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Sofia - RFC2396 URI escape
by Visciano Giovanni 25 Nov '11

25 Nov '11

According to RFC2396 a sip URI should not contain reserved character in uri syntax (ex: #<>"). It seems sofia does not apply the escape function. For example using the cli to originate a call: > originate sofia/external/*2<1#(a)192.168.1.45 55 Result in: INVITE sip:*2<1#(a)192.168.1.45 SIP/2.0 Via: SIP/2.0/UDP 138.132.50.77:5080;rport;branch=z9hG4bK6KDeZ78yH4tFS Max-Forwards: 70 Regards Giovanni Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. -----------------------------------------------------------------------------------------------------

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condition check..
by Bhrugu Mehta 24 Nov '11

24 Nov '11

hi, all asterisk setup: exten => _X.,1,GotoIf($[${status1}=on]?n1:n2) exten => _X.,n(n1),GotoIf($[${status2}=off]?n3:n4) exten => _X.,(n2),GotoIf($[${status3}=X]?n5:n6) ... ... How do i implement in Freeswitch... Regards, -- Bhrugu Mehta Sr. S/W Engineer VOIP,Telephony Team (Asterisk, Opensips, Zaptel etc.) India

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condition chaking..
by Bhrugu Mehta 24 Nov '11

24 Nov '11

hi, all asterisk setup: exten => _X.,1,GotoIf($[${status1}=on]?n1:n2) exten => _X.,n(n1),GotoIf($[${status2}=off]?n3:n4) exten => _X.,(n2),GotoIf($[${status3}=X]?n5:n6) ... ... How do i implement in Freeswitch... Regards, -- Bhrugu Mehta Sr. S/W Engineer VOIP,Telephony Team (Asterisk, Opensips, Zaptel etc.) India

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Other-Leg-uuid before bridge
by succer110＠tiscali.it 24 Nov '11

24 Nov '11

Hi everybody! I'm executing a script with exec_on_answer and exec_on_pre_answer in a bridge application. Since the execution of my application is triggered before bridging the two legs (that's exactly hat what i was looking for) i do not have the "Other-Leg-uuid" var setted in my session's channel variable. Is there any way to get the uuid of the bleg before the bridge? <action application="export" data="nolocal:execute_on_pre_answer=lua dial.lua PRGS"/> <action application="export" data="nolocal:execute_on_answer=lua dial.lua ANSW"/> <action application="bridge" data="sofia/external/111xxxx(a)192.168.1.1"/> So: in the dial.lua i need to have the uuid of "sofia/external/1111(a)192.168.1.1". Thank you in advice! (and thank you all for freeswitch!!) E' nata indoona : chiama, videochiama e messaggia Gratis. Scarica indoona per iPhone, Android e PC

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FreeSWITCH Conference Call Today
by Michael Collins 23 Nov '11

23 Nov '11

Hello all! Today's agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2011_11_23 There have been a number of changes/additions to mod_conference, so we are going to talk about those today. Also, there are several FreeSWITCH updates to talk about. If we have time and if enough subject matter experts are available we will also talk about SIP presence. See you soon! -Michael

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Submitted a patch to correct compile error of mod_shout inFC 15 & 16
by Joegen Baclor 22 Nov '11

22 Nov '11

Jira is here http://jira.freeswitch.org/browse/FS-3711

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why does app "record_session" record extra frequency range
by Han Xin 22 Nov '11

22 Nov '11

Hi all, I have a bridged call and a home brew module which adds a media bug to the local and partner session before the bridge app in the dial plan. The original speech codec used is G722, which sampling rate is 16KHz. I add a media bug which does low pass filtering the speech to 4KHz. But the spectrum of the recorded .wav file showed that it always contains the freq range from 0K to 8K. The desired spectrum should contain only 0 to 4KHz. I am confused, can anyone show me some hints? Thanks in advance. Best Regards, Han --

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Wednesday Conf Call - SIP Presence?
by Michael Collins 22 Nov '11

