June 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

How to Access To Bridged Seesion
by afshin afzali 28 Jun '10

28 Jun '10

Hi FreeSWITCH, What is the preferred method for obtaining bridged session from the current one? I've just found the signal_bond variable. BEST, -- afshin

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Question about Lua in diaplan
by DJB International 28 Jun '10

28 Jun '10

I was trying to make lua to bridge in dialplan, but somehow it did not behave like XML dialplan: If I have this in XML: <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="bridge" data="sofia/gateway/1.2.3.4/$1" /> <action application="bridge" data="sofia/gateway/5.6.7.8/$1" /> then, the second action will not be called if the first one hangs up the channel unless the first action failed. But, if I have this in Lua: session:execute("set","hangup_after_bridge=true") session:execute("set","continue_on_fail=true") for k, v in pairs(split(dst_gw_ip_list, ';')) do session:execute("bridge","sofia/gateway/" .. v .. "/" .. dest_out .. "") end then, the second action is also called. How do I make the lua script to behave like XML? Thank you, Dorn B.

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IP SIP Trunk errors/issues
by Syed Hussain 28 Jun '10

28 Jun '10

Hi , I have subscribed to sonovoip a sip trunk provider who uses IP based authentication for SIP services , I am trying to make outbound international calls which are being rejected by the provider for some reason, I am not able to figure out if there is something wrong with the configuration. I am listing out my configuration. Any help would be greatly appreciated . I have also tried #wiki.freeswitch.org/wiki/Provider_Configuration:_SonoVoIP configuration from freeswitch wiki gives errors . The voip configuration works perfectly with asterisk 1.6 My configuration is follows Under /usr/local/freeswitch/conf/sip_profiles/external cat sonovoip.xml <include> <gateway name="sonovoip"> <param name="username" value="anything"/> <param name="password" value="anything"/> <param name="realm" value="x.x.x.x"/>  <param name="register" value="false"/> <param name="caller-id-in-from" value="true"/> <param name="extension-in-contact" value="true"/> </gateway> </include> My dialplan , under /usr/local/freeswitch/conf/dialplan/public.xml <extension name="sonovoip" continue="true"> <condition field="destination_number" expression="^(00.*)$"> <action application="bridge" data="sofia/gateway/sonovoip/$1"/> </condition> </extension> I did change the vars.xml ports  <X-PRE-PROCESS cmd="set" data="internal_auth_calls=true"/> <X-PRE-PROCESS cmd="set" data="internal_sip_port=5080"/> <X-PRE-PROCESS cmd="set" data="internal_tls_port=5081"/> <X-PRE-PROCESS cmd="set" data="internal_ssl_enable=false"/> <X-PRE-PROCESS cmd="set" data="internal_ssl_dir=$${base_dir}/conf/ssl"/>  <X-PRE-PROCESS cmd="set" data="external_auth_calls=false"/> <X-PRE-PROCESS cmd="set" data="external_sip_port=5060"/> <X-PRE-PROCESS cmd="set" data="external_tls_port=5061"/> <X-PRE-PROCESS cmd="set" data="external_ssl_enable=false"/> <X-PRE-PROCESS cmd="set" data="external_ssl_dir=$${base_dir}/conf/ssl"/> The debug #pastebin.com/4MFhSWm9 Please advise if any other information is required. Thanks S

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Javascript: inconsistent session.ready() result when originating a call
by Marc Kellem 28 Jun '10

