- FreeSWITCH-users - FreeSWITCH Mailing Lists

Dialplan for OPTIONS packet
by Rajagopal, Sridhar (Sridhar) 02 Apr '09

02 Apr '09

Hi all, Whenever freeswitch recieves INVITE SIP packet, It forwards the packet based on the dial plan. I want to use the same dial plan to forward incoming OPTIONS packet. Please let me know If I need to write my own code for that or is there any such option in our code base. Regards, Sridhar

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about freeswitch conference
by bmsword 02 Apr '09

02 Apr '09

hi,all I want to use another softswitch conference that has been deployed in freeswitch,How should I do? thanks! andy 2009-04-02

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FreeSWITCH running on OpenWrt
by Carlos Talbot 02 Apr '09

02 Apr '09

Is there an interest in running FreeSWITCH on OpenWRT? I recently managed to compile and run a version for a MIPs based router, the Planex MZK-W04NU. This router has 32MB ram, 8MB flash, runs at 400MHz, draft N support (2.4GHZ), based on 2.6 kernels, a usb port and sells for about 60 bucks online. I used a 2GB USB flash drive for the FreeSWITCH directory and SWAP space. I was planning to setup a wiki page on compiling and configuring. regards, Carlos

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Missing CHANNEL_HANGUP event in mod_event_socket
by Victor Toofic 02 Apr '09

02 Apr '09

Hi all!! I'm stuck trying to use mod_event_socket in outbound mode. The problem that I'm facing is that while in a incoming call, using "myevents" to monitor for the channel's events.. the event CHANNEL_HANGUP sometimes arrives and sometimes doesn't. I can't figure it out why. The dialplan is: <extension name="9999"> <condition field="destination_number" expression="^9999$"> <action application="verbose_events" data="true"/> <action application="socket" data="192.168.251.1:8384 async full"/> </condition> </extension> The process that handles the connection does: 1. connect 2. myevents (received: Reply-Text: +OK Events Enabled) 3. sendmsg\n call-command: execute\n execute-app-name: answer (received: Reply-Text: +OK) after this it waits for events and/or for the other party to hangup the call. (The DTMFs are for testing propourses). Sometimes the events that the process receives are: <<"CHANNEL_PARK">> <<"CHANNEL_EXECUTE">> <<"CHANNEL_EXECUTE_COMPLETE">> <<"CHANNEL_EXECUTE">> <<"CHANNEL_ANSWER">> <<"CHANNEL_EXECUTE_COMPLETE">> <<"DTMF">> <<"DTMF">> <<"CHANNEL_HANGUP">> (then it receives the "text/disconnect-notice" and the socket gets closed) and sometimes are: <<"CHANNEL_EXECUTE">> <<"CHANNEL_EXECUTE_COMPLETE">> <<"CHANNEL_EXECUTE">> <<"CHANNEL_ANSWER">> <<"CHANNEL_EXECUTE_COMPLETE">> <<"DTMF">> <<"DTMF">> (then it receives the "text/disconnect-notice" and the socket gets closed) As you can see, even sometimes the first CHANNEL_PARK event doesn't arrive. I'm very concerned about the missing CHANNEL_HANGUP event. In the other hand I was watching the events in a inbound connection to mod_event_socket with "event text all" and in this case there was no problem, all the events arrived as expected. Why in outbound mode some events get lost?? I'm missing something?? I've tried it in two different machines and the results are the same. I'm using FreeSWITCH Version 1.0.3 (exported) on linux. Thnks!! -- Regards.. Victor Toofic

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Call For Help: Janitor Projects
by Michael Collins 01 Apr '09

01 Apr '09

Dear FreeSWITCH Community: As you know, FreeSWITCH has been growing leaps and bounds and it's going to keep growing as the word spreads. The core development team of Anthony, Mike, and Brian are very appreciative of the community's help and involvement in the project. Simply put: the community is awesome! Some have asked how they can help. Most of us are not software developers, but that doesn't mean we can't help to grow the FreeSWITCH ecosystem. To this end I've started a "janitor projects" wiki page: http://wiki.freeswitch.org/wiki/Janitor_Projects We say "janitor" projects because they are things that help keep the project clean and organized, just like the janitor cleans an office, takes out the trash, replaces the toilet paper, etc. These are valuable services that we sometimes take for granted. However, I think we can all appreciate that the FreeSWITCH project would be better served if the developers could focus on writing code, fixing bugs, etc. and not on the easier, not-quite-as-important janitorial tasks. To that end we are inviting all who wish to volunteer to please visit the above wiki page and check out some of the projects listed so far. Email me off list if you'd like to volunteer to help. I'm maintaining a list of "janitors" and what they are helping with. If you have ideas for other janitor projects then by all means email them to me and we'll discuss them. Thanks again for being such a great community! -Michael S Collins IRC: mercutioviz See you at ClueCon 2009! http://www.cluecon.com

