June 2007 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Authenticating calls
by Mike Murdock 08 Jun '07

08 Jun '07

So you want to get an invite, look at the headers, and if it is a toll call, authenticate the subscriber? Michael Murdock VP Operations & Development Switchmaxx Product Line 125 N. Market. Suite 1520 Wichita KS. 67202 (316) 932-2693 -----Original Message----- From: Jason Garland [mailto:jgarland@gmail.com] Sent: Thursday, June 07, 2007 10:58 PM Eastern Standard Time To: freeswitch-users(a)lists.freeswitch.org Subject: [Freeswitch-users] Authenticating calls Is there something I can put in the dialplan to authenticate some calls(Toll calls), and not others? _______________________________________________ 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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Authenticating calls
by Jason Garland 08 Jun '07

08 Jun '07

Is there something I can put in the dialplan to authenticate some calls(Toll calls), and not others?

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Unable to Configure FreeSwitch
by Dinesh Dialani 07 Jun '07

07 Jun '07

I am using freeswitch on windows platform and trying to configure it with gafachi account. I am trying to make outbound calls but the wireshark traces shows the From address given below: From: "102" <sip:103@192.168.96.57>; tag=D5NmFsgvr8ej When I dial the number with xlite , I am able to connect to other phone. Will any body tell me what wrong I did? Here are my configuration details in freeswithc.conf. <?xml version="1.0"?> <document type="freeswitch/xml"> #set "domain=abc" #set "subdomain=192.168.96.57"   <section name="configuration" description="Various Configuration"> <configuration name="switch.conf" description="Modules"> <settings>  <param name="max-sessions" value="1000"/> </settings>  <variables> <variable name="uk-ring" value="%(400,200,400,450);%(400,2200,400,450)"/> <variable name="us-ring" value="%(2000, 4000, 440.0, 480.0)"/> <variable name="bong-ring" value="v=4000;>=0;+=2;#(60,0);v=2000;%(940,0,350,440)"/> </variables> </configuration> <configuration name="modules.conf" description="Modules"> <modules>  <load module="mod_console"/> <load module="mod_syslog"/>   <load module="mod_enum"/>     <load module="mod_cdr"/>  <load module="mod_event_socket"/>        <load module="mod_portaudio"/> <load module="mod_sofia"/>    <load module="mod_bridgecall"/> <load module="mod_commands"/> <load module="mod_conference"/> <load module="mod_dptools"/> <load module="mod_echo"/>  <load module="mod_playback"/>   <load module="mod_dialplan_xml"/>  <load module="mod_g711"/> <load module="mod_gsm"/>  <load module="mod_l16"/>   <load module="mod_sndfile"/> <load module="mod_native_file"/>    <load module="mod_softtimer"/>  <load module="mod_spidermonkey"/>   <load module="mod_cepstral"/>   <load module="mod_say_en"/> </modules> </configuration> <configuration name="console.conf" description="Console Logger">   <mappings>  <param name="all" value="DEBUG"/> </mappings> </configuration> <configuration name="sofia.conf" description="sofia Endpoint"> <profiles>  <profile name="$${domain}">  <gateways> <gateway name="192.168.96.57"> /// account username *required* ///--> <param name="username" value="username"/> /// auth realm: *optional* same as gateway name, if blank /// <param name="realm" value="domain"/> /// account password *required* /// <param name="password" value="password"/> /// extension for inbound calls: *optional* same as username, if blank ///  /// proxy host: *optional* same as realm, if blank /// <param name="proxy" value="realm"/> /// expire in seconds: *optional* 3600, if blank /// <param name="expire-seconds" value="60"/> </gateway> </gateways> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU@20i"/> <param name="codec-ms" value="20"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="192.168.96.57"/> <param name="sip-ip" value="192.168.96.57"/>     <param name="accept-blind-reg" value="true"/>            </settings> </profile> </profiles> </configuration> Thanks Dinesh

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Configuration problem with gafachi account.
by Dinesh Dialani 07 Jun '07