28 Jun '10

I'm using Javascript to originate a new outbound call. The Session ready() method returns inconsistent results when the call is answered. When it returns false, the session's causecode value is 0. Am I doing something wrong here, or could it be a bug? A simple test script and log output is below. I'm executing the script from the console using jsrun. I've noticed that switch_core_state_machine.c logs a state change when ready() returns true. I'm not sure if this is significant, but it might indicate a race condition. ---- test script ---- new_session = new Session( "{ignore_early_media=true}sofia/internal/1000%192.168.1.10" ); if ( new_session.ready() ) { console_log( "info", "Call connected.\n" ); // play audio.... } else { // the causecode is 0 when this fails on a successful call attempt console_log("err", "Call not connected. Cause: " + new_session.cause + "[" + new_session.causecode + "]\n" ); } ---- call connected log ---- [NOTICE] sofia.c:4851 Channel [sofia/internal/1000] has been answered [DEBUG] switch_ivr_originate.c:3273 Originate Resulted in Success: [sofia/internal/1000] [DEBUG] mod_spidermonkey.c:2866 (sofia/internal/1000) State Change CS_CONSUME_MEDIA -> CS_SOFT_EXECUTE [DEBUG] switch_core_session.c:1027 Send signal sofia/internal/1000 [BREAK] [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000) Running State Change CS_SOFT_EXECUTE [INFO] originate.js:1 Call connected. [DEBUG] switch_core_state_machine.c:354 (sofia/internal/1000) State SOFT_EXECUTE [DEBUG] mod_sofia.c:544 SOFIA SOFT_EXECUTE [DEBUG] switch_core_state_machine.c:200 sofia/internal/1000 Standard SOFT_EXECUTE [DEBUG] switch_core_state_machine.c:354 (sofia/internal/1000) State SOFT_EXECUTE going to sleep ---- call not connected log ---- [NOTICE] sofia.c:4851 Channel [sofia/internal/1000] has been answered [DEBUG] switch_ivr_originate.c:3273 Originate Resulted in Success: [sofia/internal/1000] [DEBUG] mod_spidermonkey.c:2866 (sofia/internal/1000) State Change CS_CONSUME_MEDIA -> CS_SOFT_EXECUTE [DEBUG] switch_core_session.c:1027 Send signal sofia/internal/1000 [BREAK] [ERR] originate.js:1 Call not connected. Cause: NONE(0) [DEBUG] switch_channel.c:2261 (sofia/internal/1000) Callstate Change ACTIVE -> HANGUP [NOTICE] mod_spidermonkey.c:3066 Hangup sofia/internal/1000 [CS_SOFT_EXECUTE] [NORMAL_CLEARING]

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Use lua to replace a .wav file
by Raymond Chan 28 Jun '10

28 Jun '10

Hi, I am using freeswitch as call announcement purpose. Admin user can call in and record a sound clip and then updated the announcement sound clip. I created IVR menu and use the recording lua to record the sound clip (to record as temp...uuid...wav file). It is ok. However, I do not know how to replace the announcement sound clip after the user press a key in ivr to confirm update. I try to use lua to do it, but it seems that it do not have file replacement function. Am I correct? Thanks Raymond

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Invalid Application hash
by Lars Zeb 28 Jun '10

28 Jun '10

Recently I've been getting an Invalid Application hash error. Any ideas on what might be causing this. I have not changed anything in the configuration for a long while. I am using git-c58efca 2010-06-24 11-52-49 -0500 on Centos 5.3. Pastebin is http://pastebin.freeswitch.org/13255. Thanks Lars

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Beginner Question
by Deya M 28 Jun '10

28 Jun '10

Hi, After doing some changes in my config, I am now unable to make any calls, eg. extension to extension! Looking at the log files, I can see that calling 5000 (Demo) has a context of public instead of default (or internal) I am not sure which xml file should I change, and change it to what ? Appreciate your help, D:-

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SIP/2.0 500 No session set by user
by Erik Dekkers 28 Jun '10

