- FreeSWITCH-users - FreeSWITCH Mailing Lists

Best Way To Chat / Presence / File Xfer
by Adrian Thompson 29 Nov '16

29 Nov '16

Hello, Been a user and fan of Freeswitch for two years now - thank you for the great software! I've done days of research and would like to reach out to see if anyone else has some ideas here: We are looking into giving all of our techs a softphone for their mobile device so they can communicate back to base and to each other while out in the field. Our desired features are as follows: - Video (for troubleshooting equipment with other remote techs) - Chat (for quick updates) - Group Chat (for group reminders) - Page (reach out to a busy individual with an important message) - Group Page (reach out to a busy group of workers with an important message) - File Transfer (send each other marketing material and tech manuals on the fly) - Presence (so employees can let everyone else know if they are available) Video - Check, Working Chat - Check, SIMPLE, Working Group Chat - Not an option with SIMPLE?, Not working Page - Check, Working Group Page - Check, Working File Transfer - Not an option with SIMPLE?, Not working Presence - I have BLF working great, but what about extended presence such as 'busy' or 'I'm a leprechaun' Do I have to start using XMPP for these things? It could complicate the setup three-fold if I have to go that route and I would rather not.. But if I have to, is there documents supporting a configuration that includes Freeswitch for voice/video and XMPP for everything else? Also, does anyone have a suggestion for a mobile softphone with video, chat, file xfer and can be provisioned server side? (I've tried the Grandstream Wave, but it cannot use an HTTPS server with authentication for some reason so I'm looking for more options) Thanks in advance for advice! Adrian

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all-reg-options-ping don't work
by Paul Zillmann 29 Nov '16

29 Nov '16

Hello there, I've set the options all-reg-options-ping and registration-thread-frequency on multiple gateway profile, but they don't seem to work. siptrace on those profiles don't show any activity. I've put them in the settings part of the profiles. What do I miss? How do I enable options ping on gateways (for example every 20 seconds)? Thanks,

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[Unable to get VAD event TALK]
by Bilal Abbasi 29 Nov '16

29 Nov '16

Hi users, I am currently playing with freeswitch to detect TALK event, i have the following set at profile level, <param name="vad" value="both"/> Plus i am exporting fire_talk_events=true while originate as well. Using python pyswitch self.eventsocket.apiOriginate("{execute_on_answer='transfer 05123456789 XML default',originate_timeout='120',origination_caller_id_name='123456789',origination_caller_id_number='123456789'}sofia/external/"+NUMBER+"@X.X.X.X",cidname="godson", cidnum="123" ,channelvars={'rtp_enable_vad_in':'True','rtp_enable_vad_out':'True','origination_uuid':uid,'DTMF':UUID,'fire_talk_events':'true','fire_not_talk_events':'true'}) I am getting this on fs_cli 2016-11-27 09:49:45.160380 [DEBUG] switch_rtp.c:7962 Activate VAD codec G722 20ms 2016-11-27 09:49:45.160380 [DEBUG] switch_core_media.c:6754 AUDIO RTP Engage VAD for sofia/external/05123456789(a)X.X.X.X ( in out ) I am getting all other registered events(CHANNEL_ANSWER, CHANNEL_DESTROY etc), but i am not able to get TALK event. Can someone please help me out to get the talk event. Regards Abbasi

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Management of freeswitch with event socket library
by Tezcan ÇIRAKOĞLU 28 Nov '16

28 Nov '16

Hi there, I am currently trying to develop a management console for freeswitch which is supposed to manage couple of freeswitch instances on different servers. So i am using event socket library to manage them remotely. My question is, can i edit configuration xml files (like dialplan, acl, directory etc) with inbound or outbound socket interface? Also is there a detailed documentation about commands that i can execute over sockets? Thanks for further help Tezcan

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permissions for tmute
by Jamie McClelland 28 Nov '16

28 Nov '16

Hi all - Thanks to the devs for the excellent verto code! After a year in use, I'm planning to improve my verto web app by allowing administrators to mute or kick *other* participants from the call. I'm following these instructions: https://evoluxbr.github.io/verto-docs/tut/sending-conference-commands.html Thankfully, when I tried it as a regular logged in user that has the proper permissions to join the conference, I received a permission denied message. Now I'd like to login as a user with the proper permissions to mute another conference participant, but I'm not sure how to set those permissions. The regular conference participant has the following: <param name="verto-context" value="public"/> <param name="verto-dialplan" value="XML"/> <param name="jsonrpc-allowed-methods" value="verto"/> <param name="jsonrpc-allowed-event-channels" value="conference"/> What would I add/change for a user that has permission to mute or kick another conference participant? thanks! jamie -- May First/People Link Growing networks to build a just world http://www.mayfirst.org https://support.mayfirst.org OpenPGP Key: http://current.workingdirectory.net/pages/identity/ xmpp: jamie(a)mayfirst.org

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homer archtecture
by Luis Daniel Lucio Quiroz 28 Nov '16

28 Nov '16

I know maybe this is not the best place to ask, but does anyone have a link where to read about homer archtecture.I want to understand relationship among elements before doing a blind install Regards

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Freeswitch and STUN behind NAT.
by André Maricato 28 Nov '16

28 Nov '16

I’m trying to configure Freeswitch behind NAT but on following the instructions in https://wiki.freeswitch.org/wiki/NAT_Traversal#FreeSWITCH_ behind_NAT Freeswitch starts to listen for SIP on the random a port obtained from the STUN process, instead of the configured 5080 port for the external profile. Is this expected behaviour? Shouldn't the STUN port be ignored for the SIP profile? Can it be configured in any other way? I would appreciate any input. Already tried it with a static configuration and it works, but I wanted to have a system behind a dynamic IP link which only seems plausible with STUN enabled. Thanks. -- André

