September 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Disable DTLS in Chrome Canary?
by Thomas 11 Sep '15

11 Sep '15

Hello list, Anthony mentioned that DTLS could still be disabled in Canary with a command line option, but I could not find anything. Does anyone know what that option is? Thanks, Thomas

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Replacing leading "+" in outgoing numbers
by Jörn Ott 11 Sep '15

11 Sep '15

Hello all, I am using FusionPBX and freeswitch. Incoming calls are signalled with a leading + and also stored in the snom phones call lists. If someone tries to call back an external caller, I tried to strip off the leading + on the outgoing call with the following condition: <extension name="ext.ddddd" > <condition field="destination_number" expression="^\+(\d\d\d\d\d*)$" > <action application="set" data="sip_h_X-accountcode=${accountcode}" /> <action application="set" data="sip_h_X-Tag=" /> <action application="set" data="call_direction=outbound" /> <action application="set" data="hangup_after_bridge=true" /> <action application="set" data="inherit_codec=true" /> <action application="set" data="ignore_display_updates=true" /> <action application="set" data="callee_id_number=$1" /> <action application="set" data="continue_on_fail=true" /> <action application="set" data="outbound_prefix=+" /> <action application="bridge" data="sofia/gateway/781c8f7f-5d96-4e9e-8939-423b883cb675/$1" /> </condition> </extension> Sadly, the condition field already generates compile error. Can you give me a hint, how I can strip the + from the dialled number? Kind regards Joern -- -- Jörn Ott Systemadministrator Chefkoch GmbH Joseph-Schumpeter-Allee 33 53227 Bonn Telefon +49 228 28669-50 Telefax +49 228 28669-511 joern.ott(a)ext.chefkoch.de | http://www.chefkoch.de HRB 18761, Amtsgericht Bonn Geschäftsführer: Martin Meister, Sven Giebler, Arne Wolter Ein Unternehmen von Gruner+Jahr.

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How to capture DTMF
by Dmitriy N. Borisov 11 Sep '15

11 Sep '15

Hi, All! I want to capture any DTMF from answered call to channel variable. How can I do that? -- С уважением, Борисов Дмитрий Отдел облачных технологий 1С-Рарус Тел. +7 (495) 231-20-02 доб.: 15-94 -- with best regards, Dmitriy Borisov 1C-Rarus Cloud Services tel.: +7 (495) 231-20-02 add.: 15-94

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Verto: not able to establish media with Firefox 40 with symetric NAT
by Nicolas Trangosi 11 Sep '15

11 Sep '15

Hello, I have installed Freeswitch server 1.6.0 to test Verto. I am able to make call without issues with Chrome or Firefox 39 and browser behind symetric NAT. When I try with Firefox 40 and browser behind symetric NAT, the signaling part is OK, but no media is sent or received. I have took a wireshark trace on server, and it seems that Freeswitch send STUN binding requests always to same destination (and do not try other candidates from SDP or from incoming STUN binding requests). As NAT is symetric, port discovered using ICE are wrong but STUN request sent by browser should allow freeswitch to find right host and port. Thanks for any help, Nicolas

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Dialing two destinations simultaneously hangup early
by John Nash 11 Sep '15

11 Sep '15

I need to test a case to generate following user experience .. User dials a number and when call hits freeswitch, freeswitch plays a sound file but the same time dials out to a gateway, that means user hears sound file but at the same time call is being tried by gateway and when gateway sends ring 183 or 180, Sound file stops playing and user hears ringback or any media sent by gateway . I plan to do this by using bridge application to two destinations, One is to play sound file and other to send call to gateway. But how can I hangup the call to extension which is playing Sound file, upon receiving the 180 or 183 from gateway (Second fork )..where should I look for this?

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Audio Issue with FreeSWITCH (TLS + SRTP)
by Saurabh Kumar Verma 11 Sep '15

