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FreeSWITCH Week in Review (Master Branch) February 14th-20th
by Ken Rice 23 Feb '15

23 Feb '15

New Post on freeswitch.org from Kathleen check it out at http://ift.tt/1Flzi7G FreeSWITCH Week in Review (Master Branch) February 14th-20th Hello, again. This passed week in the FreeSWITCH master branch we had 54 commits. The features for this week include: a simplification of command line arguments to allow for using -base instead of specifying each directory when using alternate configs, added support for URL attribute in the grammar tag for mod_rayo (this is useful for MRCP engines to look up their grammars directly), implement cookie-file option for mod_xml_cdr, allowing the removal of User-Agent header from the sip message, and the addition of multiple and reversed ranges for XML dialplan date and time conditions. Join us on Wednesdays at 12:00 CT for some more FreeSWITCH fun! And head over to freeswitch.com to learn more about FreeSWITCH support. New features that were added: FS-7284 A simplification of command line arguments to allow for using -base instead of specifying each directory when using alternate configs. FS-7285 Allow eavesdrop to bridge only one leg FS-7164 Added support for URL attribute in the grammar tag for mod_rayo. This is useful for MRCP engines to look up their grammars directly. FS-7299 Implement cookie-file option for mod_xml_cdr FS-7302 Added params to fs_encode.c:-c for path to conf_dir -k for path to log_dir -m for path to mod_dir FS-7309 Allow removal of User-Agent header from the sip message FS-7304 Multiple and reversed ranges for XML dialplan date and time conditions Improvements in cross platform build supports: FS-7122 Revert a previous patch until a fix can be found to allow for old configs as well Changes to packaging: FS-7318 Debian rules update to handle a pre-bootstrapped orig file The following bugs were squashed: FS-7294 Enable -Werror when building with clang compiler FS-7250 Removed the FreeSWITCH core handler for SIG_CHLD beacuse it isn’t necessary anymore and it causes dependant libraries that tried to start a child process to hang waiting on a signal that FreeSWITCH core intercepted. FS-7066 Fixed a bug causing higher cpu load averages on older kernels with related tickets FS-7253 FS-7231 FS-7296 Fix build error on newest gcc FS-7298 Fix race condition when callcenter member cancels the call. FS-7297 FS-7297 Update the install target in the makefile for mod_com_g729 FS-7301 FS-7301 Fix for issue faxing to numbers with a pass through tone FS-7192 FS-7192 Exclude Expires header in INVITEs responding to a auth challenge in mod_sofia FS-7308 FS-7308 Only log SLA SQL query SQL when debugging is enabled in mod_sofia FS-7306 FS-7306 Fix for fs_encode in mod_spandsp sleeping too much FS-7230 FS-7230 Fixed a memory leak from 1.4.14.

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Some strange thing in LUA
by Борисов, Дмитрий / Dmitriy Borisov 23 Feb '15

23 Feb '15

Hi, All! In log i found next: 2015-02-24 00:09:05.202353 [ERR] mod_lua.cpp:203 No matching function for overloaded 'new_Event' stack traceback: /usr/local/freeswitch/scripts/lua/debugger.lua:1447: in function </usr/local/freeswitch/scripts/lua/debugger.lua:1446> [C]: ? [C]: in function 'Event' /usr/local/freeswitch/scripts/lua/event_fire.lua:23: in main chunk line 23 in event_fire.lua: local event = freeswitch.Event("CUSTOM",event_subclass) I don`t overload new_Event. What`s up? -- with best regards, Dmitriy Borisov

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TLS/SRTP on selected destinations
by Victor Medina 23 Feb '15

23 Feb '15

Hi guys! I have configured my FS server to support TLS/SRTP... but I am facing the problem of providing the service only to selected destinations. Calls fails when calling to endpoints with no tls/srtp, for example a ext registered in the UDP port. Also fails when an outgoing call is routed to an external provider with no support. Using tls ONLY works just fine if connecting to external or udp only endpoints, it seems like FS is taking care of signalling from endpoint to the server and from there is goes as needed to the b-legs. When using SRTP however it fails. Can I configure FS to support TLS/SRTP to the server and from there using it as needed? For example: A_LEG: TLS/SRTP - > B_LEG: EXT with UDP only A_LEG: TLS/SRTP -> B_LEG: EXT with TLS/SRTP A_LEG: TLS/SRTP -> B_LEG: external channel, provider with no TLS/SRTP Thanks in advance with any help. -- Víctor E. Medina M. Software [image: Zoiper Click2Dial]+58424 291 4561 <zoiper://+584242914561>[image: ve] BB #79A8AFA2 /@VMCibersys

