May 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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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FS 2 NIC's
by Blake Priddy 08 May '14

08 May '14

Ok so here is my situation. This school that I set FS up on (trying to get AR schools to make the FS move) has a 6meg connection provided by the state for their internet. During peak internet usage times, calls become affected. I have an ISP installing another connection just for this box. My server has 2 NICs, and I want to send/receive outside calls through this internet connection. And the other port (current port) be on their local network for their internal stuff. What steps do I need to take to make this happen. IP Tables? or does FS have a magical config in vars to allow me to use the external profile on eth1. Thanks, -- *Blakelund Priddy* Network & Systems Engineer Bryant Public School District Bryant, Arkansas 72022 http://www.bryantschools.org p 501-653-5038 f 501-847-5656

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many fs<-->skype gateways stop working unexpectedly
by Yuriy Nasida 08 May '14

08 May '14

Hello guys, I have several fs<-->skype gateways and today all them shows that issue in FS CLI. 2014-05-05 02:59:38.188934 [WARNING] mod_skypopen.c:1759 [084e245|084e245 ] [WARNINGA 1759 ][skype101 ][IDLE,IDLE] STARTING interface_id=1 2014-05-05 02:59:38.189666 [ERR] skypopen_protocol.c:1686 [084e245|084e245 ] [ERRORA 1686 ][none ][N/A,N/A] Received error code 5 from X Server 2014-05-05 02:59:38.189696 [ERR] skypopen_protocol.c:1687 [084e245|084e245 ] [ERRORA 1687 ][none ][N/A,N/A] Display error for 1, :101.0 2014-05-05 02:59:38.189716 [ERR] skypopen_protocol.c:1884 [084e245|084e245 ] [ERRORA 1884 ][skype101 ][DEAD,IDLE] Fatal display error for :101 - successed to jump 2014-05-05 02:59:38.189747 [WARNING] skypopen_protocol.c:1910 [084e245|084e245 ] [WARNINGA 1910 ][skype101 ][DEAD,IDLE] Removing skype interface #skype101 2014-05-05 02:59:48.476751 [ERR] mod_skypopen.c:1801 [084e245|084e245 ] [ERRORA 1801 ][skype101 ][DEAD,IDLE] Failed to connect to a SKYPE API for interface_id=1, no SKYPE client running, please (re)start Skype client. Skypopen exiting 2014-05-05 02:59:49.478930 [CRIT] switch_loadable_module.c:1446 Error Loading module /usr/local/freeswitch/mod/mod_skypopen.so **Module load routine returned an error** Meanwhile I can see skype instanses started ps -aux | grep Xvfb root 7979 0.0 0.0 47928 7248 pts/3 S 02:40 0:00 /usr/bin/Xvfb :101 -ac -nolisten tcp -screen 0 640x480x8 root 7992 0.0 0.0 47928 7252 pts/3 S 02:40 0:00 /usr/bin/Xvfb :102 -ac -nolisten tcp -screen 0 640x480x8 root 8008 0.0 0.0 47928 7252 pts/3 S 02:40 0:00 /usr/bin/Xvfb :103 -ac -nolisten tcp -screen 0 640x480x8 root 8021 0.0 0.0 47928 7252 pts/3 S 02:41 0:00 /usr/bin/Xvfb :104 -ac -nolisten tcp -screen 0 640x480x8 root 8045 0.0 0.0 47928 7252 pts/3 S 02:41 0:00 /usr/bin/Xvfb :105 -ac -nolisten tcp -screen 0 640x480x8 Full restarting didn't help. Any advice will be very helpful! Thanks.

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