- FreeSWITCH-users - FreeSWITCH Mailing Lists

Documentation Sprint - Tomorrow (2nd October) 10am CT
by Areski 01 Oct '15

01 Oct '15

Hi Everyone, We are organizing an other Sprint to work on FreeSWITCH Documentation on Friday 2 October 2015 at 10am CT. It will be 4 hours long but you can join for less time. The Doc-sprint will focus on migrating the remaining pages from old FS Wiki (https://wiki.freeswitch.org) to Confluence ( https://freeswitch.org/confluence) We will use this IRC channel during the sprint: #freeswitch-docs and we will be tracking our work on a spreadsheet: https://docs.google.com/spreadsheets/d/1qsG-kRymvKlNBapnBLw86W130VdbnK6naYa… to avoid working on the same content. So during the sprint, please change the page ""Status"" you are working on to ""Editing"" with your name next to it. Some extra information: - https://freeswitch.org/confluence/display/FREESWITCH/Wiki+Migration - https://freeswitch.org/confluence/display/FREESWITCH/Contributing+Documenta… Let us know if you have any question. We hope to get a maximum number of people signed up! Confirmed so far: Italo (+7) Bote Areski Juan Iwada Anybody else want to join for the sprint? :) -- Kind regards, /Areski ---- Arezqui Belaid, <areski(a)gmail.com> Founder at Star2Billing (www.star2billing.com) Tel: +34650784355 Twitter: http://twitter.com/areskib LinkedIn: http://www.linkedin.com/in/areski

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Replacing Abyss Webserver
by Abdul Hakeem 01 Oct '15

01 Oct '15

Hello Experts, Could someone point me to steps on how to replace the Abyss webserver with NGX/OpenResty or any other webserver ? Cheers, Abdul Hakeem

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Music on hold, send SDP
by Miha 30 Sep '15

30 Sep '15

Hi, how can I forward SDP that has indication of "hold" (sdp: c= 0.0.0.0). I need that FW forwards this SDP further and not change it to its ip? tnx miha

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WSS/Sip over Websocket - Any parameter that controls CHIPERS suites?
by Victor Medina 30 Sep '15

30 Sep '15

Hi guys! Is there any parameter that can configure what ciphers are used on the WSS interface? Im am getting... WSS interface: SSL-Session: Protocol : TLSv1.2 Cipher : AES256-GCM-SHA384 SIP interface, same channel: Expansion: NONE SSL-Session: Protocol : TLSv1.2 Cipher : ECDHE-RSA-AES256-GCM-SHA384 -- Víctor E. Medina M. Platform Architect / Chief Infrastructure +58424 291 4561 BB #79A8AFA2 @VMCibersys

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Use of execute_extension
by Dominique Jeannerod 30 Sep '15

