August 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

error loading module 'luasql.mysql'
by Adam Wilt 02 Sep '10

02 Sep '10

Hi, I followed the instructions in the Lua documentation for setting up luasql, but when I try to run my script I get: 2010-01-27 19:08:14.799250 [ERR] mod_lua.cpp:182 error loading module 'luasql.mysql' from file '/usr/local/lib/lua/5.1/luasql/mysql.so': /usr/local/lib/lua/5.1/luasql/mysql.so: cannot restore segment prot after reloc: Permission denied stack traceback: [C]: ? [C]: in function 'require' /usr/local/freeswitch/scripts/l.lua:2: in main chunk I'm running FreeSWITCH version 1.4 and luasql version 2.1.1. I changed the permissions for mysql.so and for my script to 777, so I'm not sure where the permission problem could be. I'd appreciate any suggestions. Thanks, Adam

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not able to connect freeswitch with mysql
by ovvenkat 02 Sep '10

02 Sep '10

Hi to all, I have following this wiki page to configure mysql and freeswitch. http://wiki.freeswitch.org/wiki/Mod_spidermonkey_odbc When I run the command " *isql -v maxpowersoft_odbc* ", I can able to connect to the MySql database. When , I run the following java script file, *test.js* use("ODBC"); var db = new ODBC("maxpowersoft_odbc", "dbuser", "dbpass"); db.connect(); I am getting error like , 2010-08-31 21:24:06.583606 [ERR] switch_odbc.c:318 STATE: IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data source name not found, and no default driver specified Any one help me to relove my issue plz. thanks in advance. Regards, Venkat.

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incoming calls - newbie question
by Adolfo Delorenzo 01 Sep '10

01 Sep '10

Hello, I am not receiving incoming calls via enum (my ITAD is 1325) and if someone calls for example myextension@mydomain. Where in the documentation can I learn how to enable incoming calls? Thanks

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Early Media troubles
by Jock McKechnie 01 Sep '10

01 Sep '10

Good arvo, all; I'm battling with a problem in making Early Media play and getting nowhere. I've searched the wiki, forums and the web in general without anything that seems to help. This suggests to me that I'm either doing something stupendously stupid, or there's a very, very odd thing going on here. I've been tasked with something that should be fairly simple: Accept an inbound call Play a specific audio file as early media Pickup the call after a certain time interval Play another audio file into the off-hook channel Disconnect the call. Sounds simple. My extension looks like so: <extension name="Inbound"> <condition field="destination_number" expression="^\+12125551212$"> <action application="pre_answer"/> <action application="set" data="ringback=/usr/local/freeswitch/sounds/early-media.wav"/>  <action application="sleep" data="10000"/> <action application="answer"/> <action application="playback" data="/usr/local/freeswitch/sounds/playback"/> <action application="hangup" data="16"/> </condition> </extension> (Replace the 212 number with a real inbound DID which is being routed from Verizon (via SIP) to this FreeSWITCH machine) The call is received, a 183 is sent by FreeSWITCH (pre_answer), it waits the 10 seconds, picks up the call and plays the playback.PCMU file... but there's no early media. I've tcpdumped on the FS machine and verified it definitely is /not/ sending out RTP until the call 200 OKs and the Playback begins. As you can see, I've also attempted to replace the ringback= to a cadence instead of a WAV file, just to ensure it isn't unhappy with the WAV format, with the same results. Boy, I'd sure love it if someone could point out what potentially brain damaged thing I'm doing (assuming it's nothing arcane). My thanks to all; - Jock

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no ringback tone for transit h323 - sip call
by Nikolay Kondratyev 01 Sep '10