11 Sep '15

Hi, Hope you're doing good. I'm facing an issue with having FreeSWITCH (TLS + SRTP). *Issue:* Audio not coming when I've SRTP(media) enabled on both of the soft-phone endpoints. *Observation: *Audio is coming if we don't use SRTP for media encryption. *Set-up Description:* I'm registered using TLS and able to make call through FreeSWITCH, but call is disconnected after 30 sec because there is not media flow between the endpoints. I have two softphone (we're using Linphone as soft-phone) registered on FS server with extension 1003 & 1010, call flow is like: Call Flow 1003 ------> FS Server -------> 1010 x.x.x.x - Server Public IP y.y.y.y - Server Local IP *NOTE:* *For security reasons replaced the actual IPs.* FreeSWITCH Version Information: *freeswitch@internal> version* *FreeSWITCH Version 1.4.21+git~20150901T202622Z~a223dd0236~64bit (git a223dd0 2015-09-01 20:26:22Z 64bit)* I've done some changes in default dialplan/configuration in FreeSWITCH: 1. Force FreeSWICTH to listen on TLS port only (By using this parameter *<param name="tls-only" value="true"/> *in internal as well as external profile). *freeswitch@internal> sofia status* * Name Type Data State* *=================================================================================================* * external profile sip:mod_sofia@x.x.x.x:5081 RUNNING (0) (TLS)* * external::example.com <http://example.com> gateway sip:joeuser@example.com <sip%3Ajoeuser(a)example.com> NOREG* * x.x.x.x alias internal ALIASED* * internal profile sip:mod_sofia@x.x.x.x:5061 RUNNING (0) (TLS)* *=================================================================================================* *2 profiles 1 alias* 2. Use the ODBC driver for driver (By uncomment this ** in config files) In Linphone soft-client, these are the changes has been done: 1. *TLS *as Transport. 2. *SRTP* in Media encryption. In FreeSWITCH, these are the registration logs showing I'm softphones are using TLS *freeswitch@internal> sofia status profile internal reg* *Registrations:* *=================================================================================================* *Call-ID: xmuVdotDkb* *User: 1003(a)x.x.x.x* *Contact: "" <sip:1003@180.151.83.178:63093;app-id=622464153529;pn-type=google;pn-tok=APA91bF-MsdZGvDi951jjCoTDSJc2reyR2JvYHlgtwpeE1vLAlG2zrOIrBmwzx6PPPAHKfcE8aqaOHFtYFoLVztBeqAqz9Cr6D6waN7VCWSIQk7dAdm9HXQ;transport=tls>* *Agent: LinphoneAndroid/2.4.1-28-g98516d9 (belle-sip/1.4.1)* *Status: Registered(TLS)(unknown) EXP(2015-09-09 11:16:30) EXPSECS(3137)* *Ping-Status: Reachable* *Host: ip-y-y-y-y.ec2.internal* *IP: 180.151.83.178* *Port: 63093* *Auth-User: 1003* *Auth-Realm: x.x.x.x* *MWI-Account: 1003(a)x.x.x.x* *Call-ID: 6B25YNRXb5* *User: 1010(a)x.x.x.x* *Contact: "" <sip:1010@180.151.83.178:13916;app-id=622464153529;pn-type=google;pn-tok=APA91bFqivAK_KIDpU_6PM0pf0U8rx9DOKm0vhyNRqjE1Dpq_uPRbTbT-BMwxNP5NmEyCMfnKxa-fjEhI2J-lzLkCcfFphO1hL39cE4VNqAnnfDbVeQbvmQ;transport=tls>* *Agent: LinphoneAndroid/2.4.1-28-g98516d9 (belle-sip/1.4.1)* *Status: Registered(TLS)(unknown) EXP(2015-09-09 11:11:31) EXPSECS(2838)* *Ping-Status: Reachable* *Host: ip-y-y-y-y.ec2.internal* *IP: 180.151.83.178* *Port: 13916* *Auth-User: 1010* *Auth-Realm: x.x.x.x* *MWI-Account: 1010(a)x.x.x.x* *Total items returned: 2* *=================================================================================================* For an active call I can see in the channels table, I'm able to see secure parameter is set during the call for both of the call legs. *freeswitch=# select * from channels ;* * uuid | direction | created | created_epoch | name | state | cid_name | cid_num | ip_addr * * | dest | application | application_data | dialplan | context | read_codec | read_rate | read_bit_rate | write_codec | write_rate | write_bit_rate | secure * * | hostname | presence_id | presence_data | callstate | callee_name | callee_num | callee_direction | call_uuid | sent_callee_name | sen* *t_callee_num | initial_cid_name | initial_cid_num | initial_ip_addr | initial_dest | initial_dialplan | initial_context * *--------------------------------------+-----------+---------------------+---------------+------------------------------------------+-------------------+----------------+---------+------------* *----+------+-------------+------------------------+----------+---------+------------+-----------+---------------+-------------+------------+----------------+----------------------------------* *-+------------------------------+-------------------+---------------+-----------+---------------+------------+------------------+--------------------------------------+------------------+----* *-------------+------------------+-----------------+-----------------+--------------+------------------+-----------------* * 485d07d6-56da-11e5-ac1a-a53fe62ce2f9 | outbound | 2015-09-09 10:05:22 | 1441793122 | sofia/internal/1010@180.151.83.178:13916 <http://1010@180.151.83.178:13916> | CS_EXCHANGE_MEDIA | Extension 1003 | 1003 | 180.151.83.* *178 | 1010 | | | XML | default | opus | 48000 | 0 | opus | 48000 | 0 | srtp:sdes:AES_CM_256_HMAC_SHA1_80* * | ip-172-31-42-34.ec2.internal | 1010(a)x.x.x.x | | ACTIVE | Outbound Call | 1010 | SEND | 47fa445c-56da-11e5-abf5-a53fe62ce2f9 | Extension 1003 | 100* *3 | Extension 1003 | 1003 | 180.151.83.178 | 1010 | XML | default* * 47fa445c-56da-11e5-abf5-a53fe62ce2f9 | inbound | 2015-09-09 10:05:22 | 1441793122 | sofia/internal/1003(a)x.x.x.x | CS_EXECUTE | 1003 | 1003 | 180.151.83.* *178 | 1010 | bridge | user/1010(a)x.x.x.x | XML | default | opus | 48000 | 0 | opus | 48000 | 0 | srtp:sdes:AES_CM_128_HMAC_SHA1_80* * | ip-172-31-42-34.ec2.internal | 1003(a)x.x.x.x | | ACTIVE | Outbound Call | 1010 | SEND | 47fa445c-56da-11e5-abf5-a53fe62ce2f9 | Outbound Call | 101* *0 | 1003 | 1003 | 180.151.83.178 | 1010 | XML | default* *(2 rows)* Attached herewith the complete FreeSWITCH log. Please let me know if any other information is required. Any help from your side would be really appreciated. Seeking some guidance & help from FreeSWITCH team. -- *Thanks:* Saurabh Kumar Verma VVDN Technologies Pvt Ltd *Cell* : +91 7042378747 | *Skype* : saurabh.verma001

