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Call For Participants: Lightning Talks at ClueCon 2009
by Michael Collins 14 May '09

14 May '09

*ClueCon 2009 is coming soon!* We are interested in your thoughts on subjects for lighting talks. We would love to have a number of 5-10 minute presentations by members of the community. If you would like to give a talk, or just have an idea for a talk, please let us know. How do lightning talks work? Quite simply, the presenter has just a few minutes to speak on a particular subject, usually no more than 10 minutes. He or she will deliver the information rapidly, which means keeping the presentation focused tightly on the subject being discussed. Lightning talks usually do not have enough time for audience Q&A. However, ClueCon has a long lunch period that is designed to allow attendees plenty of time to interact. Those are perfect times to discuss lightning talks or any other presentations. Those who give presentations enjoy interacting with other attendees in a relaxed atmosphere during lunch or in the evening at dinner. If you haven't already registered for ClueCon 2009 then please call us at 877.742.CLUE right away and we will complete your registration. Also, don't forget that expedia.com has some nice hotel deals for the Wyndham Chicago. Book your room today! We look forward to hearing from you and seeing you all at ClueCon in Chicago. -Michael http://www.cluecon.com 877.742.CLUE

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Cluecon 2009 - Hotel Special!
by Michael Collins 12 May '09

12 May '09

Good news! For those of you who like to get a good deal on a hotel room we've got the inside scoop on a special for the Wyndham Chicago. I just did a search on expedia.com for a hotel with these specs: Hotel only Check-in: 8/3/2009, Check-Out: 8/6/2009 1 room, 2 adults City: Chicago Hotel Name: "Wyndham Chicago" The results were very nice: the non-refundable rate per night is only $140!!! Get to expedia.com NOW and get signed up! We look forward to seeing you in Chicago this August. -Michael http://www.cluecon.com

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ANNOUNCEMENT: www.freeswitch.org site updated
by Michael Collins 12 May '09

12 May '09

FYI, We'd like to let the community know that we've moved the main site, www.freeswitch.org, to a new server and have updated it to the latest Drupal version. We are still calibrating the server so if the site seems slower than usual please let me know off list and we'll take a look. Also, because this upgrade was so major we were forced to wipe out all the users from the previous Drupal install. Please feel free to sign up again if you so desire. You only need to sign up if you wish to post comments. One other new feature is the "Share This" button. (You need JavaScript enabled to see this.) It makes it as easy as possible to share stories with Digg, SU, Deli.ico.us, etc. If there are any stories you've not already dugg then please try out the Share This button to see how easy it is to digg stuff now. As always, if you have any questions or comments then please email me off list and I'll address them quickly. -Michael http://www.cluecon.com

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Cluecon 2009 News
by Michael Collins 11 May '09

11 May '09

FYI, for those of you keeping up on ClueCon 2009 please visit the latest blog entry: http://cluecon.com/node/29 Thanks, Michael

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Seeking for a developer for a small application on top of LibSangoma
by Tzury Bar Yochay 11 May '09

11 May '09

Hi, I would like to build a server which receives modem calls (V.32/V.110) on one end, and passes the data received to an IP server on another end. It then, waits for a reply from the server and pass the reply back to the endpoint device over the modem session.I put a description of the project at: http://www.reguluslabs.com/pub/sangoma-e1-ip-bridge. The project itself consists of a full blown socket application which we intend to write by ourselves. The help we need is to design and write a wrapper around LibSangoma in order to ease our development. A draft of the wrapper is located at : http://www.reguluslabs.com/pub/sangoma-e1-ip-bridge-interface. If you have an experience with the Sangoma API and think you can join forces with us developing this during the coming weeks please reply to me. Looking forward hearing from you, Tzury Bar Yochay Regulus Labs ltd. tzury.by(a)reguluslabs.com

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FreeSWITCH Nortel Gateway Interop
by Benjamin Willis 09 May '09