01 Sep '10

Hi all, I'm rather new to freeswitch, and i'm trying to setup FS (version 1.0.6) as sip-h323 gateway between sipX and avaya ip406. I have teh following test setup: sipX 172.23.9.2 beaver.sip.nstel.ru with extension 2853 (linksys spa 942 phone) Freeswitch 172.23.22.49 fs.lab.nstel.ru with extension 2854 (xlite) avaya ip406 with extension 5840 When i call 2853 -> 5840 i can hear ringback tone and voice is ok. When calling 5840 -> 2854, rbt and voice is ok. When calling 2854 -> 2853, rbt and voice is ok. When calling 5840 -> 2853, 2853 phone rings, there is no rbt, but voice is ok after answer. I have the following h323.conf file: [freeswitch@freeswitch autoload_configs]$ cat h323.conf.xml <configuration name="h323.conf" description="Opal(h323) Endpoints"> <settings> <param name="trace-level" value="4"/> <param name="context" value="default"/> <param name="dialplan" value="XML"/> <param name="codec-prefs" value="PCMA"/> <param name="gk-address" value=""/>  <param name="gk-identifer" value=""/>  <param name="gk-interface" value=""/>  </settings> <listeners> <listener name="h323-listener"> <param name="h323-ip" value="$${local_ip_v4}"/> <param name="h323-port" value="1720"/> </listener> </listeners> </configuration> in the default.xml dialplan i have default "Local_Extension" (with the only addition of a single number 2854 to be matched). below Local_Extension i send 28xx numbers to sipx as following: <extension name="beaver"> <condition field="destination_number" expression="^(28[0-9]{2})$"> <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/> <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/> <action application="bridge" data="sofia/external/$1@beaver.sip.nstel.ru:5080"/ <mailto:sofia/external/$1@beaver.sip.nstel.ru:5080> > </condition> </extension> FS logs for 5840 -> 2854 (rbt and voice ok) and 5840 -> 2853 (no rbt, voice ok) are attached. In the log files i can see the moment when FS starts playing RBT into h323 channel in case 5840 -> 2854 call. And i see that FS does not do it in case of 5840 -> 2853 call. But i can not see why... Probably i misconfigured something... Can anybody please help with this problem? Thanks in advance, Nikolay.

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Nextone (Genband) >> Freeswitch Ringback issue
by Errol Samuels 01 Sep '10

01 Sep '10

Hello All, I have a weird situation where a carrier is sending calls to our FS server but not hearing a Ringback tone, however once the call connects they are getting audio but when this carrier sends calls to our Yate server they are getting the Ringback without a problem. I have tried setting the Ring_Ready below but it didn't help the situation either. <extension name="ringback" continue="true"> <condition field="network_addr" expression="^(.*)$"> <action application="ring_ready"/> </condition> </extension> The traces from both the FS server and the Yate server show we are sending: 100 Trying 180 Ringing 183 Session Progress with SDP The carrier is claiming they are using the same configuration for both the Yate and FS server. Has anyone had similar a similar issue with Nextone (Genband)? Just to mention we are sending calls to this carrier from FS without issue. Thanks in advance, Errol

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Skype 2.0.72 or 2.1.0.81
by Shamun toha md 31 Aug '10

31 Aug '10

Hi, Which skype to use? Isn't that 2.0.72 version ancient skype? IF 2.0.72 to be used, where can we download this for Fedora/CentOS? Any idea guys!!! Which Skype Client to use on Linux Use the *static* build of the *stable* Skype client (2.0.72). *Don't* use the build for your distro, neither the 'dynamic build'. *Don't* use the *beta* Skype client or more recent "stable" (2.0.72 is the one you want to use) if you want to have multiple channels with the same skypename.

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windows build error
by Seven Du 31 Aug '10

31 Aug '10

Hi, I'm new to windows build. I cloned code a few days ago and pulled to head. Can could not build sofia. c:\workspace\freeswitch\src\mod\endpoints\mod_sofia\mod_sofia.h(119) : fatal error C1083: Cannot open include file: 'sofia-sip/sip_extra.h': No such file or directory I tried to rebuild but the result was the same. I can start freeswitch though mod_sofia and mod file format (not exactly remember) cannot load. Any hint? I will re-clone code and try again tomorrow. Thanks. -- Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn

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mod_managed API interface question
by Malay Thakershi 31 Aug '10

31 Aug '10

Hello, Please let me know if this is feasible / possible. I need to make outgoing calls based on a scheduler using FS (SIP). I am planning to write a windows service that runs at particular time interval (say 10 minutes), checks who needs to be called, fires commands to originate these calls. I have a single FS command line running (where FS console is running). I am planning to fire FS commands such as: ---------------------- >managed apidemo a1 123 API CALL [managed(apidemo a1 123)] output: ApiDemo executed with args 'a1 123' and event type API. >managedrun apidemo a1 123 API CALL [managedrun(apidemo a1 123)] output: +OK 2008-10-08 17:44:25 [DEBUG] switch_cpp.cpp:1059 console_log() ApiDemo on a background thread #(3), with args 'a1 123'. 2008-10-08 17:44:25 [DEBUG] switch_cpp.cpp:1059 console_log() ExecuteBackground in apidemo completed. ---------------------- So I will pass some arguments to the mod_managed script and it will fire an API to originate call. My main question is: How can I communicate windows service to fire FreeSwitch commands? Thank you for any help or guidance.

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How can I keep the A-leg from establishing early media (and playing ringback in-band) until after the B-leg answers the call?
by Jason Jeffords 31 Aug '10

31 Aug '10

My specific use case is an inbound Skype call using mod_skypopen to a SIP phone. I would like the Skype call to keep ringing as a Skype call (not a PSTN call) until the B-leg is answered. The default behaviour is the Skype call is answered by freeswitch and ringback is played over the established Skype call. Thanks in advance, Jason

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