November 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Help with home setup
by Jacob Smith 10 Nov '11

10 Nov '11

I have been trying to get this working for the last week and would really appreciate your help. I have a pfSense 2.0 server with Freeswitch and FusionPBX installed. I followed the instructions at http://wiki.fusionpbx.com/index.php?title=PfSense_Install and can access the web GUI. My goal is to use a Gigaset C610A-IP connected to my Google Voice account. However, my first failure seems to be the most basic: through FusionPBX, I created a user and extension to use with the phone but I can not register it with FreeSwitch. So, how do I get the phone to work so I can test that the basic functions of FreeSwitch are working and move on? Thanks!

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Voice mail migration
by Lloyd Aloysius 10 Nov '11

10 Nov '11

HI, I am in the process of moving my current freeswitch server to a new server. What is the best way to migrate the VoiceMail from old to new server? Any help is appreciated. Thanks and regards, Lloyd

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fwd: call groups - sort order
by Weigel, Stefan 10 Nov '11

10 Nov '11

Hi list, sorry for pushing this one, but I would really appreciate any hints on this... Thanks and best regards, Stefan Von: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] Im Auftrag von Weigel, Stefan Gesendet: Mittwoch, 9. November 2011 09:59 An: FreeSWITCH Users Help Betreff: [Freeswitch-users] call groups - sort order Hi list, from the WIKI page: group call:groupname[:order] How are the entries sorted ? How can I make sure the list returned by 'group call' is sorted in a certain order ? I want to make sure a certain extension is ringed first, but sometimes another extension is listed first. Thanks and best regards Stefan

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High load on database server
by Tim St. Pierre 10 Nov '11

10 Nov '11

Hello, I'm having some issues with the number of queries that Freeswitch generates per-second. Currently, I have about 160 phones registered to FS. These are almost all behind NAT, and all of them have to share presence (there are 2 profiles depending on which network they are on), as well as get options pings and that sort of thing. Database is MySQL 5.5. I'm seeing 18 queries per second, mostly complicated internal joins on tables like sip_dialogs and sip_subscriptions. Freeswitch is Version 1.0.head (git-7531fed 2011-08-17 11-27-20 -0500) and hardly breaking a sweat, while the database server is at 100% CPU load. If I stop the internal profiles, the database server goes right back down. When I start up the profiles again, the load slowly climbs back up to near 100%. The net result of this is that sofia doesn't always return it's options ping, (so other circuits think it's down), and we get slow / no call setup. Any suggestions as to how I can reduce this? -Tim

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Patch_Up_Two_Files
by Pankaj Upadhyay 10 Nov '11

10 Nov '11

Hi all ,

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make current fails with 'config.sub not found'
by Vitalie Colosov 10 Nov '11

10 Nov '11

Today decided to upgrade FS, after almost 5 months from last upgrade, and executed 'make current' However, it failed with the following errors: ... create mode 100644 src/mod/event_handlers/mod_cdr_mongodb/mod_cdr_mongodb.c delete mode 100644 src/mod/timers/.empty create mode 100644 swig_common.i make[1]: Leaving directory `/usr/local/src/freeswitch' make all make[1]: Entering directory `/usr/local/src/freeswitch' cd . && /bin/sh /usr/local/src/freeswitch/build/config/missing --run aclocal-1.9 cd . && /bin/sh /usr/local/src/freeswitch/build/config/missing --run automake-1.9 --gnu *configure.in:27: required file `build/config/config.sub' not found* make[1]: *** [Makefile.in] Error 1 make[1]: Leaving directory `/usr/local/src/freeswitch' make: *** [current] Error 2 Could you please hint what has changed since then or what I am doing wrong? Thank you, Vitalie

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Holding Call With Bypass Media True
by Jerry Richards 09 Nov '11

09 Nov '11

Hi All, I am seeing an issue where Freeswitch is disconnecting a call when one endpoint tries to put the call on-hold (inbound-bypass-media is true and one phone has a secondary extension of the other phone). I'm just wondering if there is a rule/restriction for bypass media I'm not aware of. If I change the configuration to inbound-proxy-media true then the call is put on-hold correctly. If I set it back to inbound-bypass-media true and then remove the secondary extension from Phone A (below), then the call is put on-hold correctly. Here is the scenario: 1) Phone A registers primary extension 1002 and secondary extension 1003 2) Phone B registers only primary extension 1003 3) A calls B 4) B answers 5) A holds call 6) Freeswitch sends BYE to both A and B with reason DESTINATION_OUT_OF_ORDER (Also it says: "Re-INVITE to a no-media channel that is not in a bridge") Thanks, Jerry

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mod_rtmp compile error on Ubuntu 11.10
by Matt Stockton 09 Nov '11

09 Nov '11

Has anyone successfully built mod_rtmp on Ubuntu 11.10? I'm seeing the following errors: making install mod_rtmp Compiling rtmp.c... rtmp.c: In function ‘rtmp_send_message’: rtmp.c:529:18: error: variable ‘status’ set but not used [-Werror=unused-but-set-variable] cc1: all warnings being treated as errors I successfully compiled when I inserted a hack to use the variables into rtmp.c and mod_rtmp.c, but wondering what the right way to fix this is. Thanks, Matt

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Hunting Phase Problem
by hi tanin 09 Nov '11

09 Nov '11

I am very new with FreeSwitch. Can anyone tell me: What is mean by hunting phase? When and how it is use and why and when it is required? I want to know details about hunting phase, Also plese give me a way or tell me when and how the variables($0, $1, $2) are used? Please help me

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Best way to create a one to many 'broadcast'
by Andy Ayers 09 Nov '11

09 Nov '11

Hi, I would like to use freeswitch to host a one to many broadcast style conference call. The challenge I have is that there won't always be any listeners so the conference application isn't quite working for me. Essentially, the speaker dials in with something important to say, by default the system will simply record what they say as an archived message and then they hang up. This works fine but I would like to extend the functionality so that other people dialling in have the option to connect into this call and listen to the message live. Can you advise me on the best way of achieving this: conference, bridge, eavesdrop or something else? Many thanks Andy

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