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ClueCon Weekly! May 14th Giovanni Maruzzelli, GSMOpen and FreeSWITCH
by Ken Rice 12 May '14

12 May '14

FreeSWITCHers, Join us Wed May 14th at 1P Easter (1700 GMT) when Giovanni Maruzzelli will join us for an update on GSMOpen! What is GSMOpen you ask? Just a nice little connector you allow you to use USB GSM Devices. Also, don¹t forget ClueCon 2014 is rapidly approaching! This will be marking 10 years of ClueCon! You don¹t want to miss this one! Visit Cluecon.com for more details! K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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Re: [Freeswitch-users] many fs<-->skype gateways stop working unexpectedly
by Yuriy Nasida 12 May '14

12 May '14

Hello again! Just would to let you know that jira ticket is created. https://jira.freeswitch.org/browse/FS-6512 I understand that for you the priority is minor but for us it's really not so. Skype gateways in production (we hadn't such problem in the past) and still having these very weird segmentation faults :( Any advice will be very appreciated. Thanks. From: nasida(a)live.ru To: freeswitch-users(a)lists.freeswitch.org Date: Thu, 8 May 2014 19:26:24 +0400 Subject: Re: [Freeswitch-users] many fs<-->skype gateways stop working unexpectedly Thanks for advice Giovanni but like I said in point 3 we already tried it and unfortunately it didn't help :( Okay i will try to do backtrace and to open Jira. From: gmaruzz(a)gmail.com Date: Thu, 8 May 2014 17:09:45 +0200 To: freeswitch-users(a)lists.freeswitch.org Subject: Re: [Freeswitch-users] many fs<-->skype gateways stop working unexpectedly Yuri, try this (maybe it will help): maybe the config directory of the skype clients has gone corrupted, the skype instances crashes, and somehow make fs to crash. Try to wipe or move the config directory of skype instances (where skype instances have their own databases, etc) and recreate it using the "install.pl" script. Maybe this solves the problem. If it does not solve, please open a Jira Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 On Thu, May 8, 2014 at 4:58 PM, Michael Jerris <mike(a)jerris.com> wrote: if freeswitch is segfaulting, then we would need to see the backtrace from that core dump. On May 8, 2014, at 10:47 AM, Yuriy Nasida <nasida(a)live.ru> wrote: Thanks for your response Giovanni. Yes are not Skype instances, but Xvfb instances and FS start working after several restartings. But now we are starting to get issues when FS went down regularly. It is strange because gateway worked fine long time ago. We already tried: 1) reboot full box a lot of times. fs works ~5 min (i can make a calls) and down again. there is nothing interesting in the fs logs. I only see that error in mesages May 8 10:16:11 SkypeGW1 kernel: freeswitch[6220] general protection rip:2b1848474c31 rsp:2b1858b9e400 error:0 2) tried to change skype id. we only have inbound traffic and some people said that microsoft can prevent correct work of gw in this case. so they advised to change skype id but that didn't help. 3) we have script which does full re-creating skype configuration of skype instances after restarting of box. This way we prevent issue with high PCU usage of skype instances. I believe you know that i talking about. 4) we have core dump and can open jira ticket if this is the correct way. Please advice Thanks. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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RTP Packet Size
by Markus von Arx 11 May '14

11 May '14

Hi I searched quite a lot but couldn't find a configuration parameter for the RTP packet size. I mean the value of the "a=ptime:20" line in SDP for G.711 PCMU and PCMA. Is there such a configuration value anywhere? Thanks, Markus

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Help Crash
by Andrew 10 May '14

10 May '14

Hi, Before I open a JIR I wanted to get some help and make sure I am reading this correct. I am using freeswitch-1.5.20140501.nightly and Mod managed ( I know everyone hates windows). When I hit 25 CPS the program crashes like clockwork. I also get a crash on SWITCH_DECLARE(const char *) API::executeString(const char *cmd) This line of code. switch_safe_free(last_data); I am using FreeSWITCH as imbedded.

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pizza demo is not working
by Vincent Xia 10 May '14

10 May '14

im testing mod_pocketsphinx with the pizza demo, i followed instructions from https://wiki.freeswitch.org/wiki/Mod_pocketsphinx and i also copied the latest wsj1 and communication model to the grammar folder. then after called 74992, i heard the voice prompt of "delivery or takeout" but fs seemed not responding to my speech, nothing changed at the fs console while i say "takeout" or "delivery", the pastebin is at http://pastebin.freeswitch.org/22513 so do i missed any configuration or grammar files? any hints are appreciated.

