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403 forbidden on files.freeswitch.org
by Seven Du 14 Aug '09

14 Aug '09

Got this: Forbidden You don't have permission to access /cluecon_2009/presentations/ Dale_Building_FreeSWITCH_App_Lua.pptx on this server. Apache/2.2.3 (CentOS) Server at files-sync.freeswitch.org Port 80 $ ping files.freeswitch.org PING filessync.freeswitch.netdna-cdn.com (69.174.57.101): 56 data bytes 64 bytes from 69.174.57.101: icmp_seq=0 ttl=46 time=224.900 ms 64 bytes from 69.174.57.101: icmp_seq=1 ttl=46 time=221.597 ms ^C $ ping files-sync.freeswitch.org PING www-01.freeswitch.org (216.82.231.69): 56 data bytes 64 bytes from 216.82.231.69: icmp_seq=0 ttl=44 time=271.147 ms 64 bytes from 216.82.231.69: icmp_seq=1 ttl=44 time=273.161 ms ^C

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mod_managed users?
by Michael Giagnocavo 13 Aug '09

13 Aug '09

Hey, if there are any mod_managed users on this list, I'd love it if you were able to let me know. I'd like to get feedback, positive or negative, on what worked, what didn't, and how mod_managed can improve for you. Feel free to write on list or directly to me: mgg at giagnocavo.net Thanks! -Michael

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calling through same gateway with multiple registrations
by Timur Irmatov 13 Aug '09

13 Aug '09

Hi, I am new to FreeSWITCH and need an advice. All calls to PSTN from our server will go through single gateway, which is a soft switch supporting SIP protocol. FreeSWITCH will need to register with soft switch, but soft switch permits only single active call (in either direction) per registration. So we will need 10 SIP accounts to allow 10 simultaneous connections. Question is - how should I configure FreeSWITCH for this scenario? I see two options: 1) Create 10 gateways with different registrations, use mod_limit to route only one outgoing call per gateway; 2) Create 10 gateways with different registrations, use event socket to route calls manually and monitor used lines (incoming and outgoing calls through soft switch). Are there any other possibilities? Corrections/ suggestions are very welcome. -- Timur Irmatov, xmpp:irmatov@jabber.ru

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Confused about conferences
by Alan Chandler 13 Aug '09

13 Aug '09

I have been reading all the docs about conferences I can find and am getting somewhat confused. What I am trying to do is set up a dialplan where I have subscribers with extensions in the 1xx range, and then to set an ability to have a series of conference rooms for each subscriber in the 21xx range where if the user enters is "own" conference he is the moderator, but if not he is just a normal user. I want to be able for the moderator to do things like mute or kick people. So dialplan would probably have something like this in it <extension name="user_conference"> <condition field="destination_number" expression="^(2${caller_id_number})$"> <action application="set" data="i_am_moderator=true"> <action application="answer"/> <action application="conference" data="$1@default+flags{moderator}"/> </condition> <condition field="destination_number" expression="^(21\d{2})^"> <action application="set" data="i_am_moderator=false"> <action application="answer"/> <action application="conference" data="$1@default"/> </condition> </extension> <extension name="kick"> <condition field="i_am_moderator" expression="true"> <action application="play_and_get_digits" data="3 3 3 7000 # /ask-for-extension.wav /invalid.wav conf-user-id \d+" > WHAT GOES HERE??? </condition> </extension> in the conference.conf.xml file, I would change the caller controls to include <caller-controls> <group name="somekeys"> <control action="transfer" digits="9" data="kick XML default"/> </group> </caller-controls> My question (at the moment) is In the WHAT GOES HERE place how do it Kick extension ${conf-user-id} (DOES IT REQUIRE A SCRIPT TO CALL THE Conference API?) Re-Enter the moderator back into the conference Re-Enter the ordinary user who happened to press 9 back into the conference I am assuming I can't stop the non moderator getting the control - since all users get the same controls. -- Alan Chandler http://www.chandlerfamily.org.uk

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Grangstream Early Dial
by Yuriy Ivzhenko 13 Aug '09

13 Aug '09

Hello all. I want to use Grandstream Early Dial future. How i can enable support 484 response? I tried simply use <action application="hangup" data="484"/> and <action application="respond" data="484"/> on uncompleted extensions, but there is not work Thanks. Yuriy .

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Setting max inbound for UA
by String Larson 13 Aug '09

13 Aug '09

Is there a way to limit the number of calls a UA can receive in the FS configs? I'm doing some testing with XLite as the UA, and can not figure out how to keep line 2 from answering if line 1 is in use. THanks. -str

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Question about sharing conference between
by Tina Martinez 13 Aug '09

13 Aug '09

Thank you Michael, I will tinker around with it and definitely follow-up with the results. - T

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fs installation is broken in version: FreeSWITCH Version 1.0.trunk (14500M)
by Milena 13 Aug '09

13 Aug '09

I just did a rebootstrap on a fs box, it turned out the new revision has this at the end of mod_sofia.h: > char *sofia_glue_get_extra_headers(switch_channel_t *channel, const char > *prefix); <<<<<<< .mine void sofia_glue_set_extra_headers(switch_channel_t *channel, sip_t const > *sip, const char *prefix); > ======= void sofia_glue_set_extra_headers(switch_channel_t *channel, sip_t const > *sip, const char *prefix); >>>>>>> .r14490 no wonder why it wouldn't compile :P

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OpenZAP/Sangoma in Windows
by Peter Olsson 13 Aug '09

13 Aug '09

Hi, I'm trying to evaluate the OpenZAP/Sangoma-support in Windows, using PRI E1 connections. I'm aware it's not yet to be considered as stable, but I'd still want to try it out some, and also help detecting bugs while I'm at it :) Anyway, I have a couple of questions: 1. Has anyone tested it in Windows at all? I know the build-files for Visual Studio has only been checked in for a couple of months, so that's why I'm asking. 2. Does anyone have any directions how to configure the driver within Windows? Should I use BitStream or HDLC, and how should the channel groups be configured? I have built everything correctly, and I've succeeded to open (some of) the channels when I configure each channel in a separate group, but I don't think that's the way it should be done... Any suggestions welcome! /Peter

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Question about sharing conference between
by Tina Martinez 13 Aug '09

13 Aug '09

Michael, Thanks for the welcome, and for the response to my question. The call control and dynamic setup of conferences I have working (pretty cool stuff). The tricky part, as you said, is "linking" the servers together. Basically, what I need to do is establish a connection that will not be dependent on a live person being on the call. And I would prefer to avoid having to register actual phone extensions for every server -- and for every conference call. I apologize if I'm slow, but I'm new to working an application like this. In your example, you stated, "3900 is an extension set up in FS_B's dialplan", does this extension have to be a live person (or soft-phone connected using an auto-answer mechanism)? or can I setup something where there is not a phone actually connected? Right now I'm able to place a call from one machine to another, but I'm calling an X-Lite phone on the second server. Also, if it is possible to call a "virtual" extension, I have no problem incorporating application logic that would clean-up the orphaned conferences on all machines when a conference call is complete. I'm more concerned with being able to setup the "links" quickly and in an elegant fashion. - T

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