September 2009 - freeswitch-dev@lists.freeswitch.org

[Freeswitch-dev] portaudio and fifo
by vip killa 11 Sep '09

11 Sep '09

is there anyway to originate a call from portaudio to fifo?

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[Freeswitch-dev] port audio auto_answer
by vip killa 11 Sep '09

11 Sep '09

I have a simple question. is there anyway to have port audio automatically answer a call when an extension is dialed?

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[Freeswitch-dev] Pin-less entry into conference
by Robert Joly 11 Sep '09

11 Sep '09

Hi, I work on a SIP PBX project that is using FreeSWITCH as a audio conference server which is working great. The audio conferencing is very well integrated in out web-based configuration server and allows the owner of the conference to control it (mute, unmute, isolate, ...) from a browser. One other neat thing a conference owner can do through the browser is to invite additional parties to the conference by typing in a SIP URI. This causes our SIP PBX to do 3rd party call control to refer the invited party to the proper conference. This works very well for conferences that do not require a PIN but breaks for conferences that do. In the latter case, when the invited party answers the call, he hears "please enter your conference access number" out of context. Given that this scenario is less that ideal I started looking for solutions what would allow someone to join a 'PIN-guarded' conference without being prompted for one. I looked at a variety of dialplan-based options without success and turned to another option which I successfully prototyped. The purpose of this post is to socialize the solution I came up with to get feedback and to ultimately learn what needs to be done to get this functionality committed in the code base. The high-level solution I came up with consists the following: 1- add the conference PIN to the INVITE that is sent to FreeSWITCH; 2- have conference dialplan set a well-known dialplan variable with the content of the PIN as extracted from the INVITE; 3- have mod_conference test the pin contained in the well-known dialplan variable before deciding to prompt the user to enter a PIN. The solution I prototyped supports two ways of adding the PIN to an INVITE. First, the pin can be included as a well-known URL parameter added to the INVITE, e.g. INVITE sip:myconf@myfreeswitch.local;X-ConfPin=1234 SIP/2.0. When the PIN is passed as a URL parameter, the conference dialplan needs to be of the following form to allow PIN-less entry: <extension name="700"> <condition field="destination_number" expression="^700$"> <action application="set" data="supplied_pin=${sip_req_params}"/> <action application="conference" data="myconf@700+1234"/> </condition> </extension> The other way of adding a PIN to an INVITE is to add a user-chosen proprietary header containing the PIN to the INVITE, e.g. X-My-Custom-Pin-Header: 1234. When the PIN is passed as a header, the conference dialplan needs to be of the following form to allow PIN-less entry: <extension name="700"> <condition field="destination_number" expression="^700$"> <action application="set" data="supplied_pin=${sip_h_X-My-Custom-Pin-Header}"/> <action application="conference" data="myconf@700+1234"/> </condition> </extension> Whichever method is taken, the end results are similar, i.e. the 'supplied-pin' dialplan variable is initialized with the conference PIN passed in the INVITE. All that is left to do is for the mod_conference to look-up the variable and extract the PIN out of it if non-empty and use it as though the user entered it from his keypad. If the PIN is not the correct one, the logic defaults to prompting the user. The mod_conference patch is attached to this post which essentially consists in adding 10 lines of code (if you ignore the lines changed because of indentation). Does this approach sound reasonable to you? If so, I'll open a tracker describing the problem and solution and attach my patch to it so that it can be committed. Thank you very much in advance, Robert

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[Freeswitch-dev] Friday Meeting at 11AM CST
by Brian West 11 Sep '09

11 Sep '09

http://wiki.freeswitch.org/wiki/FS_weekly_2009_09_11 Here is the agenda please review and add to it anything you think we should cover. Thanks, Brian

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[Freeswitch-dev] REMINDER: Weekly FreeSWITCH Conference Scheduled for Friday, 11AM CST
by Michael Collins 11 Sep '09

11 Sep '09

FYI, We are on schedule for the weekly conference call. Please be sure to join us! More information is available here: http://wiki.freeswitch.org/wiki/FS_weekly_2009_09_11 Talk to you all tomorrow! -Michael

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[Freeswitch-dev] ANNOUNCEMENT: Friday Meeting Agenda - 2009-09-11
by Michael Collins 10 Sep '09

10 Sep '09

FYI, The Friday meeting agenda is posted here: http://wiki.freeswitch.org/wiki/FS_weekly_2009_09_11 Please add agenda topics as needed. We invite all to join us this coming Friday at 11AM CST. -Michael

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[Freeswitch-dev] [ERR] mod_sofia.c:2645 Invalid Gateway
by Jerry Richards 09 Sep '09

09 Sep '09

I have phones registered internally and can call among them. However, when I dial "711" from an internal phone, freeswitch replies with "484 Address Incomplete" with reason "INVALID_NUMBER_FORMAT". At the server console, I see the following error: [ERR] mod_sofia.c:2645 Invalid Gateway Does anyone know why I get this error? Is there something more I must do to add the gateway below? I already added the following to the usr/local/freesitch/conf/dialplan/default.xml: <extension name="Testing - Mediant 1000"> <condition field="destination_number" expression="^(711)$"> <action application="bridge" data="sofia/gateway/mediant1000/$1"/> </condition> </extension> I already created a usr/local/freeswitch/conf/sip_profiles/external/mediant1000.xml file: <include> <gateway name="192.168.72.253"> <param name="username" value="TEOGateWay"/> <param name="password" value="tI0W*Cj@b"/> <param name="register" value="false"/> </gateway> </include> Best Regards, Jerry

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[Freeswitch-dev] Using mod_dingaling with mod_xml_curl
by Muhammad Shahzad 09 Sep '09

09 Sep '09

Hi, Does dl_login and dl_logout commands work if mod_dingaling is configured with mod_xml_curl? It does not seems to work till i reload mod_dingaling. Here is what i am doing. 1. Using SVN Revision 15760. 2. mod_xml_curl configured to provide configuration for mod_dingaling. 3. When i issue command from FS console dl_login profile=XYZ, it suppose to use mod_xml_curl to fetch configuration from web server for mod_dingaling profile XYZ. But it never happens and dl_login command results in FAIL. Kindly suggest a solution. Thank you. -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk(a)hotmail.com Email: shaheryarkh(a)googlemail.com

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[Freeswitch-dev] Building Linux Centos Installation Package(s)
by Jerry Richards 08 Sep '09

08 Sep '09

After I have built my local Freeswitch version on my Centos machine, how do I create installation package(s) so that it can easily be installed on another generic Centos Linux server (without needing to recompile/rebuild it)? Best Regards, Jerry

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Re: [Freeswitch-dev] FreeSWITCH-dev Digest, Vol 39, Issue 4
by Sergio Alecha 08 Sep '09

08 Sep '09

hi. Duda with mod_opal How do I set the DID "FreeSWITCH" for the real phone number? 1 - I run FreeSWITCH with the user freeswitch 2 - I have a conection h323 between FreeSWITCH and Cisco CallManager. 3 - Start a call from an IP phone (2610) registered with the Cisco Call Manager, to an IP phone (2910) registered with the FreeSWITCH. 4 - In 2610 the display of the phone I get: 2610 FreeSWITCH 5 - Instead of FreeSWITCH should see "2910" What is the problem? How can I fix this makeshift?

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