September 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Recording Only 1 Leg of a Call
by Matthew Fong 08 Sep '09

08 Sep '09

Whats the best way to record only one leg of a call? uuid_record records both channels session_record does the same (but has a stereo option) is there any way to record only an a-leg of the call? Thanks so much. --matt http://www.hellohunter.com hosted dialer & voice broadcasting

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patch for debian init script
by Christian Löschenkohl 08 Sep '09

08 Sep '09

hi just a quick patch for the debian init script debian/freeswitch.init i do use the reload function and the script complains about the -C option it also would be perfect if the reload option is enabled by default (usefull for logrotating) -> combined in second patch br ********************************************************************************************* --- debian/freeswitch.init-old 2009-09-08 16:18:49.000000000 +0200 +++ debian/freeswitch.init 2009-09-08 16:19:13.000000000 +0200 @@ -103,7 +103,7 @@ # restarting (for example, when it is sent a SIGHUP), # then implement that here. # - start-stop-daemon -C $USER --stop --signal 1 --quiet --pidfile $PIDFILE --name $NAME + start-stop-daemon -c $USER --stop --signal 1 --quiet --pidfile $PIDFILE --name $NAME return 0 } ********************************************************************************************* --- debian/freeswitch.init-old 2009-09-08 16:26:01.000000000 +0200 +++ debian/freeswitch.init 2009-09-08 16:19:13.000000000 +0200 @@ -103,7 +103,7 @@ # restarting (for example, when it is sent a SIGHUP), # then implement that here. # - start-stop-daemon -C $USER --stop --signal 1 --quiet --pidfile $PIDFILE --name $NAME + start-stop-daemon -c $USER --stop --signal 1 --quiet --pidfile $PIDFILE --name $NAME return 0 } @@ -124,15 +124,15 @@ 2) [ "$VERBOSE" != no ] && log_end_msg 1 ;; esac ;; - reload|force-reload) + #reload|force-reload) # # If do_reload() is not implemented then leave this commented out # and leave 'force-reload' as an alias for 'restart'. # - log_daemon_msg "Reloading $DESC" "$NAME" - do_reload - log_end_msg $? - ;; + #log_daemon_msg "Reloading $DESC" "$NAME" + #do_reload + #log_end_msg $? + #;; restart|force-reload) # # If the "reload" option is implemented then remove the @@ -156,8 +156,8 @@ esac ;; *) - echo "Usage: $SCRIPTNAME {start|stop|restart|reload|force-reload}" >&2 - #echo "Usage: $SCRIPTNAME {start|stop|restart|force-reload}" >&2 + #echo "Usage: $SCRIPTNAME {start|stop|restart|reload|force-reload}" >&2 + echo "Usage: $SCRIPTNAME {start|stop|restart|force-reload}" >&2 exit 3 ;; esac -- Ing. Christian Löschenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl(a)xpirio.com

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Conference DTMFs heard by participants
by Bradley Brashier 08 Sep '09

08 Sep '09

I have a FreeSWITCH conference with a list of DTMFs, some of which are handled through the event socket (like mute-all), some of which are handled by FreeSWITCH itself (like mute-self). There are a number of commands available and all of them are 2 digits in length. The issue is that when a command is pressed on one phone in the conference, all users hear the tones of the first key pressed. My expectation is that no other users should hear any of the keys, at least not unless they do not correspond to any command. This happens regardless of whether it's a command processed by the event app or if it's processed by FS. I have tried using single-digit commands and the same thing happens -- that single digit is heard by all conference members. Questions: 1) Is this expected behavior (ie. is there some reason you would want this)? 2) Is this something that I can change with some parameter somewhere? BB

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mod_cdr_csv and mysql
by Hristo Benev 08 Sep '09

08 Sep '09

Hello, I saw an sql option in mod_cdr_csv. For my surprise it wrote sql code in Master.csv file instead of recording to mysql database (already setup as ODBC) Is that normal or I'm missing something? I read on the wiki that there is additional perl script to load csv to mysql. Should I do it in that way? Thanks,

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Custom Variables
by Tina Martinez 08 Sep '09

08 Sep '09

Yes, it is a conference event. In looking at the mod_conference wiki, I see verbose-events mentioned, but the example value is ???. What is an appropriate setting for this param? (i.e., is it numeric, boolean, latin?) -T