30 Sep '15

Hello, i'm trying to call an extension, to normalize the called number to FR format, needed by an external operator SIP trunk, using the sip_req_user variable, and set a variable iv_dest_number to bridge the call using the normalized dest number. I'd like to use extensions to split a big dialplan, and make my configuration more readable. I'm quite confused by the way the regex is returning the $1 variable inside this extension : ARRAY This same code is working without problem inside the main dialplan. What am i doing wrong ? What would be the best way to do that ? Thanks with anticipation for any advice Extension : <extension name="N_called_FR">  <condition field="${sip_req_user}" expression="^\+?(\d+)$" break="on-true"> <condition regex="any">  <regex field="${sip_req_user}" expression="^0(\d{9})$"/> <regex field="${sip_req_user}" expression="^33(\d{9})$"/> <regex field="${sip_req_user}" expression="^\+33(\d{9})$"/> <regex field="${sip_req_user}" expression="^0033(\d{9})$"/> <action application="set" data="iv_dest_number=0$1"/> </condition> </condition> </extension> Freeswitch log : EXECUTE sofia/external_trk/33123456789(a)10.199.16.172 execute_extension(N_called_FR XML iv_common_N_called) 2015-09-29 12:09:34.644195 [INFO] mod_dialplan_xml.c:635 Processing Test <33123456789>->N_called_FR in context iv_common_N_called Dialplan: sofia/external_trk/33123456789(a)10.199.16.172 parsing [iv_common_N_called->N_called_FR] continue=false |--- Dialplan: Processing recursive conditions level:1 [N_called_FR_recur_1] require-nested=TRUE |--- Dialplan: sofia/external_trk/33123456789(a)10.199.16.172 Regex (FAIL) [N_called_FR_recur_1] ${sip_req_user}(0698778763) =~ /^([1-9]\d{8})$/ match=any |--- Dialplan: sofia/external_trk/33123456789(a)10.199.16.172 Regex (PASS) [N_called_FR_recur_1] ${sip_req_user}(0698778763) =~ /^0(\d{9})$/ match=any |--- Dialplan: sofia/external_trk/33123456789(a)10.199.16.172 Action log(INFO N_called_FR : FR Called number) |--- Dialplan: sofia/external_trk/33123456789(a)10.199.16.172 Action set(iv_dest_number=0RRAY::04) Dialplan: sofia/external_trk/33123456789(a)10.199.16.172 Regex (PASS) [N_called_FR] ${sip_req_user}(0698778763) =~ /^\+?(\d+)$/ break=on-true 2015-09-29 12:09:34.644195 [NOTICE] switch_core_session.c:2866 Execute log(INFO N_called_FR : FR Called number) EXECUTE sofia/external_trk/33123456789(a)10.199.16.172 log(INFO N_called_FR : FR Called number) 2015-09-29 12:09:34.644195 [INFO] mod_dptools.c:1670 N_called_FR : FR Called number 2015-09-29 12:09:34.644195 [NOTICE] switch_core_session.c:2866 Execute set(iv_dest_number=0RRAY::04) EXECUTE sofia/external_trk/33123456789(a)10.199.16.172 set(iv_dest_number=0RRAY::04) 2015-09-29 12:09:34.644195 [DEBUG] mod_dptools.c:1477 sofia/external_trk/ 33123456789(a)10.199.16.172 SET [iv_dest_number]=[0RRAY::04] D. Jeannerod

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FreeTDM - TDM410P - CallerID
by Luiz Fernando Softov 29 Sep '15

29 Sep '15

Hi, i'm trying to use a Digium TDM410P with FXO device (span) So, install dahdi and create the conf as follow - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - in file /usr/local/etc/dahdi/system.conf fxsks=1 echocanceller=oslec,1 - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - I already try, with echocanceller MG2, KB1, and without echocanceller - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - in file /../freeswitch/conf/freetdm.conf [span zt FXO-01] trunk_type => FXO number => 9001 fxo-channel => 1 - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - in file /../freeswitch/conf/autoload_configs/freetdm.conf.xml <span name="FXO-01"> <param name="tonegroup" value="us"/> <param name="enable-callerid" value="true"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> </span> - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - But, when i can't get CallerID Its, like mod_dialplan_xml.c:635 Processing unknown <unknown>->9001 in context default In headers, i don't get info 2015-09-28 21:48:20.286947 [INFO] mod_dptools.c:1681 CHANNEL_DATA: Channel-State: [CS_EXECUTE] Channel-Call-State: [RINGING] Channel-State-Number: [4] Channel-Name: [FreeTDM/1:1/9001] Unique-ID: [20ada4c7-4366-e511-a93c-bc5ff437b74f] Call-Direction: [inbound] Presence-Call-Direction: [inbound] Channel-HIT-Dialplan: [true] Channel-Call-UUID: [20ada4c7-4366-e511-a93c-bc5ff437b74f] Answer-State: [ringing] Channel-Read-Codec-Name: [PCMU] Channel-Read-Codec-Rate: [8000] Channel-Read-Codec-Bit-Rate: [64000] Channel-Write-Codec-Name: [PCMU] Channel-Write-Codec-Rate: [8000] Channel-Write-Codec-Bit-Rate: [64000] Caller-Direction: [inbound] Caller-Logical-Direction: [inbound] Caller-Username: [FreeTDM] Caller-Dialplan: [XML] Caller-Caller-ID-Name: [unknown] Caller-Caller-ID-Number: [unknown] Caller-Orig-Caller-ID-Name: [unknown] Caller-Orig-Caller-ID-Number: [unknown] Caller-ANI: [unknown] Caller-Destination-Number: [9001] I'm from Brazil, i think here the operators send number with DTMF I'm using SPA-3000, detecting CalledID, after 3º Ring. And i use SPA-3000 to receive calls and send to FreeSwitch, but i need to use TDM410P Since in TDM410P i can use 4-FXO (or 2-FXO and 2-FXS) vs 1-FXO + 1-FXS of SPA-3000. Is there another way to to grab CalledID? Is there something missing in config? Att Luiz Fernando Softov +55 67 8123 4276 http://www.softov.com.br fernando(a)softov.com.br

