September 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

cepstral problem
by stephen at stephenjc 13 Dec '10

13 Dec '10

I'm working with cepstral support and they seem to have no idea. Does any one know why cepstral might produce no audio on a cent os 5.5 machine. Its works fine on another box of mine. Thanks, Stephen C -All of my email addresses go to the same place -Save Paper, think before you print

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FreeSWITCH 64bit compilation error (Solaris 10)
by email lists 08 Dec '10

08 Dec '10

Forwarding the issue below to see if anyone is familiar with this issue, and/or what our next steps should be. Thanks, Vladimir Looks like a problem with a Makefile not honoring CFLAGS,etc. Perhaps you can report this to the dev team. Other components built fine but this damn spidermonkey is buggering. # file /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch*.o | head /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-g711.o : ELF 64-bit LSB relocatable AMD64 Version 1 /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-getgat eway.o: ELF 64-bit LSB relocatable AMD64 Version 1 [CMOV] /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-igd_de sc_parse.o: ELF 64-bit LSB relocatable AMD64 Version 1 [CMOV] /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-libtel etone_detect.o: ELF 64-bit LSB relocatable AMD64 Version 1 [SSE2 SSE AMD_3DNow CMOV FPU] /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-libtel etone_generate.o: ELF 64-bit LSB relocatable AMD64 Version 1 [SSE2 SSE AMD_3DNow CMOV] /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-miniso ap.o: ELF 64-bit LSB relocatable AMD64 Version 1 /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-miniss dpc.o: ELF 64-bit LSB relocatable AMD64 Version 1 [SSE] /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-miniup npc.o: ELF 64-bit LSB relocatable AMD64 Version 1 [SSE AMD_3DNow] /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-miniwg et.o: ELF 64-bit LSB relocatable AMD64 Version 1 [SSE] /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-minixm l.o: ELF 64-bit LSB relocatable AMD64 Version 1 [SSE] # file /opt/freeradius-client/sbin/radacct /opt/freeradius-client/sbin/radacct: ELF 64-bit LSB executable AMD64 Version 1 [SSE2 SSE FXSR AMD_3DNow CMOV FPU], dynamically linked, not stripped Build output: Making all in nua LTCOMPILE nua.lo LTCOMPILE nua_common.lo LTCOMPILE nua_stack.lo LTCOMPILE nua_server.lo LTCOMPILE nua_client.lo LTCOMPILE nua_extension.lo LTCOMPILE nua_dialog.lo LTCOMPILE outbound.lo LTCOMPILE nua_params.lo LTCOMPILE nua_register.lo LTCOMPILE nua_registrar.lo LTCOMPILE nua_session.lo LTCOMPILE nua_options.lo LTCOMPILE nua_message.lo LTCOMPILE nua_publish.lo LTCOMPILE nua_subnotref.lo LTCOMPILE nua_notifier.lo LTCOMPILE nua_event_server.lo LTCOMPILE nua_tag.lo LTCOMPILE nua_tag_ref.lo LINK libnua.la LINK libsofia-sip-ua.la libtool: link: warning: `-version-info/-version-number' is ignored for convenience libraries Making all in packages Creating mod_sofia_la-mod_sofia.lo mkdir .libs Compiling mod_sofia.c ... "/builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libs/sofia-sip/libsofi a-sip-ua/nua/nua_params.h", line 221: warning: nonportable bit-field type (E_NONPORTABLE_BIT_FIELD_TYPE) Creating mod_sofia_la-sofia.lo Compiling sofia.c ... "/builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libs/sofia-sip/libsofi a-sip-ua/nua/nua_params.h", line 221: warning: nonportable bit-field type (E_NONPORTABLE_BIT_FIELD_TYPE) "sofia.c", line 3522: warning: enum type mismatch: arg #2 (E_ENUM_TYPE_MISMATCH_ARG) Creating mod_sofia_la-sofia_glue.lo Compiling sofia_glue.c ... "/builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libs/sofia-sip/libsofi a-sip-ua/nua/nua_params.h", line 221: warning: nonportable bit-field type (E_NONPORTABLE_BIT_FIELD_TYPE) Creating mod_sofia_la-sofia_presence.lo Compiling sofia_presence.c ... "/builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libs/sofia-sip/libsofi a-sip-ua/nua/nua_params.h", line 221: warning: nonportable bit-field type (E_NONPORTABLE_BIT_FIELD_TYPE) Creating mod_sofia_la-sofia_reg.lo Compiling sofia_reg.c ... "/builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libs/sofia-sip/libsofi a-sip-ua/nua/nua_params.h", line 221: warning: nonportable bit-field type (E_NONPORTABLE_BIT_FIELD_TYPE) Creating mod_sofia_la-sofia_sla.lo Compiling sofia_sla.c ... "/builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libs/sofia-sip/libsofi a-sip-ua/nua/nua_params.h", line 221: warning: nonportable bit-field type (E_NONPORTABLE_BIT_FIELD_TYPE) Creating mod_sofia.la making all mod_speex Compiling mod_speex.c... mkdir .libs Compiling mod_speex.c ... Creating mod_speex.so... making all mod_spidermonkey cd config; /usr/sfw/bin/gmake -j1 export ld: fatal: file now.o: wrong ELF class: ELFCLASS64 ld: fatal: File processing errors. No output written to now gmake[7]: *** [now] Error 1 gmake[6]: *** [export] Error 2 gmake[5]: *** [/builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libs/js/libjs.la] Error 2 gmake[4]: *** [all] Error 1 gmake[3]: *** [mod_spidermonkey-all] Error 1 gmake[2]: *** [all-recursive] Error 1 Making all in build +-------- FreeSWITCH Build Complete -----------+ + FreeSWITCH has been successfully built. + + Install by running: + + + + /usr/sfw/bin/gmake install + +----------------------------------------------+ gmake[1]: *** [all-recursive] Error 1 gmake: *** [all] Error 2