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ORIGINATOR_CANCEL when concurrent calls 175+
by Sam 09 May '14

09 May '14

Hi Guys, We need to handle 200+ concurrent calls but when concurrent call count is reaching to 175+ we are getting ORIGINATOR_CANCEL for most of the calls. We thought it might be an issue with max session, so we increased to 1000, sessions-per-second to 100, max-db-handles to 1000 and db-handle-timeout to 30 but no success. Can anybody please let me know what else can be the issue or what configuration needs to be changed to make it working correctly? Please help me. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/ORIGINATOR-CANCEL-when-concur… Sent from the freeswitch-users mailing list archive at Nabble.com.

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dial plan in mod xml curl
by raheem 09 May '14

09 May '14

Hi, I am trying to use mod xml curl with freeswitch. I have configured in "xml_curl.conf.xml " gateway url to where in at server it sends below response <document type="freeswitch/xml"> <section name="dialplan" description="RE Dial Plan For FreeSwitch"> <context name="public"> <extension name="unloop"> <condition field="${unroll_loops}" expression="^true$"/> <condition field="${sip_looped_call}" expression="^true$"> <action application="deflect" data="${destination_number}"/> </condition></extension></context></section> <section name="directory"> <domain name="123.176.38.68"> <params> </params> <users> <user id="1234" cacheable="true"> <params> </params> </user> <user id="4321" cacheable="true"> <params> </params> </user> </users> </domain> </section> </document> I am unable to make call for below error 2014-05-06 12:44:15.313984 [INFO] mod_dialplan_xml.c:558 Processing 1234 <1234>->4321 in context public 2014-05-06 12:44:15.633985 [INFO] switch_core_state_machine.c:192 No Route, Aborting I guess i am missing in dialplan part -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/dial-plan-in-mod-xml-curl-tp7… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Sangoma-FreeSWITCH QSIG support
by Siri MM 09 May '14

09 May '14

Hi, Could you please let me know if QSIG is supported with latest FreeSWITCH and Sangoma? I have seen the Sangoma Wiki which says that the feature is 'coming soon', but I am not sure if this is latest information. Would appreciate any inputs Thanks!

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Directory and ACL authentication
by Victor Chukalovskiy 09 May '14

09 May '14

Hello, Coming from wholesale background, my FS's run without any registrations. So far everything was ACL-based using "apply-inbound-acl" and I did not use any directory entries. The only problem with this is that once I have all IPs together in one big ALC, I can't identify which customer the call came from. E.g. need to set my_channel_variable=customer1 if a call came from particular IPs and my_channel_variable=customer2 if a call came from other IPs. So I'm trying to move ACL logic into directory by means of defining a user with cidr attribute. So far, no matter what I do FS challenges INVITE with "407" even-though the INVITE comes from the IP that is included in CIDR attribute for a user. I suppose for whatever reason switch does not match INVITEs against CIDR's in the directory. Please help me with that. WiKi is written from a somewhat different logic / perspective, so it's hard to apply. My SIP profile is: <profile name="test"> <gateways> </gateways> <domains> </domains> <settings> <param name="parse-invite-tel-params" value="true"/> <param name="user-agent-string" value="test"/> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="log-auth-failures" value="true"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="context" value="test"/> <param name="country" value="e164"/> <param name="dtmf-duration" value="2000"/> <param name="inbound-codec-prefs" value="$${default_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${default_codec_prefs}"/> <param name="caller-id-type" value="none"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="192.168.1.2"/> <param name="sip-ip" value="192.168.1.2"/> <param name="manage-presence" value="false"/> <param name="manage-shared-appearance" value="false"/> <param name="inbound-codec-negotiation" value="greedy"/> <param name="disable-transcoding" value="true"/> <param name="manual-redirect" value="false"/> <param name="disable-transfer" value="true"/> <param name="disable-register" value="false"/> <param name="auth-calls" value="true"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="pass-callee-id" value="false"/> </settings> </profile> Thanks! -Victor

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How to stop sending x-fs-supported header
by Victor Chukalovskiy 08 May '14

08 May '14

Good day, Would anyone know how to make FS stop sending x-fs-supported with the INVITEs? Someone's equipment is faulty (does not like the header), but needs to be fixed from the caller's side nevertheless. Thank you, Victor

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