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Bypass media only if encryption is compatible?
by Thomas Auge 23 Feb '15

23 Feb '15

Hello List :) Is there a way to enable bypass media only when the two devices are compatible in terms of encryption? Chrome insists on DTLS, which many devices don't support yet, but it seems a waste of resources and latency to bridge when it's not necessary. Thanks, Thomas

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Monitoring peak performance
by Alex Lake 23 Feb '15

23 Feb '15

Just putting in some instrumentation to my Freeswitch hosts - wondering what the best way is to report back on how heavily loaded the system is. Things like how many call setups over the last x minutes, maximum number of simultaneously active channels... Does the freeswitch API expose any of these useful bits of information, or is that best done by CDR analysis or inline instrumentation code?

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Class not found Exception : mod_java
by Anurag Rana 23 Feb '15

23 Feb '15

Hi, I installed mod_java module. When I am trying to call it from within en extension it throws the error classNotFoundException : PhoneTest **************************************** 2015-02-19 05:16:04.421064 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1001(a)192.168.64.197! 2015-02-19 05:16:04.421064 [NOTICE] mod_dptools.c:1258 Channel [sofia/internal/1001(a)192.168.64.197] has been answered Exception in thread "Thread-2" java.lang.ClassNotFoundException: PhoneTest at java.net.URLClassLoader$1.run(URLClassLoader.java:366) at java.net.URLClassLoader$1.run(URLClassLoader.java:355) at java.security.AccessController.doPrivileged(Native Method) at java.net.URLClassLoader.findClass(URLClassLoader.java:354) at java.lang.ClassLoader.loadClass(ClassLoader.java:425) at java.lang.ClassLoader.loadClass(ClassLoader.java:358) at java.lang.Class.forName0(Native Method) at java.lang.Class.forName(Class.java:274) at org.freeswitch.Launcher.launch(Launcher.java:65) ************************************************************ *here is my dialplan - * ------------------------------- <extension name="java_demo"> <condition field="destination_number" expression="^(9005)$"> <action application="answer"/> <action application="java" data="/usr/local/freeswitch/scripts/java/jar1.jar PhoneTest"/> <action application="sleep" data="2000"/> </condition> </extension> ------------------------- Class file is placed exactly at the path mentioned in dialplan. Still it is complaining. Any point I might be missing here? Thanks Anurag Rana <https://sites.google.com/site/homepagerana/> M.Tech CSE, IIIT-Delhi, https://sites.google.com/site/homepagerana/

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hangup cause in v8
by Víctor Vladimir Chávez Gallardo 23 Feb '15

23 Feb '15

Hi i’m tring to catch the hangup cause for a outbound session, this session is started by a esl (socket connection) and an jsrun command, but when the end user give USER_BUSY or NO_ANSWER (or whatever) i can’t catch the disposition, currently the caller it’s working, (i can answer the calls, and do stuff) somebody can help me? EXAMPLE session = new Session( "{ignore_early_media=yes,origination_caller_id_number=55597428100",originate_timeout=20}freetdm/1/1/5513143104 ); hangup_cause = session.cause; <— I GOT FALSE if(session.cause == "USER_BUSY") { console_log("err", "Llamada fallida CAUSA: " + hangup_cause + "\n”); <— NEVER EXECUTED } if (session.ready() ) { session.streamFile( "barracuda.wav” ); <— WORKS WHEN USER ANSWER } else { console_log("err", session.uuid + "\n”); <—— THE SESSION IS “FALSE" var hangupCause = session.getVariable("bridge_hangup_cause”); <— THIS OBJECT IS UNDEFINED } session.hangup(); exit();

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Shared voicemail notification issue
by Iskren Hadzhinedev 23 Feb '15