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Looking for more info about mod_rtmp
by Jarle Aase 25 Nov '10

25 Nov '10

I'm looking for a way to bridge rtmp to sip / audio-conferences, and came across a reference to "mod_rtmp". Does anyone have any information about that project? Jarle

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IVR menu failure/timeout action
by Raymond Chan 19 Nov '10

19 Nov '10

Hi all, I am experiencing an IVR problem. Except the top menu, all other sub menu will not hang up after reached max invalid input or timeout limit. It will go back up upper menu. I want it to load the exit-sound and then hang the call when it reached max invalid input or timeout limit. Do you tell how to configure? <menu name="m800_ivr_submenu-m_ch" greet-long="C:/FreeSWITCH/recordings/800-ivr-2nd-m.wav" greet-short="C:/FreeSWITCH/recordings/greet-short-m.wav" invalid-sound="ivr/ivr-that_was_an_invalid_entry-m.wav" exit-sound="voicemail/vm-goodbye-m.wav" timeout="5000" max-failures="2"> <entry action="menu-exec-app" digits="1" param="playback C:/FreeSWITCH/recordings/800-ivr-announcement-m_ch.wav"/> <entry action="menu-exec-app" digits="2" param="lua endcall-m.lua"/> </menu> Thanks Raymond

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Video Support
by Samuel Macedo 15 Nov '10

15 Nov '10

Hi, I've already read that FS supports video, but I've never seen anyone using or talking about how to use it. Does FS support video conferences? Or only video calls? Is there any documentation about this subject? Thanks, -- Samuel Macedo

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FreeSWITCH Nagios Plugin
by Kristian Kielhofner 28 Oct '10

28 Oct '10

Hello everyone, I've been looking for a FreeSWITCH Nagios plugin. Ideally I'd like to connect to the event socket and run some api commands and return them (as opposed to checking SIP, for example). I haven't found anything and I've started to write one in perl using ESL. I'm sure whatever I come up with is going to be pretty ugly and I'd much rather use something else. Has this been done? Thanks! -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com

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Freeswitch Video Capture and Playback
by Esben Stien 18 Oct '10

18 Oct '10

I'm using ekiga with mod_fsv, trying to record and play back video. When I dial the record extension, it seems to record something, as the video file gets bigger. Trying then to dial the extension for play back, just hangs up, with freeswitch saying: od_fsv.c:247 File version does not match! There seems to be no information on the FSV format or the mod_fsv module on the wiki. Is this at all supposed to work?. What clients and codecs were successful?. Any pointers as to what I can try?. -- Esben Stien is b0ef@e s a http://www. s t n m irc://irc. b - i . e/%23contact sip:b0ef@ e e jid:b0ef@ n n

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mod_limit does not reset count when call gets to voicemail?
by Yehavi Bourvine 15 Oct '10

15 Oct '10

Hello, We are using mod_limit in order to track user's state and know when to play busy/call-wait tone. We encountered two problems (which are probably related): - When the call gets to the voicemail the extension is still though as busy until the caller hangs. - If a call is picked-up using the intercept application the original destination is still thought as busy. When looking in the limit_data table the record of the call is indeed there until a final call hangup. According to the wiki we tried setting the variable limit_ignore_transfer to false both with set command before calling the limit application and also in limit.conf.xml. However, this does not change this behaviour. Is this the intended behaviour? Can it work as I want? (i.e. once the call arrived to voicemail or picked-up the original destination is marked as free). We are using 1.0.6; since this is a production system it is quite hard for me to test the latest git. Thanks! __Yehavi:

