October 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

hangup in event socket
by Gopal krishnan 06 Oct '08

06 Oct '08

Hi, I am trying to hangup a bridged call via event socket, but the channel is not getting hangup. I tried with the following commands in telnet, sendevent channel_hangup - no output sendevent hangup - no output sendmsg <uuid> any other manager commands are there to hangup a channel? -- Thank you with regards, Gopal,

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unload of mod_xml_cdr doesn't work properly - cdr's get posted twice
by Leon de Rooij 06 Oct '08

06 Oct '08

Hello all, I've been trying out mod_xml_cdr, and found out that if I unload and again load the module, then cdr's get posted twice to the webserver (or more often, depending on the amount of times the xml_cdr module has been unloaded/loaded). (I'm sure I'm sending back 200/Ok's) I only tested it when loading the module the first time from post_load_modules (through mod_xml_curl) the first time, and after that unloaded and loaded the mod_xml_cdr module from the cli. When unloading, the cli also gives this message: 2008-10-06 14:20:05 [CONSOLE] switch_loadable_module.c:1131 do_shutdown() mod_xml_cdr has no shutdown routine 2008-10-06 14:20:05 [CONSOLE] switch_loadable_module.c:1137 do_shutdown() mod_xml_cdr unloaded Does anyone have this same problem ? (I'm rebuilding the latest SVN update now, but it takes a while before I can test it) thanks & regards, Leon

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strange "show calls" on svn head
by jay binks 06 Oct '08

06 Oct '08

can anybody suggest how to fix this .. FreeSWITCH>show calls created,created_epoch,function,caller_cid_name,caller_cid_num,caller_dest_num,caller_chan_name,caller_uuid,callee_cid_name,callee_cid_num,callee_dest_num,callee_chan_name,callee_uuid 2008-10-06 22:24:20,1223295860,switch_ivr_multi_threaded_bridge,,,,,,,,,, 1 total. I have FreeSwitch Version 1.0.trunk (9828) I tried deleting the database and restarting, and not much help :( any ideas would be great.. -- Sincerely Jay

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Newbie: Autoconf error when installing Freeswitch on Mac OS
by John Lum-Wah 06 Oct '08

06 Oct '08

Hi, I'm new to Freeswitch. I tried installing Freeswitch using the commands from the wiki svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch 1) su root % use sudo passwd root to reset the root password if you don't know the password <password> 2) cd /usr/local 3) mkdir src % create /usr/local/src directory 4) cd src svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch 5) cd freeswitch ./bootstrap.sh I still get the error that autoconf 2.59 or above is needed. I even tried the command AUTOCONF=/usr/bin/autoconf ./bootstrap.sh However, I think this assume that I have an autoconf file in /usr/bin. Needless to say, I still get the error messages that autoconf 2.59 or above is required. Can anyone help me on how to solve this configuration error. /john

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reload/ restart gracefully from CLI
by Sheeju Alex 06 Oct '08

06 Oct '08

If there is any changes in source file say "mod_conference.c" is it required to restart freeswitch? I think it is required to load the module again... These are the commands I use to restart freeswitch //STOP # freeswitch -stop //START # freeswitch Is there any other command where I can directly restart from Freeswitch CLI? /Sheeju

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Routing with $1 variable
by Peter P GMX 05 Oct '08

05 Oct '08

Today I had a strange behaviour: I am routing calls to an Asterisk PBX. It has a very sophisticated dynamic least cost router built in - so I use it to terminate mobile and international calls. For a first test I created a dialplan which checks for ^231$ in the dialled number and then routes the call to $1/asterisk.pbx. It didn't work. Grepping the SIP traffic I could see that Freeswitch makes an invite $1(a)asterisk.pbx. Yes it's $1 in the SIP message. Second I checked for the mobile number using ^(01[6-7]\d{4,13})$ in the dialled number and then route the call to $1/asterisk.pbx. This works. I am confused as in the second case $1 is translated to the dialled number and in the frist case not. Anybody has a clue? Best regards Peter

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<caller-controls> -- Invalid caller control action name 'dial'.
by Sheeju Alex 05 Oct '08

05 Oct '08

Hi All, I am trying to include "dial" caller control in my custom profile, but I get this warning and looks like this action is not defined? 2008-10-04 18:42:11 [WARNING] mod_conference.c:4797 conference_new_install_caller_controls_custom() Invalid caller control action name 'dial'. Here is the caller-controls I included as mentioned in wiki <control action="dial" digits="888" data="sofia/default/888(a)conference.freeswitch.org 60 0000000000 FreeSwitch"/> Thanks, Sheeju

