make the freeswitch function properly we had to restart it manually. Please help me to solve this issue. Freeswitch logfile : https://pastebin.freeswitch.org/view/d22f1bb8 -- Best Regards.. Achintha Udukumbura

2 1

Personalized languages for conference locutions.
by Miguel Jesús López Valverde 29 Nov '17

29 Nov '17

Hello Newly: I send you this email to transmit a new query. I want to hold conferences where each user has the personalized language according to the landline number they call. For this, I have tried different options but none of them is offering me the expected result. I have tried to set the default_language channel variable and answer the answering call before calling the conference command, on this mode, the rest of the locutions, phrase-macros, etc, work correctly in this way, but when entering the conference, only the locutions in the language defined in the configuration of vars.xml through <X-PRE-PROCESS cmd="set" data="sound_prefix=$${sounds_dir}/en/us/callie"/> have effect. I can change this sound_prefix value but all conference locutions have the same new language, not affecting the value given for default_language. I also tried calling the conference command as follows, but it also has no effect on the setting of the language in the locutions given by the conference application: <action application="conference" data="{default_language=${default_language}} conference_room@default++flags{${conf_flags}}"/> Do you know how to offer personalized languages according to a channel variable to the users of a conference? Thank you a lot! <http://www.smartic.es/> Miguel J. López Valverde Dpto. Técnico. _____ Pº de la Castellana, 135 7ª pl 28046 Madrid España Tel: 900 900 368 | Móvil: (+34) 667 772 911 E-mail: <mailto:mjlopez@smartic.es> mjlopez(a)smartic.es | Web: <http://www.smartic.es/> http://www.smartic.es _____ <https://www.linkedin.com/company/consulting-smartic-solutions-s.l.> <https://twitter.com/smartic_spain> <https://www.facebook.com/cosutiesmatisolutio?ref=ts> Sus datos de carácter personal (nombre, apellidos, dirección postal y de correo electrónico, etc.) son tratados para la gestión de su relación con la Entidad, así como para el envío de información sobre nuestra actividad y la de terceros relacionadas con la actividad de Consulting Smartic Solutions, S.L., CIF: B85130037, Pº de la Castellana, 135 7ª pl 28046 Madrid. Usted puede ejercer sus derechos de acceso, rectificación, cancelación y oposición dirigiéndose por escrito, con copia de un documento que acredite su identidad, a la dirección info (arroba) smartic.es. Este mensaje puede contener información confidencial. Si usted no es su destinatario, no debe leerlo, copiarlo, distribuirlo, ni hacer uso de la información que contiene. En este caso, por favor, llámenos o comuníquenoslo por escrito y borre este mensaje de su sistema --- El software de antivirus Avast ha analizado este correo electrónico en busca de virus. https://www.avast.com/antivirus

1 1

Hangup() doesn't destroy the channel
by Akshaya j 29 Nov '17

29 Nov '17

Hi, It would be great to know if some got a solution for termination of python script in hangup_hook(). I used session.destroy() at the end of the script which clears the channel, but when i use the same in Hangup_hook(), it restarts the FS server. def hangup_hook(session, what): session.consoleLog("info","hangup hook for %s!!\n\n" % what) session.unsetInputCallback() session.hangup("NORMAL_CLEARING") #session.destroy() logs : [INFO] switch_cpp.cpp:1293 file location..../usr/local/freeswitch/storage/voicemail/78.wav [NOTICE] sofia.c:1012 Hangup sofia/external/14005(a)x.x.x [CS_EXECUTE] [NORMAL_CLEARING] [INFO] switch_cpp.cpp:1293 session:recordFile() = 0 freeswitch@fscli> show channels --shows the channel is active with CS_EXECUTE

1 0

MAC OS open source client
by Jerry Chinn 29 Nov '17

29 Nov '17

Good Day. We are looking for a MAC OS open source SIP client that will support TLS and SRTP to work with FreeSwitch. Any suggestions? Jerry Chinn Telecom VoIP Specialist NAVIS More Performance. More Profit. tel 541-330-3562 www.TheNavisWay.com<http://www.thenavisway.com/> Facebook<https://www.facebook.com/theNAVISway/> | Twitter<https://twitter.com/NAVISway> | LinkedIn<https://www.linkedin.com/company/navisway> | Blog<https://www.thenavisway.com/blog>

4 3

mod_portaudio, permissions and distorted audio
by Stanislav Sinyagin 28 Nov '17

28 Nov '17

hi, first of all, I solved the issue with ALSA permissions: by default, /dev/snd/* devices are writable by "audio" group, but FreeSWITCH is launched with "-g freeswitch" by default, and that excludes the "audio" group from the list of active GID's. As a result, mod_portaudio cannot find any audio devices. A file "/etc/udev/rules.d/99-freeswitchaudio.rules" with contents: SUBSYSTEM=="sound", GROUP="freeswitch" sets all sound devices to freeswitch group. If it's an appliance that is not supposed to run desktop applications, this method is the least intrusive. Otherwise, you would have to mangle the systemd unit files, to remove the -g option from exec string. But now I have a strange issue, described here, with an audio attachment: https://freeswitch.org/jira/browse/FS-10809 The hardware platform is PCengines APU2, running Debian Jessie: http://pcengines.ch/apu2.htm It has a USB 3.0 host which is otherwise working fine. Any hints will be appreciated.

1 1