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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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Erlang Examples
by David Swardstrom 27 Jun '10

27 Jun '10

I have been using JavaScript to handle a Conferencing application that started with the conf-ivr.js example program but is significantly more complex. This has been fun even though I had never used JavaScript before this year. However, there are things that seem to not be possible using JavaScript. I need to interact with several web based applications for several reasons and also need to provide some time based interactions with FreeSwitch and/or artifacts (Database entries, files of recorded conferences, etc). We (RemoteLink) have decided that the best solution for this support is to use an Erlang program and mod_erlang_event. So now I need to learn another language. But one thing that I do not find one the FreeSWITCH site is any Erlang examples. Are there some sample programs available such as one that would look for a certain type of event and print it out? I have found some semi-samples in the freeswitch-users archives but am somewhat ambivalent about using any of these without permission. Regards, David Swardstrom (profile)http://wiki.freeswitch.org/wiki/User:Dswardstrom

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ringback answer vs preanswer
by Stephen Cattaneo 26 Jun '10

26 Jun '10

i am trying to understand why when i do session.preanswer ringback works when i bridge the call but when i do session.answer it does not. Thanks, Stephen C -All of my email addresses go to the same place -Save Paper, think before you print

2 1

event-lock in mod_erlang_event
by Seven Du 25 Jun '10

25 Jun '10

Hi, As I can see, You cannot do this in inbound erlang sendmsg(FS, uuid, playback, "1.wav"); sendmsg(FS, uuid, playback, "2.wav"); sendmsg(FS, uuid, transfer, "xxxxx because it's async, and it will play 2.wav immediately. 1) Sure if I know the length of 1.wav I can sendmsg(FS, uuid, playback, 1.wav sleep(3000 sendmsg(FS, uuid, playback, 2.wav 2) Or I could wait the execute_complete event which will be a little complicated According to http://wiki.freeswitch.org/wiki/Event_socket_outbound Is it possible to send a event-lock param to lock the message temporarily? like sendmsg(FS, UUID, App, Args, [{"event-lock", "true"}]). -- Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn

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try to run mod_java but loading issue
by Ravi Kuru 25 Jun '10

25 Jun '10

Hi, I try to run PhoneTest on freeswitch and I followed the instruction but I got this error on freeswitch.log file when i start the freeswitch. 2010-06-24 15:01:47.096906 [ERR] modjava.c:124 Error loading /usr/telcan/jdk1.6.0_10/jre/lib/i386/server/libjvm.so 2010-06-24 15:01:47.096942 [CRIT] switch_loadable_module.c:882 Error Loading module /usr/local/freeswitch/mod/mod_java.so **Module load routine returned an error** what was the issue? Ravi

2 1

Voice mail
by mattias jonsson 25 Jun '10

25 Jun '10

Have freeswitch integrated voicemail?

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Looking for FreeSWITCH user in Öresund-area (Copenhagen/Malmö)
by Harry Vangberg 25 Jun '10

25 Jun '10

Hello I am looking for an experienced FreeSWITCH user/administrator in the greater Öresund area (Copenhagen/Malmö) to join an interesting project. Please let me know if you are interested or know anybody that are. Best regards, Harry Vangberg <harry(a)vangberg.name> http://harry.vangberg.name

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