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rpm.spec creates monit.d directory
by Richard Screene 29 Sep '15

29 Sep '15

Hello, The rpm.spec in the freeswitch package (at least v1.6 and master branches) will create the /etc/monit.d/ directory and copy the freeswitch.monitrc file into it. But, the monit rpm (at least monit-5.6-1.el7.x86_64)will also try to create this directory and thus produce a conflict. Given that the rpm.spec specifies no dependency on monit I suggest that the rpm.spec should not create the directory and should only copy the freeswitch.monitrc file if the destination directory (/etc/monit.d/ exists). I did try to produce a patch for this but I can’t seem to clone from the repository. I’m getting “Peer reports incompatible or unsupported protocol version.” errors. Is there any reason why it does it like this? Or, should I raise a JIRA? Regards, Richard

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MEDIA_TIMEOUT
by roy j 29 Sep '15

29 Sep '15

Hi, all We have some unclear situation. Call flow is below. 1. A --INVITE(SDP)--> FS C 2. A FS --INVITE(SDP)--> C 3. A FS <--183(SDP)-- C 4. A <--183(SDP) FS C 5. A FS <--RTP-- C 6. A <--RTP-- FS C 7. A <--480-- FS C 8. A FS --CANCEL--> C FreeSwitch (FS) sends to 'A' SIP 480 (Reason header: SIP;cause=604;text="MEDIA_TIMEOUT"). There is no RTP from 'A' while progress. Time between 4. and 7. - 30 seconds. Is this expected behaviour or we should update? Current version - FreeSWITCH Version 1.5.6b+git~20131029T184027Z~88a13f1fc2~32bit (git 88a13f1 2013-10-29 18:40:27Z 32bit)

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Getting 503 Service Unavailable while enable record_session
by bhavik patel 29 Sep '15

29 Sep '15

Hi Guys, I am using record_session applications on one of my live server, which have 100 concurrent calls are running. When i enable using dialplan then calls are dropping and getting most of call "503 Service Unavilable", so it means record_session generating load on freeswitch. I am using FreeSWITCH Version 1.4.21+git~20150901T202622Z~a223dd0236~64bit (git a223dd0 2015-09-01 20:26:22Z 64bit). Can any one please suggest me is it bug of Freeswitch ? -- Thanks, Bhavik Patel

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Porting old custom FS module (in C) to latest stable
by Imran Moinuddin 29 Sep '15

29 Sep '15

Hi all, Not sure if this is the right forum to be asking this - but we've inherited a customized Freeswitch module that was developed atop an older version (1.04), and we've been eagerly looking for ways to port this module (naturally in C) to the latest version so that we can get some of our servers off of 1.04 for good. I was curious to know if there's anyone on this list that might be able to help us in performing this code port to the latest stable ? (we'd naturally pay for this assistance) Looking forward to any recommendations. Thanks! Imran

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