07 Jun '07

Hi Everybody, I am using freeswitch on windows platform and trying to configure it with gafachi account. I am trying to make outbound calls but the wireshark traces shows the From address given below: From: "102" <sip:103@192.168.96.57>; tag=D5NmFsgvr8ej When I dial the number with xlite , I am able to connect to other phone. Will any body tell me what wrong I did? Here are my configuration details in freeswithc.conf. <?xml version="1.0"?> <document type="freeswitch/xml"> #set "domain=abc" #set "subdomain=192.168.96.57"   <section name="configuration" description="Various Configuration"> <configuration name="switch.conf" description="Modules"> <settings>  <param name="max-sessions" value="1000"/> </settings>  <variables> <variable name="uk-ring" value="%(400,200,400,450);%(400,2200,400,450)"/> <variable name="us-ring" value="%(2000, 4000, 440.0, 480.0)"/> <variable name="bong-ring" value="v=4000;>=0;+=2;#(60,0);v=2000;%(940,0,350,440)"/> </variables> </configuration> <configuration name="modules.conf" description="Modules"> <modules>  <load module="mod_console"/> <load module="mod_syslog"/>   <load module="mod_enum"/>     <load module="mod_cdr"/>  <load module="mod_event_socket"/>        <load module="mod_portaudio"/> <load module="mod_sofia"/>    <load module="mod_bridgecall"/> <load module="mod_commands"/> <load module="mod_conference"/> <load module="mod_dptools"/> <load module="mod_echo"/>  <load module="mod_playback"/>   <load module="mod_dialplan_xml"/>  <load module="mod_g711"/> <load module="mod_gsm"/>  <load module="mod_l16"/>   <load module="mod_sndfile"/> <load module="mod_native_file"/>    <load module="mod_softtimer"/>  <load module="mod_spidermonkey"/>   <load module="mod_cepstral"/>   <load module="mod_say_en"/> </modules> </configuration> <configuration name="console.conf" description="Console Logger">   <mappings>  <param name="all" value="DEBUG"/> </mappings> </configuration> <configuration name="sofia.conf" description="sofia Endpoint"> <profiles>  <profile name="$${domain}">  <gateways> <gateway name="192.168.96.57"> /// account username *required* ///--> <param name="username" value="username"/> /// auth realm: *optional* same as gateway name, if blank /// <param name="realm" value="domain"/> /// account password *required* /// <param name="password" value="password"/> /// extension for inbound calls: *optional* same as username, if blank ///  /// proxy host: *optional* same as realm, if blank /// <param name="proxy" value="realm"/> /// expire in seconds: *optional* 3600, if blank /// <param name="expire-seconds" value="60"/> </gateway> </gateways> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU@20i"/> <param name="codec-ms" value="20"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="192.168.96.57"/> <param name="sip-ip" value="192.168.96.57"/>     <param name="accept-blind-reg" value="true"/>            </settings> </profile> </profiles> </configuration> Thanks Dinesh

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OpenBSD compilation errors
by Bryan Vyhmeister 07 Jun '07

07 Jun '07

I am attempting to build Freeswitch on OpenBSD 4.1-stable. This particular machine happens to be macppc but the same thing happens on i386. make "OUR_MODULES=`if test -z "" ; then tmp_mods="\`grep -v "#" /usr/ local/src/freeswitch/modules.conf | sed -e "s|^.*/||" \`"; else tmp_mods="" ; fi ; mods=\`for i in $tmp_mods ; do echo $i-all ; done \`; echo $mods `" "OUR_CLEAN_MODULES=`if test -z "" ; then tmp_mods="\`grep -v "#" /usr/local/src/freeswitch/modules.conf | sed - e "s|^.*/||" \`"; else tmp_mods="" ; fi ; mods=\`for i in $tmp_mods ; do echo $i-clean ; done \`; echo $mods `" "OUR_INSTALL_MODULES=`if test -z "" ; then tmp_mods="\`grep -v "#" /usr/local/src/freeswitch/ modules.conf | sed -e "s|^.*/||" \`"; else tmp_mods="" ; fi ; mods= \`for i in $tmp_mods ; do echo $i-install ; done\`; echo $mods `" "OUR_UNINSTALL_MODULES=`if test -z "" ; then tmp_mods="\`grep -v "#" / usr/local/src/freeswitch/modules.conf | sed -e "s|^.*/||" \`"; else tmp_mods="" ; fi ; mods=\`for i in $tmp_mods ; do echo $i-uninstall ; done\`; echo $mods `" `test -n "" || echo -s` all-recursive Making all in . Compiling src/switch_apr.c ... src/switch_apr.c: In function `switch_vasprintf': src/switch_apr.c:666: warning: implicit declaration of function `vasprintf' *** Error code 1 Stop in /usr/local/src/freeswitch (line 659 of Makefile). Making all in src Making all in mod making all mod_console make: don't know how to make /usr/local/src/freeswitch/ libfreeswitch.la. Stop in /usr/local/src/freeswitch/src/mod/loggers/ mod_console. *** Error code 1 Stop in /usr/local/src/freeswitch/src/mod/loggers/mod_console (line 69 of /usr/local/src/freeswitch/build/modmake.rules). *** Error code 1 Stop in /usr/local/src/freeswitch/src/mod (line 392 of Makefile). *** Error code 1 Stop in /usr/local/src/freeswitch/src (line 294 of Makefile). *** Error code 1 Stop in /usr/local/src/freeswitch (line 1056 of Makefile). *** Error code 1 Stop in /usr/local/src/freeswitch (line 463 of Makefile). I am hoping there is some way to resolve this issue so that OpenBSD is a usable platform. I can provide more information if that would help in figuring this out. The majority of my network runs on OpenBSD and I am trying to keep it that way for several reasons. Any help would be greatly appreciated. Bryan

