Re: [Freeswitch-users] Freeswitch and XML-RPC
http://wiki.freeswitch.org/wiki/Mod_xml_rpc Plus, for streaming a file, see to it that you can use port 8080 for it for instance, if you want to read the logfile say - freeswitch.log, configure logfile.conf.xml to to generate the logs in the htdocs directory - root for xml rpc. You can definitely access it from there -- Regards, Gayatri Kulkarni On Wed, Aug 27, 2008 at 8:38 PM, Klaus Teller [EMAIL PROTECTED] wrote: Hi, It seems i can send commands to Freeswitch using XML-RPC. The only document i could find on this is: http://wiki.freeswitch.org/wiki/Freeswitch_XML-RPC Given that i'm using Java, i was wondering what else (beside originating a call) can I do via XML-RPC. How do i send for instance DTMF read requests, how do i send a request to stream a file? Any other document i could check to know what i can do with this? I would think that most of the functions that are available through Javascript are available via XML-RPC. But how do i send these commands? Thanks, Klaus. -- Pt! Schon das coole Video vom GMX MultiMessenger gesehen? Der Eine für Alle: http://www.gmx.net/de/go/messenger03 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Problem with PlayAndGetDigits
Hi Brian, Here is the script: digits = session:playAndGetDigits(1, 1, 3, 3000, #*, /audio/admin_menu.wav , /audio/invalid_input.wav ,1|2|3|5 ) I basically copied from the wiki. Thanks, Keith On Fri, Oct 17, 2008 at 10:08 PM, Brian West [EMAIL PROTECTED] wrote: Can you show me the script? /b On Oct 17, 2008, at 5:01 AM, Keith Wood wrote: Hi Brian, Yes, it is inband. In my lua script, I used 1|2|3|5 instead of (.*). In the log, it is shown that: lua(run_ivr.lua 1 1 3 3000 # /audio/admin_menu.wav /audio/invalid_input.wav 1|2|3|5 admin_selection ) Any suggestion on how to fix this? Also, I am trying to find a way to extract the specific terminator that the user enters. Is it possible to do within the scope of Freeswitch? Thanks, Keith ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Not passing G729 calls
Hi all, I am routing G729 calls from Gateway X to Gateway Z using FREESWITCH Y, I have enabled bypass media for G729 passthrough. my FreeSwitch Y accepts calls from X and when it route to Z it receives 183 Session Progress. from there. Problem comes after that, FS is cancelling the call after getting 183 session progress. Below is the ngrep trace of whole scenario i have mentioned above: # U xx.xx.xxx.xx:63263 - yy.yy.yy.yyy:5060 INVITE sip:[EMAIL PROTECTED] SIP/2.0. Via: SIP/2.0/UDP xx.xx.xxx.xx:63263;branch=z9hG4bK5c350148;rport. From: 1001 sip:[EMAIL PROTECTED];tag=as237d0908. To: sip:[EMAIL PROTECTED]. Contact: sip:[EMAIL PROTECTED]. Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE. User-Agent: Asterisk PBX. Max-Forwards: 70. Date: Sun, 19 Oct 2008 12:46:14 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Supported: replaces. Content-Type: application/sdp. Content-Length: 261. . v=0. o=root 6936 6936 IN IP4 xx.xx.xxx.xx. s=session. c=IN IP4 xx.xx.xxx.xx. t=0 0. m=audio 63264 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. # U yy.yy.yy.yyy:5060 - xx.xx.xxx.xx:63263 SIP/2.0 100 Trying. Via: SIP/2.0/UDP xx.xx.xxx.xx:63263;branch=z9hG4bK5c350148;rport=63263. From: 1001 sip:[EMAIL PROTECTED];tag=as237d0908. To: sip:[EMAIL PROTECTED]. Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.1-exported. Content-Length: 0. . # U yy.yy.yy.yyy:5080 - zz.zz.zz.zzz:5060 INVITE sip:[EMAIL PROTECTED] SIP/2.0. Via: SIP/2.0/UDP yy.yy.yy.yyy:5080;rport;branch=z9hG4bKN0NFB8QrS096F. Max-Forwards: 69. From: 1001 sip:[EMAIL PROTECTED];tag=NUZ4UvUv8tKvK. To: sip:[EMAIL PROTECTED]. Call-ID: eb8148d6-17ab-122c-93b5-003048911494. CSeq: 106046574 INVITE. Contact: sip:[EMAIL PROTECTED]:5080. User-Agent: FreeSWITCH-mod_sofia/1.0.1-exported. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO. Supported: 100rel, timer, precondition, path, replaces. Allow-Events: talk. Min-SE: 120. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 279. Remote-Party-ID: 1001 sip:[EMAIL PROTECTED];screen=yes;privacy=off. . v=0. o=FreeSWITCH 1666183455704603611 2165736585687934666 IN IP4 yy.yy.yy.yyy. s=FreeSWITCH. c=IN IP4 yy.yy.yy.yyy. t=0 0. a=sendrecv. m=audio 20094 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. # U zz.zz.zz.zzz:5060 - yy.yy.yy.yyy:5080 SIP/2.0 100 Trying. Via: SIP/2.0/UDP yy.yy.yy.yyy:5080;rport;branch=z9hG4bKN0NFB8QrS096F. From: 1001 sip:[EMAIL PROTECTED];tag=NUZ4UvUv8tKvK. To: sip:[EMAIL PROTECTED]. Call-ID: eb8148d6-17ab-122c-93b5-003048911494. CSeq: 106046574 INVITE. Content-Length: 0. . # U zz.zz.zz.zzz:5060 - yy.yy.yy.yyy:5080 SIP/2.0 183 Session Progress. Require: 100rel. Via: SIP/2.0/UDP yy.yy.yy.yyy:5080;rport;branch=z9hG4bKN0NFB8QrS096F. RSeq: 1. To: sip:[EMAIL PROTECTED];tag=3433323190-141009. From: 1001 sip:[EMAIL PROTECTED];tag=NUZ4UvUv8tKvK. Call-ID: eb8148d6-17ab-122c-93b5-003048911494. CSeq: 106046574 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: sip:[EMAIL PROTECTED]:5060. Call-Info: sip:zz.zz.zz.zzz;method=NOTIFY;Event=telephone-event;Duration=1000. Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 143. . v=0. o=diamond-msc2 6683 705 IN IP4 zz.zz.zz.zzz. s=sip call. c=IN IP4 213.170.194.56. t=0 0. m=audio 43078 RTP/AVP 101. a=rtpmap:101 /8000. # U yy.yy.yy.yyy:5080 - zz.zz.zz.zzz:5060 CANCEL sip:[EMAIL PROTECTED] SIP/2.0. Via: SIP/2.0/UDP yy.yy.yy.yyy:5080;rport;branch=z9hG4bKN0NFB8QrS096F. Max-Forwards: 69. From: 1001 sip:[EMAIL PROTECTED];tag=NUZ4UvUv8tKvK. To: sip:[EMAIL PROTECTED]. Call-ID: eb8148d6-17ab-122c-93b5-003048911494. CSeq: 106046574 CANCEL. Content-Length: 0. . # U zz.zz.zz.zzz:5060 - yy.yy.yy.yyy:5080 SIP/2.0 200 OK. Via: SIP/2.0/UDP yy.yy.yy.yyy:5080;rport;branch=z9hG4bKN0NFB8QrS096F. To: sip:[EMAIL PROTECTED];tag=3433323190-141009. From: 1001 sip:[EMAIL PROTECTED];tag=NUZ4UvUv8tKvK. Call-ID: eb8148d6-17ab-122c-93b5-003048911494. CSeq: 106046574 CANCEL. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: sip:[EMAIL PROTECTED]:5060. Content-Length: 0. . # U zz.zz.zz.zzz:5060 - yy.yy.yy.yyy:5080 SIP/2.0 487 Request Terminated. Via: SIP/2.0/UDP yy.yy.yy.yyy:5080;rport;branch=z9hG4bKN0NFB8QrS096F. To: sip:[EMAIL PROTECTED];tag=3433323190-141009. From: 1001 sip:[EMAIL PROTECTED];tag=NUZ4UvUv8tKvK. Reason: Q.850;cause=16. Call-ID: eb8148d6-17ab-122c-93b5-003048911494. CSeq: 106046574 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: sip:[EMAIL PROTECTED]:5060. Call-Info:
Re: [Freeswitch-users] Freeswitch and XML-RPC