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FreeSWITCH Friday FreeForAll Reminder!
by Ken Rice 11 Sep '15

11 Sep '15

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888@conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon

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Using Lua to extract TOD conditions from Database‏ -How to do it with multiple entries.
by Jonathan Hunter 11 Sep '15

11 Sep '15

Hi Guys, Wonder if anyone can help. I am testing with time of day routing, and I want to be able to check yearly/monthly conditions before weekly conditions are picked up. I can do this successfully by just inputting XML into the dialplan, and works as expected, my issue is now when extracting the detail required from a database using lua. I for example have 3 entries below for different times of the year; +------------+-------+------+----------------+ | dayofmonth | month | year | dayofmonthtime | +------------+-------+------+----------------+ | 25 | 12 | 0 | 0 | | 1 | 1 | 2016 | 540-1050 | | 10-18 | 9 | 2015 | 600-840 | So when creating directly using xml I get; table.insert(xml, [[<condition mon="12" mday="25" break="on-true">]]); table.insert(xml, [[<action application="set" data="TOD=sales" inline="true"/>]]); table.insert(xml, [[<action application="set" data="Status=closed inline="true"/>]]); table.insert(xml, [[</condition>]]); table.insert(xml, [[<condition year="2016" mon="1" mday="1" break="on-true">]]); table.insert(xml, [[<action application="set" data="TOD=sales" inline="true"/>]]); table.insert(xml, [[<action application="set" data="Status=closed inline="true"/>]]); table.insert(xml, [[</condition>]]); table.insert(xml, [[<condition year="2015" mon="9" mday="10-18" break="on-true">]]); table.insert(xml, [[<action application="set" data="TOD=sales" inline="true"/>]]); table.insert(xml, [[<action application="set" data="Status=closed inline="true"/>]]); table.insert(xml, [[</condition>]]); However my issue is when my lua script extracts them, it loops through the 3 entries (each row) , so that only the 3rd entry is available for on-wards routing as it is populating the dialplan 3 separate times. Can someone let me know what Im doing wrong? The data return is fine and I can implement lua and freeswitch for all other scenarios, its just this one which I am having issue with. Im just doing a query as; local todnew_query = string.format("select * from timeofday_new where todname='sales' and context_name='xx.xx.xx' order by priority") assert (dbh:query(todnew_query,function(todresult) And using the result with some if statements, I wondered if anyone had any advice on this? Thanks Jon

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mod_rtmp concurrent calls on same session
by Paul Cuttler 11 Sep '15

11 Sep '15

I was wondering if it's possible to have more than one call running in mod_rtmp without logging in, that is, more than one call running on the one session. I'm getting an error with an invalid chunksize that is terminating the TCP socket. thanks, Paul

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fail to add opus to freeswitch
by Assaf Dahary 11 Sep '15

11 Sep '15

Hi, When I installed freeswitch 1.4 long time ago I've commented out #mod_opus because I use SILK only. Now I would like to add the opus codec so I enabled it in the modules.conf and try to compile with: /usr/local/src/freeswitch/make mod_opus-install But it fails with error: Unknown target mod_opus-install make[2]: *** [mod_opus-install-all] Error 1 make[1]: *** [mod_opus-install] Error 2 make: *** [mod_opus-install] Error 2 I also download opus from source and compiled OK but still fail to 'load mod_opus' from cli: switch_loadable_module.c:1391 Error Loading module /usr/local/freeswitch/mod/mod_opus.so How can I add the opus codec? Regards Assaf

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