July 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Help with cidlookup
by Torstein Knutsen 13 Jul '12

13 Jul '12

Hi I have cidlookup partly working. Im using a norwegian service, which returns a whole lot more than just the number. Anybody here have some Ideas on how I could proceed to map "Pizza & Kina Expressen" to the calling_id_name ? Thank you! Torstein snipplet *** : freeswitch@--hidden-ip--@internal> cidlookup 22222222 { "title" : "Gule Sider firma API", "query" : " http://api.eniro.com/cs/search/basic?country=no&search_word=22222222&to_lis…", "totalHits" : 1 , "totalCount" : 1 , "startIndex" : 1, "itemsPerPage" : 1, "adverts" : [ { "eniroId" : "P10000836357" , "companyInfo" : { "companyName" : "Pizza & Kina Expressen" , "orgNumber" : null , "companyText" : null }, "address" : { "streetName" : "Vitaminveien 11 B" , "postCode" : "0485" , "postArea" : "Oslo" , "postBox" : null }, "location" : { "coordinates" : [ { "longitude" : 10.7725744944096 , "latitude" : 59.9471107465998 }, { "use" : "route", "longitude" : 10.7725744944096 , "latitude" : 59.9471107465998 } ] }, "phoneNumbers" : [ { "type" : "std" , "phoneNumber" : "22 22 22 22" , "label" : null } ], "companyReviews" : " http://www.anbefalt.no/omtale/0003292695/22222222" , "homepage" : " http://api.eniro.com/proxy/homepage/uANwPf5aVK3QsMrfdwYjz8Olp1PSJ6L1-mCsL3_…" , "infoPageLink" : " http://www.gulesider.no/firma/pizza-kina-expressen:p10000836357?search_word…" } ] } freeswitch@--hidden-ip--@internal>

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Polycom IP 501 DOA ?
by Todd Bailey 13 Jul '12

13 Jul '12

Hey all, I know this isn't the best place for phone troubleshooting, but can someone tell me how long it takes the phone to behave like it's working? when I plugged every thing in, no display, no network connection indicated at the ip switch, no dial tones, lights etc, the only indication is a popping or clicking sound in the handset. Any thoughts ?

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Receive response of an event in ESL
by Jose Miguel Sucasas Mejuto 13 Jul '12

13 Jul '12

Hello, I am trying to send an SMS using mod_sms via ESL using: require ESL; my $con = new ESL::ESLconnection("localhost", "8021", "ClueCon"); my $e = new ESL::ESLevent("custom", "SMS::SEND_MESSAGE"); $e->addHeader("to", "1004\(a)www.freeswitch.org"); $e->addHeader("from", "testing\(a)foobar.com"); $e->addHeader("sip_profile", "internal"); $e->addHeader("dest_proto", "sip"); $e->addBody(shift); $con->sendEvent($e); as it appears in mod_sms wiki. But, how can I read the result of the event?. sendRcv waits for the response, but only works with commands not with events. Thanks in advance. Regards

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Re: [Freeswitch-users] Polycom IP 501 DOA ?
by JRichey 13 Jul '12

13 Jul '12

You should see the LEDs blink and the Polycom logo on the screen followed by a boot countdown message. -----Original Message----- From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org]On Behalf Of Todd Bailey Sent: Thursday, July 12, 2012 5:28 PM To: freeswitch-users(a)lists.freeswitch.org Subject: Re: [Freeswitch-users] Polycom IP 501 DOA ? Are you implying that I should have a display or some functionality the moment power is applied? On Thu, 2012-07-12 at 20:17 -0400, Brian Foster wrote: > DOA. Sorry :( > > Brian Foster > Endigo Computer LLC > > Sent from a mobile device. > > On Jul 12, 2012 8:16 PM, "Todd Bailey" <toddb(a)toddbailey.net> wrote: > Hey all, > > I know this isn't the best place for phone troubleshooting, > but can > someone tell me how long it takes the phone to behave like > it's working? > > when I plugged every thing in, no display, no network > connection > indicated at the ip switch, no dial tones, lights etc, the > only > indication is a popping or clicking sound in the handset. > > Any thoughts ? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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Media Bypass in Mod_rtmp.
by qasimakhan＠gmail.com 12 Jul '12

12 Jul '12

Hi, A quick question if someone could help me out. Can we use mediabypass mode when using mod_rtmp? Regards, Qasim

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Security settings for FXO gateway
by ocset 12 Jul '12

12 Jul '12

Hi There is a wiki on the web which describes how to use a Grandstream FXO device with freesiwtch and recommends the following settings. How big a security hole is this creating and is there a better way to do this? Also, what would the syntax be for the "Add ip=" recommendation? *add file conf/directory/default/pstn.xml containing* <include>  <user id="Anonymous"/> </include> *change following in conf/sip_profiles/internal.xml* <param name="accept-blind-reg" value="true"/> <param name="accept-blind-auth" value="true"/> Thanks advance O

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bridge_pre_execute_bleg_app playback
by Anita Hall 12 Jul '12

12 Jul '12

Hi! I am trying to bridge the caller to the callee and play them different sound files foo.wav and bar.wav before they are "connected". Taking help from the wiki http://wiki.freeswitch.org/wiki/Variable_bridge_pre_execute_aleg_app, I put up this dialplan but it plays back the audio files to the a-leg and b-leg one after the other and not simultaneously. What will be the way to do simultaneous playback to both the legs and then bridge them? My current dialplan is this <action application="answer" /> <action application="set" data="bridge_pre_execute_aleg_app=playback"/> <action application="set" data="bridge_pre_execute_bleg_app=playback"/> <action application="set" data="bridge_pre_execute_aleg_data=foo.wav"/> <action application="set" data="bridge_pre_execute_bleg_data=bar.wav"/> <action application="bridge" data="freetdm/1/a/9512236723"/> Thanks! regards, Anita

