August 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

ClueCon Video Cameras for Sale!!
by Ken Rice 14 Aug '14

14 Aug '14

Hey FreeSWITCHers, Here¹s your chance to own a little Piece of ClueCon 2014. I have 2 Cannon Vixia HF R500 HD camcorder setups used only for Recording Video at ClueCon! ( http://www.amazon.com/gp/product/B00HLDFTRS/ref=oh_aui_detailpage_o05_s00?ie =UTF8&psc=1 ) Included with the Camera is a Tripod and 2 SD Cards (all were purchased brand new just to use @ ClueCon). Whats included with the camera? Battery, Charger, cables for HDMI and USB, 2 x 32G SD Cards (good for 4hours of 1080P Video per card), and a Tripod with a carrying bag for the tripod. All original Docs are included. Sorry the Original boxes were sacrificed in Chicago at the train station so I could make it home with the cameras. I am asking $350 for each Kit! (the camera alone is worth that!) Contact me off list if you have any questions or you want to add one to your list of toys from ClueCon. -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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mod_radius_cdr - unwanted connections to at&t ip
by Stefan Kainz 14 Aug '14

14 Aug '14

Hello, After a clean install of both, debian and freeswitch ( with mod_radius_cdr enabled ) i noticed a few accounting-packages sent to strange ip's on port 1813. 120.0.0.224 ( at&t ) 32.45.27.224 ( chinaunicom ) i'm pretty sure those packages originate from mod_radius_cdr. since we're not related with these companys in any way, i'm wondering why mod_radius_cdr is ( at least trying ) to sent cdr's to those ip's. here is a sample of a cdr sent to at&t: ------------------------------ /U *xxx*.*xxx*.*xxx*.*xxx*:59323 -> *120.0.0.224*:1813// //....r...`/..I....I..(.....,&0670776c-2388-11e4-aef1-8b582cc877c8......l...*xxxxxxxxxxx*....l.......l...*xxxxxxx*....l.............l.......l...default....l...*xxxxxxxxxxxx*....l...."..l...src/switch_ivr_originate.c.'..l..!2014-08-14T09:53:05.848983+0200......).......O..f/ ---------------------------- I also searched the source files of mod_radius_cdr, but sadly ( or luckily ) no match. Somebody cares to look into this? If i'm in the wrong place for this question, maybe some of you can point me in the right direction ;) have a nice day, regards,

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CDR XMLs don't get generated sometimes
by Siri MM 14 Aug '14

14 Aug '14

Hi, I am facing an unusual problem off late. My setup is as follows: I use my webserver with SQL DB, and have setup the url in xml_cdr.conf.xml for posting xml cdr information. It works all well in a normal scenario - xml cdrs are generated, and http post onto my url is also called. Now, when I try to add a new column to my database (which is by the way very large), and then restart freeswitch, I see that the cdr xmls are not getting created - http post does occur though. I don't see any CDR related error messages in the freeswitch logs, either during start up, or after a call - any pointers on how I could debug why xml cdrs are not created?

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Freetdm libsng_isdn and Redirecting Number in SETUP - no rdnis
by Adam Kuśmirek 14 Aug '14

14 Aug '14

Hi All I have FS with libsng and Sangoma A108DE conected to PBX that is redirecting calls A -> B to C. Received SETUP message: 2014-08-13 15:33:21.148143 [DEBUG] ftmod_sangoma_isdn_trace.c:224 [SNGISDN Q931] wp2 FRAME INCOMING: Prot Disc:Q.931/I.451 (0x08) Call Ref:007D (Originating side) Type:SETUP (0x5) Sending complete: Bearer Capability:Coding:ITU-T(0) TransferCap:Speech(0) TransferRate:64 Kbit/s(16) L1Prot:G.711 A-Law(3) Channel Id:No:31 Type:B-chans(3) Exclusive/Implicit Calling Party Number:69214XXXX(l:9) plan:isdn(1) type:national(2)scr:user, passed(1) pres:allowed(0) Called Party Number:77544XXXX(l:9) plan:isdn(1) type:unknown(0) Redirecting Number:773XXXX(l:7) plan:isdn(1) type:national(2)scr:user, not screened(0) pres:allowed(0)reason:Call forwarding unconditional(15) I need Redirecting Number to make correct dialplan decisions, but I cannot get it from channel variables. I hoped, that rdnis channel variable will contain this number, but info application tells nothing about it. Is there any way to get acces to this Redirecting Number information in dialplan ? FreeSWITCH Version 1.2.24+git~20140617T204847Z~9dde4b1548~64bit (git 9dde4b1 2014-06-17 20:48:47Z 64bit) libsng_isdn-7.29.0.x86_64 Thank you for any help Adam

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Callgroup intercept when using multiple FS Servers
by Errol Samuels 13 Aug '14

13 Aug '14

Hello, I have a multi-tenant PBX setup with 2+ Freeswitch 1.4.x servers sharing the same PostgreSQL 9.3.x DB behind Kamailio for Round Robin LB generally working great however, there is an issue where Callgroup intercep <https://wiki.freeswitch.org/wiki/Callgroup_intercept>t '*8' feature only works if the call is on the same server that I am registered to. For example, if I am registered to Freeswitch-1 and the inbound call I am trying to pickup via *8 is on Freeswitch-2 then it doesn't work. Has anyone got any ideas or experience with this scenario? regards, Errol

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multiple sip gateway configuration
by Louie Liu 13 Aug '14