28 Jun '10

Hi guys, I've got a problem with a Aastra SIP Phone. When I make a call to freeswitch from this Phone, I get an return SIP message "SIP/2.0 500 No session set by user". This occurs when I call from the Aastra Phone to a Cisco Skinny Phone (mod_skinny). When I call from Xlite to that same phone, it just Works. Besides that, when I call from the Aastra to Xlite and vice versa no problem occurs (SIP to SIP). I've searched the internet on this and came on the Sofia-Sip website. There I found this message is generated by soa_static.c (line 1131). There it says: if (user == NULL) return soa_set_status(ss, 500, "No session set by user"); But what does it mean by "USER"? Can somebody point me in the right direction so I can find out why it's replying with No session set by User Below i've attached the sip trace Thanks in advance. Regards, Erik (wvds-nl) =========================================================================== sip:111@192.168.10.10:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.204:5060;branch=z9hG4bKaa627edaa1360db77.0caf3c848d78869e6 Max-Forwards: 70 From: "Erik Dekkers" <sip:201@192.168.10.10:5060>;tag=ed6e722b28 To: "111" <sip:111@192.168.10.10:5060> Call-ID: 86be725683a1a511 CSeq: 19626 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "Erik Dekkers" <sip:201@192.168.10.204:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D2433AD>" Supported: gruu, path, timer, 100rel, replaces User-Agent: Aastra 57i/2.6.0.66 Content-Type: application/sdp Content-Length: 236 v=0 o=MxSIP 0 0 IN IP4 192.168.10.204 s=SIP Call c=IN IP4 192.168.10.204 t=0 0 m=audio 3000 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:on - - - - a=fmtp:101 0-15 a=ptime:20 a=sendrecv =========================================================================== SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.204:5060;branch=z9hG4bKaa627edaa1360db77.0caf3c848d78869e6 From: "Erik Dekkers" <sip:201@192.168.10.10:5060>;tag=ed6e722b28 To: "111" <sip:111@192.168.10.10:5060> Call-ID: 86be725683a1a511 CSeq: 19626 INVITE User-Agent: FreeSWITCH Rocks! Content-Length: 0 =========================================================================== SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.10.204:5060;branch=z9hG4bKaa627edaa1360db77.0caf3c848d78869e6 From: "Erik Dekkers" <sip:201@192.168.10.10:5060>;tag=ed6e722b28 To: "111" <sip:111@192.168.10.10:5060>;tag=KgpQKj9UD29mj Call-ID: 86be725683a1a511 CSeq: 19626 INVITE User-Agent: FreeSWITCH Rocks! Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Proxy-Authenticate: Digest realm="192.168.10.10", nonce="852fcd72-b282-df11-9043-00e081630bad", algorithm=MD5, qop="auth" Content-Length: 0 =========================================================================== ACK sip:111@192.168.10.10:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.204:5060;branch=z9hG4bKaa627edaa1360db77.0caf3c848d78869e6 Max-Forwards: 70 From: "Erik Dekkers" <sip:201@192.168.10.10:5060>;tag=ed6e722b28 To: "111" <sip:111@192.168.10.10:5060>;tag=KgpQKj9UD29mj Call-ID: 86be725683a1a511 CSeq: 19626 ACK User-Agent: Aastra 57i/2.6.0.66 Content-Length: 0 =========================================================================== INVITE sip:111@192.168.10.10:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.204:5060;branch=z9hG4bK97ad060aed9a928de.58ac06998b369e607 Proxy-Authorization: Digest username="201",realm="192.168.10.10",nonce="852fcd72-b282-df11-9043-00e081630bad",uri="sip:111@192.168.10.10:5060",response="b5416cff70cb923195c78c9c441d2186",algorithm=MD5,qop=auth,cnonce="deaeb147",nc=00000001 Max-Forwards: 70 From: "Erik Dekkers" <sip:201@192.168.10.10:5060>;tag=ed6e722b28 To: "111" <sip:111@192.168.10.10:5060> Call-ID: 86be725683a1a511 CSeq: 19627 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "Erik Dekkers" <sip:201@192.168.10.204:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D2433AD>" Supported: gruu, path, timer, 100rel, replaces User-Agent: Aastra 57i/2.6.0.66 Content-Type: application/sdp Content-Length: 236 v=0 o=MxSIP 0 0 IN IP4 192.168.10.204 s=SIP Call c=IN IP4 192.168.10.204 t=0 0 m=audio 3000 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:on - - - - a=fmtp:101 0-15 a=ptime:20 a=sendrecv =========================================================================== SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.204:5060;branch=z9hG4bK97ad060aed9a928de.58ac06998b369e607 From: "Erik Dekkers" <sip:201@192.168.10.10:5060>;tag=ed6e722b28 To: "111" <sip:111@192.168.10.10:5060> Call-ID: 86be725683a1a511 CSeq: 19627 INVITE User-Agent: FreeSWITCH Rocks! Content-Length: 0 =========================================================================== UPDATE sip:201@192.168.10.204:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.10;rport;branch=z9hG4bKpSjty6aygUreN Max-Forwards: 70 From: "111" <sip:111@192.168.10.10:5060>;tag=mSFgNDtZaB07D To: "Erik Dekkers" <sip:201@192.168.10.10:5060>;tag=ed6e722b28 Call-ID: 86be725683a1a511 CSeq: 132746103 UPDATE Contact: <sip:192.168.10.10> User-Agent: FreeSWITCH Rocks! Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0 P-Asserted-Identity: "test" <sip:111@192.168.10.10> =========================================================================== SIP/2.0 500 No session set by user Via: SIP/2.0/UDP 192.168.10.204:5060;branch=z9hG4bK97ad060aed9a928de.58ac06998b369e607 From: "Erik Dekkers" <sip:201@192.168.10.10:5060>;tag=ed6e722b28 To: "111" <sip:111@192.168.10.10:5060>;tag=mSFgNDtZaB07D Call-ID: 86be725683a1a511 CSeq: 19627 INVITE Contact: <sip:192.168.10.10> User-Agent: FreeSWITCH Rocks! Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 =========================================================================== SIP/2.0 501 Not Implemented Via: SIP/2.0/UDP 192.168.10.10;rport=5060;branch=z9hG4bKpSjty6aygUreN;received=192.168.10.10 From: "111" <sip:111@192.168.10.10:5060>;tag=mSFgNDtZaB07D To: "Erik Dekkers" <sip:201@192.168.10.10:5060>;tag=ed6e722b28 Call-ID: 86be725683a1a511 CSeq: 132746103 UPDATE Server: Aastra 57i/2.6.0.66 Supported: gruu, path Content-Length: 0 =========================================================================== ACK sip:111@192.168.10.10:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.204:5060;branch=z9hG4bK97ad060aed9a928de.58ac06998b369e607 Max-Forwards: 70 From: "Erik Dekkers" <sip:201@192.168.10.10:5060>;tag=ed6e722b28 To: "111" <sip:111@192.168.10.10:5060>;tag=mSFgNDtZaB07D Call-ID: 86be725683a1a511 CSeq: 19627 ACK User-Agent: Aastra 57i/2.6.0.66 Content-Length: 0