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Minimum/Recommended Freeswitch System Configuration
by Jerry Richards 08 Sep '09

08 Sep '09

Under the Minimum/Recommended System Requirements, what is meant by "We recommend you plan for 50% duty cycle"? What is this duty cycle? Also, I see that the system requirements indicate Freeswitch recommends 1GB RAM and 50MB disk space. I guess I'm wondering how the number of extensions and external interfaces drive size of RAM and disk space? For example, would these recommendations support 100 extensions and one external interface? 500 extensions and 10 external interfaces? Etc.? Best Regards, Jerry

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Custom Variables
by Tina Martinez 08 Sep '09

08 Sep '09

Hello, I have created custom variables that I pass to my FreeSWITCH dialplan. In my code I am monitoring the events and using the variables (now using ESL, very nice). The custom variables appear in events such as CHANNEL_ORIGINATE, but I do not see the variable in the CUSTOM event. For example, in CHANNEL_ORIGINATE, I see the custom variables as: variable_sip_h_X-custom_variable1 : ABC variable_sip_h_X-custom_variable2 : XYZ Is there a way for me to see the variables in the CUSTOM event as well? In my dialplan, I do have action application="verbose_events" data="true". But it does not seem to help. Any thoughts? - T

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Usefull option for mod_cdr_csv
by Anatoliy Kounitskiy 08 Sep '09

08 Sep '09

There is another useful option for the module Mod_cdr_csv, that is not described in the default configuration file: <param name="master-file-only" value="true"/> If you use the default configuration file and you're using the variable accountcode, probably you've seen for every accountcode you have separate cdr file for it. But if you need only one file with all cdr - use the option above. Possible values are "true" or "false". Cheers, -- Anatoliy Kounitskiy ------------------------- E-mail: anatoliy(a)kounitskiy.com Mobile: +359898913540

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482 Request merged, in serial forking
by Humberto Quintana 08 Sep '09