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Unable to set effective_caller_id_number when bridging using Openzap
by Marcus Mülbüsch 12 Jul '12

12 Jul '12

Hello all, yes, for hardware reasons (very old Sangoma card with very old firmware) I have to use OpenZAP instead of freetTDM. So, when setting the effective_caller_id that value isn't used. See that part of my dialplan here: ---snip--- <action application="set" data="effective_caller_id_number=####"/> <action application="bridge" data="OpenZAP/3/a/${dialed_ext}"/> ---snip--- That configuration works with SIP, however. The line is capable to accept a change of callerid; that was tested with asterisk. Here is the relevant part of my openzap.conf.xml ---snip--- <span name="PRI_3"> <param name="mode" value="user"/> <param name="dialect" value="q931"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> </span> ---snip--- I am at my wits end and have no more idea what I'm doing wrong. Any idea is appreciated (but don't expect a reply or thank you before Tuesday, sorry) Marcus

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new freeswitch/fusionpbx user
by Terry Barnum 12 Jul '12

12 Jul '12

I'm starting to find my way around freeswitch via fusionpbx and have it running with a free DID, two X-Lite softphones and 3CXPhone on an iPhone. After creating a recording, an IVR menu and an inbound route to the IVR, a test call coming in on the DID complains it can't find the menu: 2012-07-08 11:28:36.451506 [NOTICE] switch_channel.c:926 New Channel sofia/external/<masked>@www.xxx.yyy.zzz [82327424-822d-44a4-9aa5-c20fdce2f356] 2012-07-08 11:28:36.451506 [INFO] mod_dialplan_xml.c:485 Processing <masked> <<masked>>-><masked> in context public 2012-07-08 11:28:36.471501 [NOTICE] switch_ivr.c:1729 Transfer sofia/external/<masked>@www.xxx.yyy.zzz to XML[5002@default] 2012-07-08 11:28:36.471501 [INFO] mod_dialplan_xml.c:485 Processing <masked> <<masked>>->5002 in context default 2012-07-08 11:28:36.551572 [NOTICE] mod_dptools.c:1148 Channel [sofia/external/<masked>@www.xxx.yyy.zzz] has been answered 2012-07-08 11:28:37.571514 [ERR] mod_dptools.c:1775 Unable to find menu 2012-07-08 11:28:37.571514 [NOTICE] switch_core_state_machine.c:249 sofia/external/<masked>@www.xxx.yyy.zzz has executed the last dialplan instruction, hanging up. 2012-07-08 11:28:37.571514 [NOTICE] switch_core_state_machine.c:251 Hangup sofia/external/<masked>@www.xxx.yyy.zzz [CS_EXECUTE] [NORMAL_CLEARING] 2012-07-08 11:28:37.591511 [NOTICE] switch_core_session.c:1446 Session 1 (sofia/external/<masked>@www.xxx.yyy.zzz) Ended 2012-07-08 11:28:37.591511 [NOTICE] switch_core_session.c:1448 Close Channel sofia/external/<masked>@www.xxx.yyy.zzz [CS_DESTROY] I can see that the newly created choosevm ivr menu exists: $ sudo ls -l *.xml /usr/local/freeswitch/conf/ivr_menus/ -rw-rw---- 1 _www _www 2878 Jun 20 00:08 demo_ivr.xml -rw-r--r-- 1 _www _www 632 Jul 8 11:08 v_choosevm.xml $ sudo cat /usr/local/freeswitch/conf/ivr_menus/v_choosevm.xml <include> <menu name="choosevm" greet-long="/usr/local/freeswitch/recordings/recording1.wav" greet-short="/usr/local/freeswitch/recordings/" invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav" exit-sound="" confirm-macro="" confirm-key="" tts-engine="flite" tts-voice="rms" confirm-attempts="3" timeout="3000" inter-digit-timeout="2000" max-failures="3" max-timeouts="3" digit-len="5"> <entry action="menu-exec-app" digits="1" param="transfer 2002 XML default"/> <entry action="menu-exec-app" digits="2" param="transfer 2003 XML default"/> </menu> </include> $ sudo cat /usr/local/freeswitch/conf/dialplan/default/333_v_choosevm.xml <extension name="choosevm" > <condition field="destination_number" expression="^5002$" > <action application="answer" /> <action application="sleep" data="1000" /> <action application="set" data="hangup_after_bridge=true" /> <action application="ivr" data="choosevm" /> </condition> </extension> Is there something else necessary for the ivr menu to be found? I realize that I'm abstracted from freeswitch a bit by using fusionpbx as a front-end, but is this still the right place to ask? Thanks, -Terry

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make current fails today
by Mario G 12 Jul '12

12 Jul '12

Was ok Tuesday...... on osX. c/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DMACOSX -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -g -O2 -pipe -no-cpp-precomp -MT libfreeswitch_la-switch_core.lo -MD -MP -MF .deps/libfreeswitch_la-switch_core.Tpo -c src/switch_core.c -fno-common -DPIC -o .libs/libfreeswitch_la-switch_core.o cc1: warnings being treated as errors src/switch_core.c: In function 'send_heartbeat': src/switch_core.c:86: warning: format '%lu' expects type 'long unsigned int', but argument 5 has type 'switch_time_t' make[2]: *** [libfreeswitch_la-switch_core.lo] Error 1 make[1]: *** [all] Error 2 make: *** [current] Error 2

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