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Another FreeSWITCH First!
by Anthony Minessale 01 Apr '09

01 Apr '09

The FreeSWITCH team is excited to announce that FreeSWITCH is the first telephony application to support the new SIP 4.1 protocol specification. Unlike its predecessors, SIP 4.1 has been created with the collaboration of both the jabber foundation and the IETF. With this match made in heaven, one can now encapsulate an xml representation of a sip message, which in turn can encapsulate a standard SIP 2.0 message making it possible to do more than ever before. Other exciting features include: *) RFC 4109 support: (SIP over SMTP), allowing packets to traverse NAT with ease. *) Full circle presence: endpoints must subscribe to each character in the printable ASCII range that may be used to indicate presence and the server will send an xml notification to the client for each character that is enabled whenever a call takes place which in turn can be used to build a SIP 4.1 FYI packet that can be sent to all the neighboring SIP devices so they may send themselves a NOTIFY telling them that the light should blink if the same packet happens to be sent from a neighbor. Then when the neighbor wants to send a presence packet it establishes a dialog with the Third Party Presence Agent TPPA and leaves the message there. Then it sends the server a PRESENCE packet, which is then, relayed to the subscribers with the TPPA id so all the subscribers can connect to the TPPA server to make the little light blink. *) Retirement of SDP: SDP is deprecated in favor of a list of URL’s describing the desired codec. The UA can then request this URL and get the full details of the media requirements. The media port is negotiated through trial and error where the calling UA asks the called UA if the port it has guessed randomly is correct via direct TCP connection and an exchange of XML PORT MARKUP LANGUGE XPML INVITE bob(a)alice.com SIP 4.1 Content-type: sip-xml-encapsulated <SIP version=”4.1”> <content type=”SIP-INVITE”> <INVITE recipient=”bob(a)alice.xn--com-9o0a> <data type=”sip-2/0”/> <![CDATA[INVITE bob(a)alice.com SIP 2.0 To: bob(a)alice.com From: alice(a)bob.com Subject: SIP Rocks ]]> </data> </INVITE> </content> </SIP> -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale(a)hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale(a)gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888(a)conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888(a)conference.freeswitch.org> pstn:213-799-1400

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Originate and Conference
by Peter P GMX 01 Apr '09

01 Apr '09

Hello, when I originate a call via event socket and transfer it to a conference, it doesn't ask for a PIN. Example: originate sofia/default/222331 &conference(222500) The same happens when I originate a call and transfer it to the 222500 destination, which then transfers the call to the conference. originate sofia/default/222331 &transfer(222500) If I manually dial from the phone (222331) to the conference it correctly asks for the PIN. Anybody has a clue why this happens and how to enable the PIN while originating? Best regards Peter

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Long Lost Comments Surface, Now We Know...
by Anthony Minessale 01 Apr '09

01 Apr '09

In one of the most suprising events in current technology history in this modern era, the long lost comments to many of the now-adopted internet RFC's have finally surfaced. Aparently the mail server was misconfigured at "The Internet Society" and most of the comments were redirected to the local lost-and-found box on the server whey they sat for decades. In a suprising twist, it appears that RFC 2543, the predacessor to 3261 regarding the session initnation protocol had several critisisms that went unanswered. A few examples are included below. Some other comments in regards to RFC2833 (RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals) could not be published due to the graphic nature of the content. From: alice(a)anywhere.com to:comments@tis.org <to%3Acomments(a)tis.org> Subject: RFC 2543 Ahem, who gave you permission to use my name in your document? Also, how did you find out about me and Bob? Thanks to you it's all over the net >=0 From: jwlt(a)columbia.edu to:comments@tis.org <to%3Acomments(a)tis.org> Subject: RFC 2543 Are you guys sure about this? We were pretty drunk last night. I didn't think you would actually go through with it! lol I was just kiddding! -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale(a)hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale(a)gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888(a)conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888(a)conference.freeswitch.org> pstn:213-799-1400

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Compiler error for Windows XP (SP2)
by Lewis Liu 01 Apr '09

01 Apr '09

We download FreeSWITCH from SVN Trunk and want to build it on MS Visual Studio 2008 with platform. But we got one error message when we build it. FreeSWITCH\libs\win32\sofia\debug\libsofia_sip_ua_static.lib is built fail. So many files are lost, such as mod_sofia.dll..... Could you help me me for this, Please?? Whether something is lost in MS Visual Studio 2008 ?? Thanks a lot!! Lewis

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FS and Skypiax on Windows Video How To
by Giovanni Maruzzelli 01 Apr '09

01 Apr '09

Kulwinder Singh contributed this HOW TO: Freeswitch & Skype- OS Microsoft Windows Download 118MB HD: http://www.celliax.org/final.avi Sincerely, Giovanni Maruzzelli ========================================= www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039

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