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Question about max-members
by Sergey Scheglov 28 Jun '10

28 Jun '10

Hi All, max-members limits members in conference room or limits members using profile (i.e. default)? Regards Sergey Scheglov

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Build error - on linux, during make, near mod_spandsp
by Fraser Redmond 27 Jun '10

27 Jun '10

Ubuntu Lucid (10.04) via nightly tarball. Configure seems to run fine, but it breaks partway through the make. This is the output right before it exits: Making all in src Creating mod_spandsp.la /usr/bin/ld: cannot find -ljpeg collect2: ld returned 1 exit status gcc -shared .libs/mod_spandsp_la-mod_spandsp.o .libs/mod_spandsp_la-udptl.o .libs/mod_spandsp_la-mod_spandsp_fax.o .libs/mod_spandsp_la-mod_spandsp_dsp.o .libs/mod_spandsp_la-mod_spandsp_codecs.o -Wl,--rpath -Wl,/usr/src/freeswitch-snapshot/.libs -Wl,--rpath -Wl,/usr/local/freeswitch/lib -L/usr/src/freeswitch-snapshot/libs/tiff-3.8.2/libtiff/.libs -lm -lz -ljpeg /usr/src/freeswitch-snapshot/.libs/libfreeswitch.so -L/usr/src/freeswitch-snapshot/libs/apr-util/xml/expat/lib -L/usr/src/freeswitch-snapshot/libs/apr-util/xml/expat/lib/.libs -L/usr/src/freeswitch-snapshot/libs/apr/.libs -L/usr/src/freeswitch-snapshot/libs/srtp /usr/src/freeswitch-snapshot/libs/spandsp/src/.libs/libspandsp.a -L/usr/src/freeswitch-snapshot/libs/tiff-3.8.2/libtiff /usr/src/freeswitch-snapshot/libs/tiff-3.8.2/libtiff/.libs/libtiff.a -lssl -lcrypto -lncurses -Wl,-soname -Wl,mod_spandsp.so -o .libs/mod_spandsp.so make[4]: *** [mod_spandsp.la] Error 1 make[3]: *** [mod_spandsp-all] Error 1 make[2]: *** [all-recursive] Error 1 The output above that (and in the configure) looks ok, unless there's something subtle to look out for. I carried on with the make install the first time, just in case, but the mod's that come after spandsp (alphabetically) weren't created. Any ideas? Cheers, Fraser ps. I have the prerequisites listed on the first line of http://wiki.freeswitch.org/wiki/Installation_Guide#Debian_Linux installed. I've installed on a Jaunty Ubuntu on AWS several times before without problem (using same prerequisites and installation steps.) pps. I know the git version is preferred, but I generally don't need to be on the cutting edge - but prove me wrong in this instance if you want :-)

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