08 Sep '09

Hi Brian, Yes , the Call-Id is the same for the 2nd and 3rd transaction but the branch parameter in the Via header is different. Please check the capture below. Thanks, Humberto ----------> Route 1 U 2009/09/08 09:17:38.759129 kamailio:5060 -> freeswitch:5060 INVITE sip:5145555555@gw1:5060 SIP/2.0. Record-Route: <sip:kamailio;lr=on;ftag=8c90b1379825fa62;nat=yes;vsf=QllAAwMAAglyAgIIbgoHFhwIGRwDAC4yMTA6NDUwNjA-;did=0b8.111bca94>. Via: SIP/2.0/UDP kamailio;branch=z9hG4bKede5.5c558357.0. Via: SIP/2.0/UDP 192.168.2.13:52060;rport=52060;received=UserIP;branch=z9hG4bK4d46edc4e1623ae5. From: "hq160" <10092020@kamailio>;tag=8c90b1379825fa62. To: <sip:15145555555@kamailio>. Contact: <sip:10092020@UserIP:52060>. Supported: replaces, timer, path. Call-ID: 04d14ab631843dc1(a)192.168.2.13. CSeq: 31348 INVITE. Max-Forwards: 68. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK. Content-Type: application/sdp. Content-Length: 396. . v=0. o=10092020 8000 8001 IN IP4 192.168.2.13. s=SIP Call. c=IN IP4 MediaServer. t=0 0. m=audio 50362 RTP/AVP 0 101. a=sendrecv. a=rtpmap:0 PCMU/8000. a=ptime:20. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11. m=video 50366 RTP/AVP 99. a=sendrecv. a=rtpmap:99 H264/90000. a=fmtp:99 profile-level-id=42801E; packetization-mode=0; sprop-parameter-sets=J0KAFJWgUH5A,KM4CfIC=. a=framerate:15. U 2009/09/08 09:17:38.861646 freeswitch:5060 -> kamailio:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP kamailio;branch=z9hG4bKede5.5c558357.0. Via: SIP/2.0/UDP 192.168.2.13:52060;rport=52060;received=UserIP;branch=z9hG4bK4d46edc4e1623ae5. Record-Route: <sip:kamailio;lr=on;ftag=8c90b1379825fa62;nat=yes;vsf=QllAAwMAAglyAgIIbgoHFhwIGRwDAC4yMTA6NDUwNjA-;did=0b8.111bca94>. From: "hq160" <10092020@kamailio>;tag=8c90b1379825fa62. To: <sip:15145555555@kamailio>. Call-ID: 04d14ab631843dc1(a)192.168.2.13. CSeq: 31348 INVITE. UserIP-Agent: FreeSWITCH-mod_sofia/1.0.4-exported. Content-Length: 0. . U 2009/09/08 09:17:39.000958 freeswitch:5060 -> gw1:5060 INVITE sip:5145555555@gw1:5060 SIP/2.0. Via: SIP/2.0/UDP freeswitch;rport;branch=z9hG4bK151FSXQmjX4KH. Max-Forwards: 67. From: "hq160" <sip:10092020@freeswitch>;tag=3mrtKm2rma0De. To: <sip:5145555555@gw1:5060>. Call-ID: d2bda062-171c-122d-c787-005056aa5fb7. CSeq: 120089593 INVITE. Contact: <sip:mod_sofia@kamailio>. UserIP-Agent: FreeSWITCH-mod_sofia/1.0.4-exported. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO. Supported: timer, precondition, path, replaces. Allow-Events: talk, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 372. X-rsbc: 04d14ab631843dc1(a)192.168.2.13. Remote-Party-ID: "hq160" <sip:10092020@freeswitch>;party=calling;screen=yes;privacy=off. . v=0. o=10092020 8000 8001 IN IP4 192.168.2.13. s=SIP Call. c=IN IP4 MediaServer. t=0 0. m=audio 50362 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11. a=ptime:20. m=video 50366 RTP/AVP 99. a=rtpmap:99 H264/90000. a=fmtp:99 profile-level-id=42801E; packetization-mode=0; sprop-parameter-sets=J0KAFJWgUH5A,KM4CfIC=. a=framerate:15. U 2009/09/08 09:17:39.056082 gw1:5060 -> freeswitch:5060 SIP/2.0 100 Trying. Call-ID: d2bda062-171c-122d-c787-005056aa5fb7. Content-Length: 0. CSeq: 120089593 INVITE. From: "hq160"<sip:10092020@freeswitch>;tag=3mrtKm2rma0De. To: <sip:5145555555@gw1:5060>;tag=d05714dc-26334. UserIP-Agent: Quintum/1.0.0 SN/0030E100A224 SW/P106-12-00. Via: SIP/2.0/UDP freeswitch;rport;branch=z9hG4bK151FSXQmjX4KH. Quintum: 0b06343032343333. . U 2009/09/08 09:17:39.058998 gw1:5060 -> freeswitch:5060 SIP/2.0 503 Service Unavailable. Call-ID: d2bda062-171c-122d-c787-005056aa5fb7. Content-Length: 0. CSeq: 120089593 INVITE. From: "hq160"<sip:10092020@freeswitch>;tag=3mrtKm2rma0De. To: <sip:5145555555@gw1:5060>;tag=d05714dc-26334. UserIP-Agent: Quintum/1.0.0 SN/0030E100A224 SW/P106-12-00. Via: SIP/2.0/UDP freeswitch;rport;branch=z9hG4bK151FSXQmjX4KH. . U 2009/09/08 09:17:39.059341 freeswitch:5060 -> gw1:5060 ACK sip:5145555555@gw1:5060 SIP/2.0. Via: SIP/2.0/UDP freeswitch;rport;branch=z9hG4bK151FSXQmjX4KH. Max-Forwards: 67. From: "hq160" <sip:10092020@freeswitch>;tag=3mrtKm2rma0De. To: <sip:5145555555@gw1:5060>;tag=d05714dc-26334. Call-ID: d2bda062-171c-122d-c787-005056aa5fb7. CSeq: 120089593 ACK. Content-Length: 0. . U 2009/09/08 09:17:39.061148 freeswitch:5060 -> kamailio:5060 SIP/2.0 503 Service Unavailable. Via: SIP/2.0/UDP kamailio;branch=z9hG4bKede5.5c558357.0. Via: SIP/2.0/UDP 192.168.2.13:52060;rport=52060;received=UserIP;branch=z9hG4bK4d46edc4e1623ae5. From: "hq160" <10092020@kamailio>;tag=8c90b1379825fa62. To: <sip:15145555555@kamailio>;tag=2BZ1HSHNQ19tj. Call-ID: 04d14ab631843dc1(a)192.168.2.13. CSeq: 31348 INVITE. UserIP-Agent: FreeSWITCH-mod_sofia/1.0.4-exported. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO. Supported: timer, precondition, path, replaces. Allow-Events: talk, refer. Reason: Q.850;cause=41;text="NORMAL_TEMPORARY_FAILURE". Content-Length: 0. . U 2009/09/08 09:17:39.062085 kamailio:5060 -> freeswitch:5060 ACK sip:5145555555@gw1:5060 SIP/2.0. Via: SIP/2.0/UDP kamailio;branch=z9hG4bKede5.5c558357.0. From: "hq160" <10092020@kamailio>;tag=8c90b1379825fa62. Call-ID: 04d14ab631843dc1(a)192.168.2.13. To: <sip:15145555555@kamailio>;tag=2BZ1HSHNQ19tj. CSeq: 31348 ACK. Max-Forwards: 70. UserIP-Agent: Kamailio (1.4.4-tls (i386/linux)). Content-Length: 0. . -------> Route 2 U 2009/09/08 09:17:41.426541 kamailio:5060 -> freeswitch:5060 INVITE sip:15145555555@gw2:5061 SIP/2.0. Record-Route: <sip:kamailio;lr=on;ftag=8c90b1379825fa62;nat=yes;vsf=QllAAwMAAglyAgIIbgoHFhwIGRwDAC4yMTA6NDUwNjA-;did=0b8.111bca94>. Via: SIP/2.0/UDP kamailio;branch=z9hG4bKede5.5c558357.1. Via: SIP/2.0/UDP 192.168.2.13:52060;rport=52060;received=UserIP;branch=z9hG4bK4d46edc4e1623ae5. From: "hq160" <10092020@kamailio>;tag=8c90b1379825fa62. To: <sip:15145555555@kamailio>. Contact: <sip:10092020@UserIP:52060>. Supported: replaces, timer, path. Call-ID: 04d14ab631843dc1(a)192.168.2.13. CSeq: 31348 INVITE. Max-Forwards: 68. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK. Content-Type: application/sdp. Content-Length: 396. . v=0. o=10092020 8000 8001 IN IP4 192.168.2.13. s=SIP Call. c=IN IP4 MediaServer. t=0 0. m=audio 50362 RTP/AVP 0 101. a=sendrecv. a=rtpmap:0 PCMU/8000. a=ptime:20. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11. m=video 50366 RTP/AVP 99. a=sendrecv. a=rtpmap:99 H264/90000. a=fmtp:99 profile-level-id=42801E; packetization-mode=0; sprop-parameter-sets=J0KAFJWgUH5A,KM4CfIC=. a=framerate:15. U 2009/09/08 09:17:41.427280 freeswitch:5060 -> kamailio:5060 SIP/2.0 482 Request merged. Via: SIP/2.0/UDP kamailio;branch=z9hG4bKede5.5c558357.1. Via: SIP/2.0/UDP 192.168.2.13:52060;rport=52060;received=UserIP;branch=z9hG4bK4d46edc4e1623ae5. From: "hq160" <10092020@kamailio>;tag=8c90b1379825fa62. To: <sip:15145555555@kamailio>;tag=2BZ1HSHNQ19tj. Call-ID: 04d14ab631843dc1(a)192.168.2.13. CSeq: 31348 INVITE. Content-Length: 0. . U 2009/09/08 09:17:41.427901 kamailio:5060 -> freeswitch:5060 ACK sip:15145555555@gw2:5061 SIP/2.0. Via: SIP/2.0/UDP kamailio;branch=z9hG4bKede5.5c558357.1. From: "hq160" <10092020@kamailio>;tag=8c90b1379825fa62. Call-ID: 04d14ab631843dc1(a)192.168.2.13. To: <sip:15145555555@kamailio>;tag=2BZ1HSHNQ19tj. CSeq: 31348 ACK. Max-Forwards: 70. UserIP-Agent: Kamailio (1.4.4-tls (i386/linux)). Content-Length: 0. . --------> Route 3 U 2009/09/08 09:17:44.206445 kamailio:5060 -> freeswitch:5060 INVITE sip:5145555555@gw3:5060 SIP/2.0. Record-Route: <sip:kamailio;lr=on;ftag=8c90b1379825fa62;nat=yes;vsf=QllAAwMAAglyAgIIbgoHFhwIGRwDAC4yMTA6NDUwNjA-;did=0b8.111bca94>. Via: SIP/2.0/UDP kamailio;branch=z9hG4bKede5.5c558357.2. Via: SIP/2.0/UDP 192.168.2.13:52060;rport=52060;received=UserIP;branch=z9hG4bK4d46edc4e1623ae5. From: "hq160" <10092020@kamailio>;tag=8c90b1379825fa62. To: <sip:15145555555@kamailio>. Contact: <sip:10092020@UserIP:52060>. Supported: replaces, timer, path. Call-ID: 04d14ab631843dc1(a)192.168.2.13. CSeq: 31348 INVITE. Max-Forwards: 68. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK. Content-Type: application/sdp. Content-Length: 396. . v=0. o=10092020 8000 8001 IN IP4 192.168.2.13. s=SIP Call. c=IN IP4 MediaServer. t=0 0. m=audio 50362 RTP/AVP 0 101. a=sendrecv. a=rtpmap:0 PCMU/8000. a=ptime:20. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11. m=video 50366 RTP/AVP 99. a=sendrecv. a=rtpmap:99 H264/90000. a=fmtp:99 profile-level-id=42801E; packetization-mode=0; sprop-parameter-sets=J0KAFJWgUH5A,KM4CfIC=. a=framerate:15. U 2009/09/08 09:17:44.207261 freeswitch:5060 -> kamailio:5060 SIP/2.0 482 Request merged. Via: SIP/2.0/UDP kamailio;branch=z9hG4bKede5.5c558357.2. Via: SIP/2.0/UDP 192.168.2.13:52060;rport=52060;received=UserIP;branch=z9hG4bK4d46edc4e1623ae5. From: "hq160" <10092020@kamailio>;tag=8c90b1379825fa62. To: <sip:15145555555@kamailio>;tag=2BZ1HSHNQ19tj. Call-ID: 04d14ab631843dc1(a)192.168.2.13. CSeq: 31348 INVITE. Content-Length: 0. . U 2009/09/08 09:17:44.207762 kamailio:5060 -> freeswitch:5060 ACK sip:5145555555@gw3:5060 SIP/2.0. Via: SIP/2.0/UDP kamailio;branch=z9hG4bKede5.5c558357.2. From: "hq160" <10092020@kamailio>;tag=8c90b1379825fa62. Call-ID: 04d14ab631843dc1(a)192.168.2.13. To: <sip:15145555555@kamailio>;tag=2BZ1HSHNQ19tj. CSeq: 31348 ACK. Max-Forwards: 70. UserIP-Agent: Kamailio (1.4.4-tls (i386/linux)). Content-Length: 0. . ============================================ I'm going to gess the call-id is the same for the second transaction... can you provide a more detailed trace? /b On Sep 4, 2009, at 11:06 AM, Humberto Quintana wrote: > Hello, > > I'm a new Freeswitch user. After some reading I put Freeswitch > (Version 1.0.4) to work as Session Border Controller. I have only > one problem that I dont know how to solve it ( or which parameter to > set) and I'd appreciate if someone could give me a clue about this. > > Kamailio is sitting behind FS and it selects the route or routes in > case of failure (serial forking) . Freeswitch is configured to use > directly the Request-URI sent by Kamailio. > > So, when the 1st route fails, Kamailio receives the Reply from FS > and sends back the ACK to end the transaction. More than 1 second > later, a new INVITE from Kamailio with the next route is tried (With > the To-header's tag is empty. Same Callid, From and Cseq header but > different VIA-header's branch parameter) and FS is answering back > 482 Merged Request. It happens the same for the 3rd route. > > It seems that the transaction is still 'alive' in FS even if the ACK > was received ? > > > Thanks, > > Humberto _________________________________________________________________ Click less, chat more: Messenger on MSN.ca http://go.microsoft.com/?linkid=9677404

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Mod xml cdr
by tom 08 Sep '09

08 Sep '09

hi, reading the wiki tells me i need that module to make cdr happen. please correct me if i outline wrong steps here: - enable module - make install - create file called xml_cdr.conf.xml - based on the parameters make a http post does anyone have a xml file + a php file to make the appropriate mysql-isnerts? thx

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