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socket2me questions
by Tamas Cseke 05 Jun '07

05 Jun '07

Hello, I can't build socket2me http://pastebin.freeswitch.org/3015 i installed this package ii libspandsp-dev 0.0.2pre26-1 Telephony signal processing library and i see t30_stats_s has really no these members. should i install other version of spandsp-dev, or have you any idea? other: i am trying to understand what the code do, and dont understand what is the purpose of local and remote ip. addr.sin_family = AF_INET; addr.sin_addr.s_addr = htonl(INADDR_ANY); sendaddr.sin_addr.s_addr = inet_addr(remote_ip); addr.sin_port = htons(remote_port); sendaddr.sin_family = AF_INET; sendaddr.sin_addr.s_addr = inet_addr(local_ip); sendaddr.sin_port = htons(local_port); sendaddr.sin_addr.s_addr is assigned twice...is it a typo? Thanks any help, Tamas

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Porting to Blackfin
by Shidan 05 Jun '07

05 Jun '07

Hi list, If any of you are interested in porting Freeswitch to Blackfin/uClinux. then please contact me. We have started actively working on this project and would like to set up a conference call to talk about possible collaboration with others and a roadmap. Regards, Shidan Gouran Jazinga Inc.

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CCITT V.23 Source Code
by Brian West 02 Jun '07

02 Jun '07

We need to find CCITT V.23 Modem Source Code anyone know where we can get something that is license compatible? Thanks, Brian

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Re: [Freeswitch-users] Hangup the call n seconds AFTER called party picks up
by Anthony Minessale 02 Jun '07

02 Jun '07

before you call bridge add another action that executes the sched_hangup application <action application="sched_hangup" data="+<seconds>[ <cause>]"/> so to make the call last 1 minute and end with cause SWITCH_CAUSE_ALLOTTED_TIMEOUT (which happens to be the default so you can omit it unless you need another cause.) <action application="sched_hangup" data="+60 alloted_timeout"/> Also if you omit the '+' character the value will be considered the absolute epoch time you wish the call to end. Once a call is already up you can do it from the sched_hangup api command from xmlrpc or the command cli. do a show channels to get the uuid of the call then enter sched_hangup +60 <uuid> alotted_timeout all the params besides uuid have the same meaning as the dialplan app. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com JABBER:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Mindaugas Kezys <mkezys(a)gmail.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Saturday, June 2, 2007 4:09:07 AM Subject: [Freeswitch-users] Hangup the call n seconds AFTER called party picks up  Hello, What is the equivalent in FreeSwitch for Asterisk‘s Dial command setting: S(n): Hangup the call n seconds AFTER called party picks up?? Also how to do this in application? Maybe there‘s some variable to be set? Like channel->max_call_length or similar? Thank you for your help. Regards/Pagarbiai, Mindaugas Kezys http://www.kolmisoft.com _______________________________________________ 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 ____________________________________________________________________________________ Fussy? Opinionated? Impossible to please? Perfect. Join Yahoo!'s user panel and lay it on us. http://surveylink.yahoo.com/gmrs/yahoo_panel_invite.asp?a=7

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Re: [Freeswitch-users] Dial from inside application
by Anthony Minessale 02 Jun '07

02 Jun '07

The exact equiv of what you showed would be this: static int pbx_dial(switch_core_session_t *session, void *data) { const switch_application_interface_t *application_interface; int ret; char *app_name = "bridge"; char *app_arg = (char *) data; if ((application_interface = switch_loadable_module_get_application_interface(app_name))) { application_interface->application_function(session, app_arg); return 0; } else { return -1; } } Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com JABBER:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Mindaugas Kezys <mkezys(a)gmail.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Tuesday, April 17, 2007 11:59:19 AM Subject: [Freeswitch-users] Dial from inside application  Hello, How to dial from inside application in FreeSwitch? I have such Asterisk code: static int pbx_dial(struct ast_channel *chan, void *data) { struct ast_app *app; int ret; app = pbx_findapp("Dial"); if (app) { ret = pbx_exec(chan, app, data); } else { ast_log(LOG_WARNING, "Could not find application (Dial)\n"); ret = -2; } return ret; } What is the equivalent in FreeSwitch world? Thank you for your help. Regards/Pagarbiai, Mindaugas Kezys http://www.kolmisoft.com _______________________________________________ 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 __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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