23 Feb '15

Hello! I have two users in a group that share a mailbox. After numerous messages going in and out, when one of the phones isn't used, the message count for the two users doesn't match. Here's what I got from freeswitch.log: [DEBUG] mod_voicemail.c:2806 Deliver VM to 101@default [DEBUG] mod_voicemail.c:1923 Update MWI: Processing for 101@default in inbox [DEBUG] mod_voicemail.c:1946 Update MWI: Messages Waiting yes [DEBUG] mod_voicemail.c:1947 Update MWI: Update Reason NEW [DEBUG] mod_voicemail.c:1948 Update MWI: Message Account 101@default [DEBUG] mod_voicemail.c:1949 Update MWI: Voice Message 1/0 [DEBUG] mod_voicemail.c:2806 Deliver VM to 100@default [DEBUG] mod_voicemail.c:1923 Update MWI: Processing for 100@default in inbox [DEBUG] mod_voicemail.c:1946 Update MWI: Messages Waiting yes [DEBUG] mod_voicemail.c:1947 Update MWI: Update Reason NEW [DEBUG] mod_voicemail.c:1948 Update MWI: Message Account 100@default [DEBUG] mod_voicemail.c:1949 Update MWI: Voice Message 17/0 and the user configuration, along with the dialplan extension: <user id="100" mailbox="100"> <params> <param name="vm-enabled" value="true"/> <param name="password" value="1234"/> <param name="vm-password" value="1234"/> </params> <variables> <variable name="user_context" value="default"/> <variable name="destinations" value="internal,local,national,international"/> </variables> </user> <user id="101" mailbox="100"> <params> <param name="vm-enabled" value="true"/> <param name="vm-password" value="1234"/> <param name="password" value="1234"/> </params> <variables> <variable name="user_context" value="default"/> <variable name="destinations" value="internal,local,national,international"/> </variables> </user> <extension continue="False" name="Ring or leave a message"> <condition field="${destination_number}" expression="^my-number$" break="on-false"> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="call_timeout=15"/> <action application="set" data="vm_cc=100@{domain_name}"/> <action application="set" data="skip_instructions=true"/> <action application="bridge" data="${group_call(1000@${domain_name})}"/> <action application="answer"/> <action application="sleep" data="500"/> <action application="voicemail" data="default ${domain_name} 101"/> </condition> </extension> First time user 100 logged into the mailbox got a "17 new messages" prompt but after hanging up and redialling, he heard "0 new messages" and the log confirms that he hadn't delete any messages the first time. Is this a matter of misconfiguration on the shared mailbox or there's something else I'm missing? Thanks! Kind regards, -- Iskren Hadzhinedev

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AUDIO RTP REPORTS ERROR: [Bind Error! on Multipple profile
by Steve Kieu 23 Feb '15

23 Feb '15

Hi everyone, My setup is that I have * internal.xml on 192.168.0.2 * vpn.xml on 10.8.0.1 * external.xml Freeswitch behind NAT router ; wlan0 has 192.168.0.2 and 10.8.0.1 is tun0 which is the openvpn internface. Default gateway is wlan0. Thing like normal using internal.xml But for call from A registered in vpn profile I always got 2015-02-20 17:41:44.591664 [ERR] sofia_glue.c:3989 AUDIO RTP REPORTS ERROR: [Bind Error! XXX:23324] XXX is my external internet IP. The vpn.xml I set: <param name="rtp-ip" value="$${external_rtp_ip}"/>  <param name="sip-ip" value="10.8.0.1"/> <param name="ext-sip-ip" value="auto-nat"/> <param name="ext-rtp-ip" value="auto-nat"/> no matter what I set in ext-rtp-ip (did replace auto-nat with my external IP) it still gives the errors. I want for the vpn profile, just use the vpn IP for sip for the endpoint register with this, but using correct sip rtp when sending out to endpoint registered in other profile - basically making call between profile working - Can freeswitch support that and how ? I did set all IP to use vpn address only, and call making between same profile (vpn) works. But not to other profile and not to external gateway. Same problem. Thanks a lot in advance. -- Steve Kieu

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Limit max concurrent calls on gateways in lua
by Zoltán Szabó 22 Feb '15

22 Feb '15

Hi, I have a lua script where I do some business logic. After the logic I call a number on a gateway, then when the called party picked up, I call an other number on a gateway and bridge the two calls. The only problem is that I have 5 gateways and each can have maximum of 3 calls (so I can have 15 calls on this system). Here is what I have at the moment (simplified version): dial_call_1 = "{leg=1}sofia/gateway/gw-1/0035312345678"; dial_call_2 = "{leg=2}sofia/gateway/gw-2/0035387654321"; session1 = freeswitch.Session(dial_call_1); if (session1:ready()) then session2 = freeswitch.Session(dial_call_2, session1); if (session2:ready()) then freeswitch.bridge(session1, session2); if (session2:ready()) then session2:hangup(); end end if (session1:ready()) then session1:hangup(); end end How can I use the limit application here to set the max concurrent calls on my five gateway, each with 3 max calls and distribute outbound calls between them? Many thanks, Zoltan

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