January 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Would like to pickup a call that is on hold on another extension
by Scott Ellis 16 Jan '09

16 Jan '09

I would like to be able to place a call on hold on one extension, walk to another phone and then dial a sequence (like the barge sequence) say 55+extension number and have the call taken off hold and transferred to the extension I am on. Has anyone done this? (Before I try and work it out for myself!) Scott

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Re: [Freeswitch-users] mod_managed failing to load on CentOS 5.2
by Tim B 16 Jan '09

16 Jan '09

Yeah I compile mono ... i tried both 2.0.1 and 2.2 both error on loading the mod_managed. Tim _________________________________________________________________ Windows Live™: Keep your life in sync. http://windowslive.com/explore?ocid=TXT_TAGLM_WL_t1_allup_explore_012009

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Freeswitch and CELT:
by Terrance Harris 16 Jan '09

16 Jan '09

Hello, I have recently found out about FS and how great it is. We are trying to use FS as a voip client for radio shows. We have been using Trixbox and Skype but Skype isn't getting it done. I have heard about how great the celt codec is but I don't have enough 'skill' to compile both FS and celt in MSVC++. Is there a binary out there that would make my day or a guide? Thanks much!

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No Sound Heared
by Klaus Teller 15 Jan '09

15 Jan '09

Hi, Need your help on this. I have the following Javascript statement: session.execute("bridge","sofia/gateway/sip.gafachi.com/someNumber") in a file called gafachiDialout.js Then, i have the following extension in default.xml: <extension name="6337"> <condition field="destination_number" expression="^6337$"> <action application="javascript" data="gafachiDialout.js" /> </condition> </extension> When i call this extension (6337), it rings as it should. But then there is NO sound going in either direction. Any idea what i'm doing wrong here? Thanks, Klaus. -- Sensationsangebot verlängert: GMX FreeDSL - Telefonanschluss + DSL für nur 16,37 Euro/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K1308T4569a

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mod_managed failing to load on CentOS 5.2
by Tim B 15 Jan '09

15 Jan '09

Got mod_managed compiled and installed. Now it isn't loading. See below... 1) Donwloaded fresh from SVN 2) Compiled... and installed.. OK [root@phone2 mod_managed]# export PKG_CONFIG_PATH=/usr/local/lib/pkgconfig [root@phone2 mod_managed]# make [root@phone2 mod_managed]# make install 3) Added to modules.conf.xml : <load module="mod_managed"/> 4) Started freeswitch from command line ... Error: 2009-01-14 20:01:42 [ERR] mod_managed.cpp:220 loadRuntime() mono_domain_assembly_open failed. 2009-01-14 20:01:42 [CRIT] switch_loadable_module.c:839 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_managed.so **Module load routine returned an error** 5) I know mono2 is working because I compiled and executed a helloworld test class on machine. Any ideas? _________________________________________________________________ Windows Live™: Keep your life in sync. http://windowslive.com/howitworks?ocid=TXT_TAGLM_WL_t1_allup_howitworks_012…

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Freeswitch crashed !!!
by ahgindia 15 Jan '09

15 Jan '09

Hi All, Recently I was testing the new freeswitch release 1.0.2 The system has Fedora 8 with 2 GB RAM with Intel(R) Core(TM)2 Duo CPU E7200 @ 2.53GHz processor. But it crashed, when there were 96 active calls in it (as can be seen from "show calls" on freeswitch cli) There is a dump file for it, in the folder from where i started the freeswitch. Let me know how can we know the cause of the crash. -- View this message in context: http://www.nabble.com/Freeswitch-crashed-%21%21%21-tp21386948p21386948.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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No Audio when dial out via gateway
by Will Smith 15 Jan '09

15 Jan '09

Hi, I have successfully installed and configured the FS thanks to the community help. Greatly appreciate all. Now I have some basic error: I can dial out from extension 1000 (all default ext) to any number not in the same network. I got the other number rung, and answered, but cannot hear anything from both ends. Strange thing is I can broadcast an audio into the conversation, and both ends can hear the audio, but just cannot talk. I just hope that it would be something easy to be pointed out by experienced users. Thank you, Will

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Re: [Freeswitch-users] Problem connecting with gtalk
by Milena 15 Jan '09

15 Jan '09

Hello, Isn't that what I did? if not, what is the right way to "install gnutls and dev packages and reconfigure/recompile" thank you

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waitForAnswer on the Socket Interface
by Klaus Teller 15 Jan '09

15 Jan '09

Hi, Can somebody tell me how to achieve the same behavuior as session.waitForAnswer via the socket interface? That is, when i call a device, i want to block until the call is completely answered (not just early media). Thanks, Klaus. -- Sensationsangebot verlängert: GMX FreeDSL - Telefonanschluss + DSL für nur 16,37 Euro/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K1308T4569a

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Problem connecting with gtalk
by Milena 15 Jan '09

15 Jan '09

Hello, I'm running fs 1.0.2 on CentOS 5.2 I've been trying to setup my fs to talk with googletalk following the instructions in http://wiki.freeswitch.org/wiki/Mod_dingaling#Sample_Configuration I got the error of TLS not supported so i: INSTALLED: yum install gnutls-devel gnutls REMOVED: rm -f /usr/src/freeswitch-1.0.2/libs/iksemel/.complete rm -f /usr/src/freeswitch-1.0.2/libs/libdingaling/.complete RE-INSTALLED cd /usr/src/freeswitch-1.0.2/ make sure make installall Even after this i keep getting the same error on the console. This is the error: 2009-01-15 14:06:02 [INFO] libdingaling.c:1304 on_log() RECV: ------------------------------------------------------------------------------- <stream:stream from="gmail.com" id="E607DA864F13575A" version="1.0" xmlns:stream="http://etherx.jabber.org/streams" xmlns="jabber:client"></stream:stream> 2009-01-15 14:06:02 [DEBUG] libdingaling.c:1175 on_stream() TLS NOT SUPPORTED IN THIS BUILD! 2009-01-15 14:06:02 [INFO] libdingaling.c:1304 on_log() RECV: ------------------------------------------------------------------------------- <stream:features> <starttls xmlns="urn:ietf:params:xml:ns:xmpp-tls"> <required></required> </starttls> <mechanisms xmlns="urn:ietf:params:xml:ns:xmpp-sasl"> <mechanism>X-GOOGLE-TOKEN</mechanism> </mechanisms> </stream:features> and now when i *shutdown* my fs, the core gets dumped when trying to stop mod_dingaling: 2009-01-15 14:14:21 [CONSOLE] switch_loadable_module.c:1231 do_shutdown() Stopping: mod_dingaling Segmentation fault Thank you for any help or suggestions you can give me.

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