09 May '09

Greetings, We are performing interop testing to have FreeSWITCH as a gateway with a Nortel CS1000 box and would appreciate any help resolving our issues. There is a box that is a registrar and a redirect server. We can successfully register against this box. The box is picky about the invites we send; it expects something like the following : INVITE sip:2270;phone-context=cdp.udp@newdomain.com;user=phone SIP/2.0 After great help from the freeSWITCH IRC we came up with the following dialplan entry to achieve this style header : <action application="bridge" data="sofia/internal/sip:$1;phone- context=cdp.udp(a)newdomain.com;user=phone;fs_path=sip: 192.168.15.8:5060;"/> At this point we are sending the formatted INVITE message as the redirect server expects. Our next issue where we would appreciate help is after the INVITE is sent, the redirect server sends back a 302 Moved Temporarily as expected. SIP/2.0 302 Moved Temporarily Via: SIP/2.0/TCP 192.168.15.11;branch=z9hG4bKy0NcZvFcNr1ZB;received=192.168.15.11 From: "User 210" <sip:2300@192.168.15.11>;tag=jjmgrmHN6Qjaa To: <sip:2270;phone- context=cdp.udp(a)newdomain.com;user=phone>;tag=49582 Call-ID: 6998551e-b04c-122c-2d80-39a48cb53b8d CSeq: 114437328 INVITE Contact: <sip:2270;phone-context=cdp.udp@newdomain.com: 5060;maddr=192.168.15.10;transport=tcp;user=phone;x-nt- redirect=redirect-server> Content-Length: 0 The subsequent INVITE message is, again, sent back to the redirect server, not to the machine indicated by the maddr entry in the Contact field. The new INVITE looks like this : INVITE sip:2270;phone-context=cdp.udp@newdomain.com: 5060;maddr=192.168.15.10;transport=tcp;user=phone;x-nt- redirect=redirect-server SIP/2.0 Via: SIP/2.0/UDP 192.168.15.11;rport;branch=z9hG4bK93BXjtmKD9pcN Route: <sip:192.168.15.8:5060> Max-Forwards: 69 From: "User 210" <sip:2300@192.168.15.11>;tag=ppSKZmm0rpS3e To: <sip:2270;phone-context=cdp.udp@newdomain.com;user=phone> Call-ID: ce59ac9e-b058-122c-2780-39a48cb53b8d CSeq: 114439989 INVITE Contact: <sip:mod_sofia@192.168.15.11:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.3-12163 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session- description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 360 Remote-Party-ID: "User 210" <sip: 2300(a)192.168.15.11>;screen=yes;privacy=off v=0 o=FreeSWITCH 6002387420083056305 1437766807142321498 IN IP4 192.168.15.11 s=FreeSWITCH c=IN IP4 192.168.15.11 t=0 0 m=audio 28220 RTP/AVP 10 0 8 9 3 101 13 a=rtpmap:10 L16/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 The following is a wireshark capture of the repeated INVITE messages back to the same machine, it repeats 3 times after the initial invite : Conv.| Time | 192.168.15.11 | 192.168.15.8 | 0 |17.284 | INVITE SDP ( 16-bit audio, stereo g711U g711A ...2 GSM teleph) |SIP From: sip:2300@192.168.15.11 To:sip:2270 | |(5060) ------------------> (5060) | 0 |17.291 | 302 Moved Temporarily |SIP Status | |(5060) <------------------ (5060) | 0 |17.291 | ACK | |SIP Request | |(5060) ------------------> (5060) | --------------------------------------------------------- 1 |17.291 | INVITE SDP ( 16-bit audio, stereo g711U g711A ...2 GSM teleph) |SIP From: sip:2300@192.168.15.11 To:sip:2270 | |(5060) ------------------> (5060) | 1 |17.298 | 302 Moved Temporarily |SIP Status | |(5060) <------------------ (5060) | 1 |17.298 | ACK | |SIP Request | |(5060) ------------------> (5060) | --------------------------------------------------------- 2 |17.298 | INVITE SDP ( 16-bit audio, stereo g711U g711A ...2 GSM teleph) |SIP From: sip:2300@192.168.15.11 To:sip:2270 | |(5060) ------------------> (5060) | 2 |17.304 | 302 Moved Temporarily |SIP Status | |(5060) <------------------ (5060) | 2 |17.305 | ACK | |SIP Request | |(5060) ------------------> (5060) | --------------------------------------------------------- 3 |17.305 | INVITE SDP ( 16-bit audio, stereo g711U g711A ...2 GSM teleph) |SIP From: sip:2300@192.168.15.11 To:sip:2270 | |(5060) ------------------> (5060) | 3 |17.311 | 302 Moved Temporarily |SIP Status | |(5060) <------------------ (5060) | 3 |17.311 | ACK | |SIP Request | |(5060) ------------------> (5060) | At this point we are a little stuck. We feel that the use of fs_path may be having unwanted effects on our redirects after looking at sofia_glue.c:sofia_glue_do_invite() and the parsing of the fs_path variable. We haven't filed this as a bug as it may be intended functionality and would like to get some insight from the more experience members before going down such a path. Thanks in advance.