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How to drop a voice mail to user who doesn't exist
by Naveen Tamanam 28 Nov '16

28 Nov '16

Hi, How to create a dialplan to facilitate to drop a voicemail to the user and domain who/which doesn't exist in freeswitch. -- Thanks & Regards, Naveen Tamanam

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SMPP Incoming Message
by Uday kumar 28 Nov '16

28 Nov '16

Hi, I have installed freeswitch 1.6.12 and SMPP on this. Configured SMPP account and able to send outgoing message from softphone but when some replying to me then message coming on freeswitch but its coming on my softphone. I followed https://quentustech.com/smpp-support-in-freeswitch.html created by by William King. Log showing same as per William included but in my case its never reach to my extension. Local extension to extension chat working but only when message coming on my DID then its not going routed extension. *My chat plan:- * <context name="default"> <extension name="smpp34_demo_in"> <condition field="to_user" expression="^DID_Number$"/> <condition field="from_user" expression="^(\d{11})$"> <action application="set" data="to=2001001(a)xx.xx.xx.xx"/> <action application="set" data="from=$1(a)xx.xx.xx.xx"/> <action application="set" data="proto="/> <action application="set" data="dest_proto=sip"/> <action application="set" data="final_delivery=1"/> <action application="info"/> <action application="send"/> </condition> </extension> </context> <context name="public"> <extension name="smpp34_demo_out"> <condition field="from_user" expression="2001001"/> <condition field="to_user" expression="^(\d{11})$"> <action application="set" data="from_user=DID_Number"/> <action application="set" data="final_delivery=1"/> <action application="info"/> <action application="smpp_send" data="brightlinkIP"/> </condition> </extension> </context> Please guide me. -- Thanks & Regard Uday Site:- www.shareyourknowledge.in Mobile:- +91-9377579349

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SIP TLS failed with FSClient 1.2.3.5
by xiyu zhao 27 Nov '16

27 Nov '16

Hi All, Please help me when you get a chance. I've follow the instruction link below to configure TLS in my freeswitch server, but it failed with my FSClient 1.2.3.5. I copied cafile.pem from my freeswitch to my windows desktop and gived the right directory under "TLS Certificate Directory" shown as below screenshot (also attached). https://freeswitch.org/confluence/display/FREESWITCH/SIP+TLS But I still cannot log in with tls, console log output, and configuration files are below. Kindly take a look and let me know if additional info is needed. P.S. I used IP instead of domain name to create the certificate, is it a problem? E.g: I used ./gentls_cert setup -cn pbx.freeswitch.org -alt DNS: 52.35.22.204 -org 52.35.22.204. Console output: tport.c:2749 tport_wakeup_pri() tport_wakeup_pri(0x7fcee8050770): events IN tport.c:862 tport_alloc_secondary() tport_alloc_secondary(0x7fcee8050770): new secondary tport 0x7fcee8252ea0 tport_type_tcp.c:203 tport_tcp_init_secondary() tport_tcp_init_secondary(0x7fcee8252ea0): Setting TCP_KEEPIDLE to 30 tport_type_tcp.c:209 tport_tcp_init_secondary() tport_tcp_init_secondary(0x7fcee8252ea0): Setting TCP_KEEPINTVL to 30 tport_type_tls.c:610 tport_tls_accept() tport_tls_accept(0x7fcee8252ea0): new connection from tls/50.187.205.251:56612/sips tport_tls.c:955 tls_connect() tls_connect(0x7fcee8252ea0): events NEGOTIATING tport_tls.c:1044 tls_connect() tls_connect(0x7fcee8252ea0): TLS setup failed (error:00000001:lib(0):func(0):reason(1)) tport.c:2090 tport_close() tport_close(0x7fcee8252ea0): tls/50.187.205.251:56612/sips tport.c:2263 tport_set_secondary_timer() tport(0x7fcee8252ea0): set timer at 0 ms because zap freeswitch@ip-172-31-28-201> sofia status Name Type Data State ============================================================================ ===================== external-ipv6 profile sip:mod_sofia@[::1]:5080 RUNNING (0) 172.31.28.201 alias internal ALIASED external profile sip:mod_sofia@52.35.22.204:5080 RUNNING (0) external::example.com gateway sip:joeuser@example.com NOREG internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) internal-ipv6 profile sip:mod_sofia@[::1]:5061 RUNNING (0) (TLS) internal profile sip:mod_sofia@52.35.22.204:5060 RUNNING (0) internal profile sip:mod_sofia@52.35.22.204:5061 RUNNING (0) (TLS) ============================================================================ ===================== 4 profiles 1 alias Under vars.xml: <X-PRE-PROCESS cmd="set" data="sip_tls_version=sslv23"/>  <X-PRE-PROCESS cmd="set" data="sip_tls_ciphers=ALL:!ADH:!LOW:!EXP:!MD5:@STRENGTH"/>  <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=true"/> <X-PRE-PROCESS cmd="set" data="internal_ssl_dir=/usr/local/freeswitch/conf/ssl"/> Under internel.xml:  <param name="tls" value="true"/>  <param name="tls-only" value="false"/>  <param name="tls-bind-params" value="transport=tls"/>  <param name="tls-sip-port" value="$${internal_tls_port}"/>    <param name="tls-passphrase" value=""/>  <param name="tls-verify-date" value="true"/>   <param name="tls-verify-depth" value="2"/>  <param name="tls-verify-in-subjects" value=""/>  <param name="tls-version" value="$${sip_tls_version}"/>  <param name="tls-ciphers" value="$${sip_tls_ciphers}"/> Thanks, Clarie

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