22 Nov '11

Hi all, I would like to have someone speak on the subject of SIP presence this coming Wednesday. If you are up for giving the community a brief introduction to the concepts involved with SIP presence please let me know. Also, if you are experienced and/or knowledgeable on the subject but would rather not give a presentation also let me know as you may be in a position to type up some material that could be used by the presenter. Please email me off list. Thanks all! -MC

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Connect Voice mail flow before sending Answer.
by Sanath Prasanna 18 Nov '11

18 Nov '11

Hi all, I am new to Freeswitch & need to do following. When called party not available, I need to route call to voice mail system without sending answer.(or OK) If send answer, calling party will be charged. So I need to avoid that. After that I need to prompt message as "Press 1 for go to voicemail or Press 2 to exit" If user press1 then send answer (Charging is start) & connect to voice mail call flow. For doing above, shall I need to do code level changes ?? Pls advice. Br, Sanath

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Loading an IVR menu on the fly through an URL
by RaviRaj Mulasa 18 Nov '11

18 Nov '11

Hi FreeSWITCH Enthusiastics/Gurus Requirement 1. Customers can build a custom IVR using a UI. 2. Back end code persists the IVR details in a DB. 3. A URL will be exposed and will return the a XML file in the format as follows <menu name="CUSTOMER_ID_ivr" greet-long="phrase:demo_ivr_main_menu" greet-short="phrase:demo_ivr_main_menu_short" invalid-sound="/mnt/callControlStorage/en/invalidSelection.wav" exit-sound="/mnt/callControlStorage/en/tollFreeForwardingDotCom.wav" tollFreeForwardingDotCom timeout ="10000" inter-digit-timeout="2000" max-failures="3" digit-len="4"> <entry action="menu-exec-app" digits="1" param="bridge sofia/$${domain}/888(a)conference.freeswitch.org"/<mailto:sofia/$$%7bdomain%7d/888@conference.freeswitch.org%22/>> <entry action="menu-exec-app" digits="2" param="transfer 9996 XML default"/>  <entry action="menu-exec-app" digits="3" param="transfer 9999 XML default"/>  <entry action="menu-sub" digits="4" param="3107767546_ivr_submenu"/>  <entry action="menu-exec-app" digits="5" param="transfer 1234*256 enum"/>  <entry action="menu-exec-app" digits="/^(10[01][0-9])$/" param="transfer $1 XML default"/>  <entry action="menu-top" digits="9"/>  </menu> 4. Load and start executing the IVR 'CUSTOMER_ID_ivr' from events socket on the fly via HTTP URI. Gave it a shot with current setup ivr.conf.xml <configuration name="ivr.conf" description="IVR menus"> <menus> <X-PRE-PROCESS cmd="include" data="/mnt/<<Folder Name for the NETWORK DIRVE mapped>>/ivr_menus/*.xml"/> </menus> </configuration> The ivr.conf.xml is auto loaded at startup of FreeSWITCH. To load any new IVR menus , we need to call API command reloadxml which reloads the whole XML configs in autload_configs folder. The reloadxml might take long time based on the number of XML files present in the folder "/mnt/<<Folder Name for the NETWORK DIRVE mapped>>/ivr_menus" Good /Hope to have for the IVR menu 1. Load /Unload IVR menus on the fly using API command(s) freeswitch@localhost> loadivr << URL to fetch the XML describing the IVR menu(s)>> freeswitch@localhost> unloadivr <<IVR menu name>> 2. From the event socket SendMsg <uuid> call-command: execute execute-app-name: ivr execute-app-arg: <<MAIN_MENU to start execution>>SPACE <<URL to fetch the XML describing the IVR menu(s)>> Educated/Wild Guess We might be able to play IVR menu on the fly by combining the mod_xml_curl and dialplan tool - ivr. Please let us know how can we achieve the functionality , we are more interested in the second approach(Event Socket) under Good /Hope to have for the IVR menu. Thanks RaviRaj.

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