xml-rpc is mostly for sending FSAPI commands to the system, anything you see when you type show api at the cli which happens to include the show command itself you can also write your own code in C or js or lua or any other lang and access it directly over the web in the same way you would access a cgi where your code can write out the content-type etc. if you want to control calls on a per-call basis you would want to look and mod_event_socket (socket dialplan application) On Sat, Oct 18, 2008 at 1:12 AM, Gayatri Kulkarni [EMAIL PROTECTED]wrote: http://wiki.freeswitch.org/wiki/Mod_xml_rpc Plus, for streaming a file, see to it that you can use port 8080 for it for instance, if you want to read the logfile say - freeswitch.log, configure logfile.conf.xml to to generate the logs in the htdocs directory - root for xml rpc. You can definitely access it from there -- Regards, Gayatri Kulkarni On Wed, Aug 27, 2008 at 8:38 PM, Klaus Teller [EMAIL PROTECTED]wrote: Hi, It seems i can send commands to Freeswitch using XML-RPC. The only document i could find on this is: http://wiki.freeswitch.org/wiki/Freeswitch_XML-RPC Given that i'm using Java, i was wondering what else (beside originating a call) can I do via XML-RPC. How do i send for instance DTMF read requests, how do i send a request to stream a file? Any other document i could check to know what i can do with this? I would think that most of the functions that are available through Javascript are available via XML-RPC. But how do i send these commands? Thanks, Klaus. -- Pt! Schon das coole Video vom GMX MultiMessenger gesehen? Der Eine für Alle: http://www.gmx.net/de/go/messenger03 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[EMAIL PROTECTED] [EMAIL PROTECTED] GTALK/JABBER/PAYPAL:[EMAIL PROTECTED][EMAIL PROTECTED] IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[EMAIL PROTECTED] [EMAIL PROTECTED] iax:[EMAIL PROTECTED]/888 googletalk:[EMAIL PROTECTED][EMAIL PROTECTED] pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Hardware crypto support
Hello everyone, I've been a big fan of hardware crypto acceleration for some time. On x86 I especially like VIA Padlock (available in C3/C7 cpus): http://www.logix.cz/michal/devel/padlock/ I've patched several apps using OpenSSL 0.9.7 to support padlock and the results really are pretty amazing. There are now patches available for OpenSSL 0.9.8 to init the hardware engine for any app compiled against the patched version of OpenSSL. Like the author says, no more patching apps for padlock! However for those of us stuck with OpenSSL 0.9.7 for the time being, where might I begin to look in the sources to patch SSL/TLS support in FreeSWITCH? 1) SIP-TLS 2) SRTP 3) Curl w/ HTTPS 4) What else? The other question (maybe the first question) is - what ciphers are typically negotiated for SRTP (where I expect most of the work to be)? All I've ever seen is AES_CM_128_HMAC_SHA1_32, which *should* do fairly well on cores that have hashing in hardware (Esther/C7). P.S. - I understand that for many configurations I can side step RTP handing all together, or simply pass it through FreeSWITCH. However, in many situations (SIP-TLS SRTP on handset - SIP UDP RTP SIP provider) this isn't possible and FreeSWITCH would need to decrypt the incoming RTP stream/encrypt the outgoing stream (which works perfectly, btw). Thoughts? -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
