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can't load mod_mISDN in freeswitch....
by Tamer Higazi 27 Sep '13

27 Sep '13

Hi people! I have successfully compiled for my openvox BRI board mISDN v2 and got the module from "stkn" compiled. Still I have problems loading the module successfully and get this output: http://pastebin.com/raw.php?i=d4RkgVCz My Freetdm.conf.xml: http://pastebin.com/raw.php?i=CjeCwd1P <configuration name="freetdm.conf" description="FreeTDM Configuration"> <settings> <param name="debug" value="0"/>    </settings> <libpri_spans> <span name="BRI_1"> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="mode" value="user"/> <param name="dialect" value="euroisdn"/> <param name="ton" value="unknown"/> <param name="overlapdial" value="yes"/> </span> <span name="BRI_2"> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="mode" value="user"/> <param name="dialect" value="euroisdn"/> <param name="ton" value="unknown"/> <param name="overlapdial" value="yes"/> </libpri_spans> </configuration> and my freetdm.conf: http://pastebin.com/raw.php?i=Kj2hmrmL If anyone of you have an idea what I did wrong, I would kindly thank you. Tamer

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Freeswitch register as UA
by Murthy Gandikota 27 Sep '13

27 Sep '13

Hi All I want Freeswitch to register with Asterisk as a user agent. To illustrate with Yealink phone I have the following configs: a) username: cs3 b) SIP server: fs-murthy.nts.net c) port: 5060 d) the ip of YeaLink phone itself (xx.xx.85.130) Now I want Freeswitch to register with Asterisk with similar data. Since I don't know as much about Asterisk as I know about Freeswitch, my goal is to bounce a call from Asterisk to Freeswitch and let it play an IVR. Here are some links I already checked: https://wiki.freeswitch.org/wiki/Connecting_Freeswitch_and_Asterisk wiki.freeswitch.org/wiki/Freeswitch_softphone Any help gratefully appreciated. Murthy

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Calls disconnected with NORMAL_TEMPORARY_FAILURE
by Adrian Andrei 27 Sep '13

27 Sep '13

Hello, I have the following setup: CISCO gateway -> FS-> SIP endpoint Some of the calls are perfectly connected with long average call duration but some them are disconnected with NORMAL_TEMPORARY_FAILURE with very low ACD. I don't know if is a FS problem or SIP endpoint issue. In FS is enable param inbound_late_negotiation and in dial_plan bridge_early_media=true is true. The codec negotiation should be made between CIsco and SIP endpoint Here is the logs for one sample call disconnected with Norm_temp_failure http://pastebin.com/HcUwUJHQ Thank you.

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Not really a Freeswitch issue, but...
by Alex Lake 27 Sep '13

27 Sep '13

...I was wondering if any kind souls would be able to suggest some next steps for a problem I have with a lua script. It uses a third-party library called LuaXML, but my module seems to be the wrong architecture, with the result that I get this error message: 2013-09-27 08:10:08.598859 [ERR] mod_lua.cpp:200 error loading module 'LuaXML_lib' from file '/usr/local/lib/lua/5.1/LuaXML_lib.so': /usr/local/lib/lua/5.1/LuaXML_lib.so: wrong ELF class: ELFCLASS32 Any suggestions gratefully received!

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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 87, Issue 203
by Labolinux | Sonerep 27 Sep '13

27 Sep '13

Hi Charles, Can I know where you are located? Some ISP Providers and especially in Africa, block the SIP port "5060" and I see in your log that all your accounts are connected to the port 5060. So, try to change the SIP Port 5060 to another new one like 5962 or other and Make a new call for seeing the issue. Awaiting your feedback, have a good day. Thanks, Sylvain Gotri Labolinux Le 26/09/13 21:11, freeswitch-users-request(a)lists.freeswitch.org a écrit : > Sujet: > [Freeswitch-users] Unable to transfer calls between 2 locations > De : > "Charles Bujold" <cjbujold(a)accra.ca> >

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Getting an [ERROR] fs_cli.c:1493 main() Error Connecting [Socket Connection Error] after raspberry pi installation
by Pranav Lal 26 Sep '13