13 Aug '14

Hi, There is requirement for me to add multiple sip gateways to Freeswitch, the first sip gateway uses the external.xml in sip profile to route traffic to Sip provider A. The second sip gateway will use a different UA to route traffic to Sip provider B. Here are my questions: 1. Do I need to create another external.xml with its own ip and port which point to the Sip Provider B? if that's case, what do I need to change in the sip profile external.xml file? 2. What is the rtp-ip and sip-ip in the external.xml file? Should that be the local IP of my sip server? 3. What do I need to change in my dialplan so that I can reference the Sip_Provider_B? an example would be good. External.xml <param name="sip-port" value="$${external_sip_port}"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="1.2.3.4"/> <param name="ext-sip-ip" value="1.2.3.4"/> External/sip_provider_A.xml <gateway name="Sip_Provider_A"> <param name="username" value="xxx"/> <param name="password" value="xxx"/> <param name="from-user" value="03xxxxxxx"/> <param name="from-domain" value="x.y.z"/> <param name="auth-username" value="yyyyy"/> <param name="realm" value="bwas02.voip.izzz.zz.au"/> <param name="proxy" value="sipconnect.voip.zzzz.zz.au"/> <param name="register" value="true"/> <param name="register-transport" value="udp"/> <param name="context" value="public"/> </gateway> External/sip_provider_B.xml <gateway name="Sip_Provider_B"> <param name="username" value="aaa"/> <param name="password" value="bbb"/> <param name="from-user" value="03xxxxxxx"/> <param name="from-domain" value="x.y.z"/> <param name="auth-username" value="yyyyy"/> <param name="realm" value="bwas02.voip.izzz.zz.au"/> <param name="proxy" value="sipconnect.voip.zzzz.zz.au"/> <param name="register" value="true"/> <param name="register-transport" value="udp"/> <param name="context" value="public"/> </gateway> Cheers, Louie

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play back and moh
by 刘宁 13 Aug '14

13 Aug '14

Hi all Sorry to trouble you ,but I really came across a problem still remaining unsolved when using freeswitch to develop some features. Assume that Bob calls Alice, before Bob here the ringtone of Alice,may be a clip of music. When bob use voip phone, after he dials, maybe there will be a bit longer time before he receives the ringtone. I want to play some music in this period. Here is the diaplan config The path is :/usr/local/freeswitch/conf/dialplan/default/call.xml Here is the content: </extension> <extension name="testlua"> <condition field="destination_number" expression="^(\d{4}\d+)$"> <action application="playback" data="voicemail/qav_video_incoming.wav"/>  <action application="lua" data="gen_dialplan_xml.lua"/> </condition> </extension> </include> The lua scripts “gen_dialplan_xml.lua" have to set some arguments for the session and finally it calls freeswitch.bridge(session,newsession) Assume that the time when entering the bridge function is t1, and the time when the ringtone of Alice arrives is t2, thus a time of (t1— t2) exists. I want to play a music to Bob during this time. But the action of “playback” and “lua” seem to be applied sequentially, that means ,the music is not played during the (t1—t2)time. I also tried <extension name="testlua"> <condition field="destination_number" expression="^(\d{4}\d+)$">   <action application="set" data="hold_music=${sound_prefix}/voicemail/qav_video_incoming.wav"/>  <action application="set" data="tmp_hold_music=${sound_prefix}/voicemail/qav_video_incoming.wav"/>  <action application="lua" data="gen_dialplan_xml.lua"/> </condition> </extension> </include> And the music file does exist, but it still did not work. How could I complete this feature?Do I have to complete this in the lua scripts? Than you in advance,looking forward to your reply.

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SOLVED: problems with mod_skypopen
by Giovanni Maruzzelli 13 Aug '14

13 Aug '14

Following the hint from ThunderZ, I can announce a rough solution (I'll document it better next week): 1) you install the skype client downloaded by install.pl in a regular desktop linux 2) on desktop you load skypopen.ko and make the sound device as per start-skype-clients script 3) you start the skype instance on desktop 4) on desktop you connect to the skype net with username and password and set the "login at startup" checkbox 5) on desktop you compile the client.c you found in configs dir of the mod_skypopen source, as per the first line of the client.c file 6) on desktop you exec the client.c program you just compiled 7) on desktop you give the skype client authorization to be connected from skypopen application and check the "no ask again" checkbox 8) on desktop you exec the client.c program again to be sure it connect no problem 9) on desktop you save a copy of the entire ~/.Skype directory and cp it on server 10) on server you use that copy as a substitute of /usr/local/freeswitch/skypopen/skype_clients_configuration_dir/skype101 to skypeN (eg you copy it N times to skype101 skype102 etc) 11) on server you modify the startup_skype_clients.sh script so it simply launches the skype client, without username and password on stdin and without --pipelogin option It takes a little time and patience, but it works, and seems robust as before. The gateway skype-FreeSWITCH conference call is back online, answering at skypeusername "freeswitch_conf_call", as always :). -giovanni

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Postgres Configuration
by Joel White 13 Aug '14

13 Aug '14

I am implementing Postgres in the core and would like to also store usernames and passwords as well. I already have a script that allows for auto provisioning of Polycom devices from a db. Does anyone know the correct settings I should have in the Postgres DB ? I am following the Freeswitch Dbh Lua instructions here https://wiki.freeswitch.org/wiki/Lua_freeswitch_dbh I just want to make sure I create the right table and columns in the database as to not have issues later. Thank you in advance

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Stash down?
by Steven Ayre 13 Aug '14

13 Aug '14

Is anyone else finding that stash.freeswitch.org is unavailable? git clone/pull commands are hanging before timing out with 502, web interface is also unavailable Confluence also seems to be down, although Jira is fine.

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