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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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mod_sofia: NAT-ping & RPID bounties
by kokoska rokoska 15 Jan '09

15 Jan '09

Hi all, I have just post two bounties (NAT-ping and RPID changes). So I hope someone may be interested :-) BTW: Both changes have to be (and stay in the future) integral part of FreeSWITCH code. Best regards, kokoska.rokoska

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Using mod_managed Linux/Mono 2.02
by Adam Long 15 Jan '09

15 Jan '09

Has anyone had any luck using mod_managed under linux with mono yet? The Wiki looks to still be lacking some linux installation instructions. I feel like I'm close but missing something simple. I got as far as adding "languages/mod_managed" to the /usr/src/freeswitch-1.0.2/modules.conf without quotes obviously. My installed mono version is [root@sipcore-alpha mod]# mono -V Mono JIT compiler version 2.2 (tarball Wed Jan 14 09:44:57 PST 2009) Copyright (C) 2002-2008 Novell, Inc and Contributors. www.mono-project.com TLS: __thread GC: Included Boehm (with typed GC) SIGSEGV: altstack Notifications: epoll Architecture: x86 Disabled: none I can successful compile freeswitch and it indeed compiles mod_managed.so I added <load module="mod_managed" /> to my /usr/local/freeswitch/conf/autoload_configs/modules.conf.xml I did also create the /usr/local/freeswitch/mod/managed directory as stated in the wiki as requirement. But when I start freeswitch I get the following in regards to the mod_managed loading. 2009-01-14 14:19:12 [INFO] mod_managed.cpp:309 mod_managed_load() Loading mod_managed (Common Language Infrastructure), Mono Version 2009-01-14 14:19:12 [INFO] mod_managed.cpp:213 loadRuntime() Calling mono_assembly_loaded. 2009-01-14 14:19:12 [INFO] mod_managed.cpp:217 loadRuntime() Calling mono_domain_assembly_open. 2009-01-14 14:19:12 [ERR] mod_managed.cpp:220 loadRuntime() mono_domain_assembly_open failed. 2009-01-14 14:19:12 [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** One thing I think I may be missing is a FreeSWITCH.Managed.dll (this exists on windows environment but doesn't seem to be compiled under linux) I thought perhaps mod_managed_lib.dll was the linux equivalent but that exists and still no a no go. Any ideas would be very welcome? Thank you! Regards, -Adam

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No caller/called ID received (Wildcard X101P)
by Tomás 15 Jan '09

15 Jan '09

Hi all, I'm a new FreeSwitch user and this is my first email to the list. I'm trying to configure my Home PBX with a Wildcard X101P (configured as FXO) and I have a problem receiving the caller/called ID from PSTN. This is the content of file "openzap.conf": [span zt] name => OpenZAP number => 1 fxo-channel => 1 And this is the content of file "openzap.conf.xml": <configuration name="openzap.conf" description="OpenZAP Configuration"> <settings> <param name="debug" value="0"/> </settings> <analog_spans> <span id="1"> <param name="tonegroup" value="es"/> <param name="digit-timeout" value="2000"/> <param name="max-digits" value="11"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="enable-callerid" value="true"/> </span> </analog_spans> </configuration> As you can see the param "enable-callerid" is set to "true", but when I received and incoming call, FreeSwitch doesn't get neither the caller number nor the called number (instead of my home number, I receive a number 1, as can be seen on the following log of an incoming call): 2009-01-14 20:43:12 [NOTICE] switch_channel.c:565 switch_channel_set_name() New Channel OpenZAP/1:1/1 [941d6234-e273-11dd-bcdf-89190a30fe61] 2009-01-14 20:43:12 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing OpenZAP->1 in context default 2009-01-14 20:43:12 [NOTICE] switch_ivr.c:1255 switch_ivr_session_transfer() Transfer OpenZAP/1:1/1 to enum[1@default] 2009-01-14 20:43:16 [INFO] switch_core_state_machine.c:122 switch_core_standard_on_routing() No Route, Aborting 2009-01-14 20:43:16 [NOTICE] switch_core_state_machine.c:123 switch_core_standard_on_routing() Hangup OpenZAP/1:1/1 [CS_ROUTING] [NO_ROUTE_DESTINATION] 2009-01-14 20:43:16 [NOTICE] switch_core_session.c:960 switch_core_session_thread() Session 1 (OpenZAP/1:1/1 ) Ended 2009-01-14 20:43:16 [NOTICE] switch_core_session.c:962 switch_core_session_thread() Close Channel OpenZAP/1:1/1 [CS_HANGUP] 2009-01-14 20:43:23 [NOTICE] switch_channel.c:565 switch_channel_set_name() New Channel OpenZAP/1:1/1 [9af53280-e273-11dd-bcdf-89190a30fe61] Any idea of what's the problem? Thank you very much in advance.

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Changes in PlayAndGetDigits
by Juan Backson 15 Jan '09

15 Jan '09

Hi, Is there a change in the playAndGetDigits api? In the old release, 11102, my lua script is working but is not working in the latest release. The error I am getting is " Error in playAndGetDigits expected 10..10 args, got 9 ". Thanks, JB

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Country specific tones - how to contribute?
by Scott Ellis 15 Jan '09

15 Jan '09

I have tracked down a set of au tones from the mailing list, which I am going to verify. How do I go about getting these added into the default build so that they are available for all in future? I tried <action application="set" data="ringback=${au-ring}"/> and this did not work - where does it try and load the ring tone from? I have entries in the tones.conf file, but these do not seem to be used. Scott

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