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Adding a new embedded language
by Kevin Raison 08 May '09

08 May '09

I am thinking about writing a new embedded language module for Freeswitch using Common Lisp, specifically ECL (http://ecls.sourceforge.net/) Beyond looking over the code in the Freeswitch repository for the other embedded languages, does anyone have a recommendation on how to begin? I don't see any documentation specific to this task; does it exist somewhere? Thanks in advance for your help. Kevin Raison

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Interesting Blog About HD Telephony
by Michael Collins 06 May '09

06 May '09

FYI, Here's a nice story <http://www.freeswitch.org/node/182> for you all to check out. Please check it out and pass it on. -Michael

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ANNOUNCEMENT: FreeSWITCH 1.0.4pre7 Now Available
by Michael Collins 06 May '09

06 May '09

FYI, Please update your installations as soon as possible. More information on this update is available here <http://www.freeswitch.org/node/181>. Thanks for all of your feedback - please keep it coming and join us on IRC if you have any questions about the newest version. -Michael S Collins http://www.cluecon.com

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fifo in occasionally drops call
by Scott Shen 06 May '09

06 May '09

Using FS r13231, when testing fifo in, experienced occasionally call drops with no obvious reason, is it something to do with early media? See the debug log below Dialplan: sofia/internal/sip:1007@192.168.0.31:5342 Regex (PASS) [TEST_Q] destination_number(40001) =~ /^40001$/ break=on-false Dialplan: sofia/internal/sip:1007@192.168.0.31:5342 Action fifo(test_q in) 2009-05-06 15:18:46 [DEBUG] switch_core_state_machine.c:114 switch_core_standard_on_routing() (sofia/internal/sip:1007@192.168.0.31:5342) State Change CS_ROUTING -> CS_EXECUTE 2009-05-06 15:18:46 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/sip:1007@192.168.0.31:5342 [BREAK] 2009-05-06 15:18:46 [DEBUG] switch_core_state_machine.c:483 switch_core_session_run() (sofia/internal/sip:1007@192.168.0.31:5342) State ROUTING going to sleep 2009-05-06 15:18:46 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/sip:1007@192.168.0.31:5342) Running State Change CS_EXECUTE 2009-05-06 15:18:46 [DEBUG] switch_core_state_machine.c:490 switch_core_session_run() (sofia/internal/sip:1007@192.168.0.31:5342) State EXECUTE 2009-05-06 15:18:46 [DEBUG] mod_sofia.c:173 sofia_on_execute() sofia/internal/sip:1007@192.168.0.31:5342 SOFIA EXECUTE 2009-05-06 15:18:46 [DEBUG] switch_core_state_machine.c:151 switch_core_standard_on_execute() sofia/internal/sip:1007@192.168.0.31:5342 Standard EXECUTE EXECUTE sofia/internal/sip:1007@192.168.0.31:5342 set(outside_call=true) 2009-05-06 15:18:46 [DEBUG] mod_dptools.c:748 set_function() sofia/internal/sip:1007@192.168.0.31:5342 SET [outside_call]=[true] EXECUTE sofia/internal/sip:1007@192.168.0.31:5342 fifo(test_q in) 2009-05-06 15:18:46 [DEBUG] switch_ivr.c:1342 switch_ivr_session_transfer() (sofia/internal/sip:1007@192.168.0.31:5342) State Change CS_EXECUTE -> CS_ROUTING 2009-05-06 15:18:46 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/sip:1007@192.168.0.31:5342 [BREAK] 2009-05-06 15:18:46 [NOTICE] switch_core_state_machine.c:179 switch_core_standard_on_execute() Hangup sofia/internal/sip:1007@192.168.0.31:5342 [CS_EXECUTE] [NORMAL_CLEARING] 2009-05-06 15:18:46 [DEBUG] switch_channel.c:1641 switch_channel_perform_hangup() Send signal sofia/internal/sip:1007@192.168.0.31:5342 [KILL] 2009-05-06 15:18:46 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/sip:1007@192.168.0.31:5342 [BREAK] 2009-05-06 15:18:46 [DEBUG] switch_core_state_machine.c:490 switch_core_session_run() (sofia/internal/sip:1007@192.168.0.31:5342) State EXECUTE going to sleep 2009-05-06 15:18:46 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/sip:1007@192.168.0.31:5342) Running State Change CS_HANGUP 2009-05-06 15:18:46 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (sofia/internal/sip:1007@192.168.0.31:5342) State HANGUP 2009-05-06 15:18:46 [DEBUG] mod_sofia.c:323 sofia_on_hangup() Channel sofia/internal/sip:1007@192.168.0.31:5342 hanging up, cause: NORMAL_CLEARING 2009-05-06 15:18:46 [DEBUG] mod_sofia.c:378 sofia_on_hangup() Sending BYE to sofia/internal/sip:1007@192.168.0.31:5342 Regards Scott

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