April 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Database schema
by Richard Lamkin 02 Apr '09

02 Apr '09

Are there documents or wiki page [I've missed during my searches] that detail the records and their types that are stored in the various FS databases; e.g. sofia_reg_<profile name>.db, core.db ? Regards Richard Lamkin ************************************************************************* This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. http://www.mettoni.com Datapulse Ltd (part of the Mettoni Group) Registered in England and Wales: 4485978 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN *************************************************************************

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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? andy 2009-04-02

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mod_shout delay in trunk
by Rupa Schomaker 02 Apr '09

02 Apr '09

I've setup a conference bridge that has perpetual-sound set to a icecast stream. When the first person connects, there is an ~7s delay before any audio is heard. This is similar to a problem reported by Dan here and concluded with Tony adding the channel var "enable_file_write_buffering". The list discussion ended here: http://lists.freeswitch.org/pipermail/freeswitch-users/2009-February/011104… I set this var in my dialplan: <action application="set" "enable_file_write_buffering=false"/> prior to joining the conference. The first person in still sees a 7s delay on audio the first time in. Like dan, I have icecast setup with burst_on_connect set to 1 but my burst_size is the default 64k so quite a bit of data. Has anyone been able to get an on-demand shoutcast stream from an icecast server to start immediately (or at least within a second)? Thanks. -- -Rupa

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new module: mod_memcache
by Rupa Schomaker 02 Apr '09

02 Apr '09

Announcing a new module: mod_memcache Up until now one had two choices for storing arbitrary key/value pairs. hash or db. hash is fast, but it is local to the current FreeSWITCH instance. If you run multiple instances of FreeSWITCH then one could use db, an ODBC connection and a centralized database server (eg: postgresql). The choice was between fast but isolated or slow and distributed. memcached (http://www.danga.com/memcached/) is a high-performance, distributed memory object caching system, generic in nature, but intended for use in speeding up dynamic web applications by alleviating database load. Only now you can use it for dynamic phone applications. Check out the wiki page at: http://wiki.freeswitch.org/wiki/Mod_memcache Try this module out and file bug (jira) reports for problems / enhancement requests. -- -Rupa

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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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