26 Sep '13

Hi all, I have installed freeswitch on to my raspberry pi by following the instructions at http://tomoconnor.eu/blogish/freeswitch-raspberry-pi/#.UjVmkNS6a00 As far as I know the entire building from source process went flawlessly. I am trying to connect to my freeswitch instance by using fs_cli. When I run fs_cli, I get the following error. [ERROR] fs_cli.c:1493 main() Error Connecting [Socket Connection Error] I have tried searching the list archives. I found the following link but that did not help since I have not changed anything from the defaults. http://lists.freeswitch.org/pipermail/freeswitch-users/2011-February/069764. html I cannot understannd what to look for in the freeswitch log. I have uploaded the log to dropbox. https://dl.dropboxusercontent.com/u/3688386/fs.log The freeswitch daemon appears to be running is /etc/init.d/freeswitch status is to be believed. Pranav

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Problem when using alias in sip profile
by smrdoshi 26 Sep '13

26 Sep '13

Hi Guys, I am using mod_xml_curl module to handle directory and dialplan. Directory is loading properly using mod_xml_curl (means extensions are registering properly) and calls were also working fine without enabling alias. Now I need to use alias field as we are going to use domain name to register sip devices. But after enabling alias field my calls stops working. I debug log and found without alias field when we are making call its calling directory section first and then dialplan. But when we enable alias option its calling dialplan section directly in place of calling directory section first and because of that my code is not getting proper variables and calls are getting failed. Can anybody please help me to point out the issue? Maybe I am missing something in configuration. Please let me know. Thanks, Sam -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Problem-when-using-alias-in-s… Sent from the freeswitch-users mailing list archive at Nabble.com.

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mod_shout on Windows...
by Gerry Hull 26 Sep '13

26 Sep '13

I'm have trouble getting this module to load. I've tried both 32-bit and 64-bit versions on XP and Windows 7 64 respectively. I am getting 1261 or 1271 load errors when trying to load the module. I notice other people have asked this question in the past, but with no valid responses. Any ideas?? Gerry

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call out fail when using Freeswitch as sip server of polycom mcu
by tony 26 Sep '13

26 Sep '13

Anybody use freeswitch as sip server of polycom mcu (rmx1500) ? It is ok when call in from endpoint (hdx6000) , but fail when call out from polycom mcu to endpoint . when mcu call out to endpoint, ringing and ack is ok,but after ack ,endpoint invite mcu mcu(10.15.40.58) ---> FS(10.65.6.144) ----> endpoint (10.71.81.15) recv from tcp/[10.71.81.15]:5060 ------------------------------------------------------------------------ INVITE sip:mod_sofia@10.65.6.144:5060;transport=tcp SIP/2.0 From: <sip:1005@10.71.81.15:5060;transport=tcp>;tag=plcm_267848146-3395 To: "ALIPAY_MCU" <sip:test@10.65.6.144>;tag=r9UvSS7DQ51UF send to tcp/[10.15.40.58]:5060 ------------------------------------------------------------------------ INVITE sip:test@10.15.40.58:5060;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 10.65.6.144;branch=z9hG4bKSjHv7aU76y71H Max-Forwards: 69 From: <sip:1005@10.65.6.144>;tag=Dm347F7QQQcXF To: ALIPAY_MCU <sip:test@alibaba-inc.com:5060>;tag=rmx2k_4122954403-2363 recv from tcp/[10.15.40.58]:5060 ------------------------------------------------------------------------ SIP/2.0 481 Call Leg/Transaction Does Not Exist detail log file： mcu_call.rtf <http://freeswitch-users.2379917.n2.nabble.com/file/n7595224/mcu_call.rtf> Thanks -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/call-out-fail-when-using-Free… Sent from the freeswitch-users mailing list archive at Nabble.com.

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mod_callcenter question
by Ira Tessler 26 Sep '13

26 Sep '13

Does the calls_answered column in the agent table ever get reset to 0? If I am using a “agent-with-fewest-calls” strategy and say I add a new agent. Does that new agent get all the calls until the calls_answered counter catches up? Or if an agent logs out and then logs back into a queue say a day later, does he get all the calls until he catches up? Thanks, Ira Tessler Lead Software Engineer ConnectMe (732) 490-9007 x2 ira(a)connectmevoice.com

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