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GotoIf
by henkoegema 05 Oct '08

05 Oct '08

How do I do this in FS ? : exten => s,n,GotoIf($[${CALLERID(num)}=32484120169]?default,1000,1) exten => s,n,GotoIf($[${CALLERID(num)}=4732803429]?default,terje,1) exten => s,n,GotoIf($[${CALLERID(num)}=32476378561]?henk_gsm_disa,s,1) The Asterisk to Freeswitch Rosetta stone says: GotoIf Conditions in dialplan (<condition field="blah" expression="foo"> but I can't figure out the right syntax. Henk -- View this message in context: http://www.nabble.com/GotoIf-tp19793170p19793170.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Re: [Freeswitch-users] Load test - performance not even matching Asterisk
by Mark Crane 04 Oct '08

04 Oct '08

"your last response is surprising me. I appreciate the work you have done in supporting people doing load testing. " Countless hours at no cost to you or me have been put in by Anthony, Brian, Mike and many others in building documenting and supporting FreeSWITCH. If you truly do appreciate the work then join the community and do things that help make it better. These include helping with documentation as you learn it, helping to support it or contributing to it with money. Load testing has not been done by many in the community. So it is logical that the road less used will have less documentation. As you learn the load testing you should document it on the wiki. There has been enormous man hours put into this documentation already. Hundreds of pages worth of documentation and examples are already available to anyone. A suggestion if you choose to be a part of the FreeSWITCH community. Recognize that some things should help Keep out comments that portray a feeling of entitlement and word your questions and comments with a little gratitude. Then as people voluntarily answer your questions give back to the community in some way. Mark --- On Sat, 10/4/08, Jon Bruel <jbr(a)consiglia.dk> wrote: > From: Jon Bruel <jbr(a)consiglia.dk> > Subject: Re: [Freeswitch-users] Load test - performance not even matching Asterisk > To: freeswitch-users(a)lists.freeswitch.org > Date: Saturday, October 4, 2008, 3:22 PM > Anthony, your last response is surprising me. I appreciate > the work you > have done in supporting people doing load testing. But the > information > about this is rather unorganized and difficult to access > for newbies. > Further, many threads on this and other topics are written > in telegram > style language, which increases the learning time. As an > example, the > first time I heard about the testing tool sipp was > yesterday - through > the users list - thanks. What a great tool! The > documentation though, is > not aimed at making the learning curve easy. A "these > are the 20 steps > to do test" manual does not exist to my knowledge. > Bear in mind that I'm > not a programmer but a generalist with all-round knowledge > about > telecoms and virtual PBXs. > I hope you can use this feedback in a productive way in > order to improve > the overall level of documentation. And it also pinpoints > the need for > the userlist and repetitive questions. > Setting up commercial relations also makes sense when we > have reached > the proof of concept and a business plan can be made. The > proof of > concept includes a decision about which switch to use, > Asterisk is still > an alternative to FS. Asterisk may have some architectural > drawbacks, > but it has been tested for a longer time and in more setups > than FS. > Further, Asterisk has a bad reputation. > As you have mentioned before, it may be worth while > visiting your team > soon, and I'm preparing for that. /Jon > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org

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Re: [Freeswitch-users] Load test - performance not even matching Asterisk
by Jon Bruel 04 Oct '08

04 Oct '08

Anthony, your last response is surprising me. I appreciate the work you have done in supporting people doing load testing. But the information about this is rather unorganized and difficult to access for newbies. Further, many threads on this and other topics are written in telegram style language, which increases the learning time. As an example, the first time I heard about the testing tool sipp was yesterday - through the users list - thanks. What a great tool! The documentation though, is not aimed at making the learning curve easy. A "these are the 20 steps to do test" manual does not exist to my knowledge. Bear in mind that I'm not a programmer but a generalist with all-round knowledge about telecoms and virtual PBXs. I hope you can use this feedback in a productive way in order to improve the overall level of documentation. And it also pinpoints the need for the userlist and repetitive questions. Setting up commercial relations also makes sense when we have reached the proof of concept and a business plan can be made. The proof of concept includes a decision about which switch to use, Asterisk is still an alternative to FS. Asterisk may have some architectural drawbacks, but it has been tested for a longer time and in more setups than FS. Further, Asterisk has a bad reputation. As you have mentioned before, it may be worth while visiting your team soon, and I'm preparing for that. /Jon

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