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icall's carrier service returns 480 Temporarily Unavailable for outbound INVITE
by mdale111＠verizon.net 14 Oct '10

14 Oct '10

Hello all, I'm going on the fourth day with no response from icall's support on this issue, so I would greatly appreciate any guidance. Freeswitch successfully registers with icalls outbound server, sbc01-car.dal.us.icall.net, and incoming calls are handled properly. But when I try to call a land-line with pa call 1813968XXXX, I get a response of 480 Temporarily Unavailable. I've also registered my external ip address with icalls servers. Also, this works "originate sofia/internal/9996(a)conference.freeswitch.org 1000", but this doesn't "originate sofia/internal/1813968XXXX@sbc01-car.dal.us.icall.net:5060 1000." Windows XP SP3, 32-bit, 1.5gb FreeSWITCH Version 1.0.4 (14460) Here's the pa call sip trace from wireshark: INVITE sip:1813968XXXX@sbc01-car.dal.us.icall.net:5060 SIP/2.0 Via: SIP/2.0/UDP 173.65.36.19;rport;branch=z9hG4bKg0DHp06gr9Utj Max-Forwards: 70 From: "username" <sip:cust_username@sbc01-car.dal.us.icall.net:5060;transport=udp>;tag=43Sy3cymXr8vH To: <sip:1813968XXXX@sbc01-car.dal.us.icall.net:5060> Call-ID: 2896ecba-2e54-122e-88b1-073747d47dd9 CSeq: 1221935 INVITE Contact: <sip:gw+icall@173.65.36.19:5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.4-14460 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: 316 Remote-Party-ID: "username" <sip:813995XXXX@sbc01-car.dal.us.icall.net:5060>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1283092013 1283092014 IN IP4 173.65.36.19 s=FreeSWITCH c=IN IP4 173.65.36.19 t=0 0 m=audio 23986 RTP/AVP 10 0 8 3 101 13 a=rtpmap:10 L16/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 SIP/2.0 100 Trying Via: SIP/2.0/UDP 173.65.36.19;rport=5060;branch=z9hG4bKg0DHp06gr9Utj From: "username" <sip:cust_username@sbc01-car.dal.us.icall.net:5060;transport=udp>;tag=43Sy3cymXr8vH To: <sip:1813968XXXX@sbc01-car.dal.us.icall.net:5060> Call-ID: 2896ecba-2e54-122e-88b1-073747d47dd9 CSeq: 1221935 INVITE User-Agent: CoreSSP SBC Content-Length: 0 SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP 173.65.36.19;rport=5060;branch=z9hG4bKg0DHp06gr9Utj From: "username" <sip:cust_username@sbc01-car.dal.us.icall.net:5060;transport=udp>;tag=43Sy3cymXr8vH To: <sip:1813968XXXX@sbc01-car.dal.us.icall.net:5060>;tag=FQK8XBFmHggUg Call-ID: 2896ecba-2e54-122e-88b1-073747d47dd9 CSeq: 1221935 INVITE User-Agent: CoreSSP SBC Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 Remote-Party-ID: "1813968XXXX" <sip:1813968XXXX@8.19.97.6> ACK sip:1813968XXXX@sbc01-car.dal.us.icall.net:5060 SIP/2.0 Via: SIP/2.0/UDP 173.65.36.19;rport;branch=z9hG4bKg0DHp06gr9Utj Max-Forwards: 70 From: "username" <sip:cust_username@sbc01-car.dal.us.icall.net:5060;transport=udp>;tag=43Sy3cymXr8vH To: <sip:1813968XXXX@sbc01-car.dal.us.icall.net:5060>;tag=FQK8XBFmHggUg Call-ID: 2896ecba-2e54-122e-88b1-073747d47dd9 CSeq: 1221935 ACK Content-Length: 0 Thanks in advance.

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no bridge_uuid variable in xml cdr
by babak yakhchali 13 Oct '10

13 Oct '10

Hi I'm using the following simple extension: <extension name="babak-test"> <condition field="destination_number" expression="^(223|224)$"> <action application="set" data="call_timeout=8"/--> <action application="bridge" data="sofia/internal/$1%${domain}"/> </condition> to test my aspx handler for mod_xml_cdr. it seems no bridge_uuid and bridge_hangup_cause exist on the xml_cdr. should I add them manually or set something to add them to channel variables? thanx

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