August 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Latest master and latest v1.4 consume more CPU for conferences
by Oleg Stolyar 01 Aug '14

01 Aug '14

Hi guys, I have my FreeSWITCH running a bunch of conferences with two or three users using PCMU/8000 from all the user agents. I have the conference sample rate set to 12000 to get around a strange bug I reported a couple of months ago. In my old installation with master from May 22 2014, the servers consumed about 1.2%-1.3% CPU per two-person conference. In the latest master AND in the latest v1.4 it went up to about 1.6%-2.0% CPU per two-person conference. CPU usage by the FreeSWITCH process also jumps up and down more. Additionally (not sure if it's related) the latest master crashed less than 24 hours after it started taking live traffic. I ma trying to get the core dump and see if v1.4 also crashes but including the info here in case it gives someone an idea of what's going on with the CPU. Any ideas as to what's causing the increased CPU usage would be greatly appreciated.

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FreeSWITCH Developers Dinner!
by Ken Rice 01 Aug '14

01 Aug '14

Hey FreeSWITCHers: Next week while everyone is in Chicago at ClueCon, Help treat the FreeSWITCH Dev Team to dinner! Visit FreeSWITCH.org and hit the donate button and in the notes let them know its for Dinner! Anthony Mike and Brian and the whole crew work many hours to make FreeSWITCH what it is so lets send them a thank you! Also, check out how much work they have been doing! We¹ve added graphing to https://stash.freeswitch.org so you can follow how often and how much everyone is committing! K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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FreeSWITCH Friday BugHunt Reminder!
by Ken Rice 01 Aug '14

01 Aug '14

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday BugHunt Call sip:888@conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken http://ift.tt/vGSsGm http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon

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How to get conference call status?
by jobin dcruz 01 Aug '14

01 Aug '14

Hi All, I initiate call conference from this command "conference conference_name bgdial {originate_timeout=60}user/1000". But how to get live call status? if a call is answered,how to get call status and triggered to a url from lua script?. How to find conference member id from uuid?. Thanks -- Jobin D'cruz (LAMP Developer) Email: jobindcruz(a)gmail.com | lampdeveloper.jobindcruz(a)gmail.com Php MySql Linux Apache Ajax FFmpeg FFmpeg-php Mplayer Zend Framework Codeigniter CakePhp Wordpress Drupal Joomla Html5 Css3 jQuery YUI MooTools Smarty Firefox Extension

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SIP Gateway registration fails on IPv4
by Micha Lehmann 01 Aug '14

01 Aug '14

Hi, I have a fresh FS install from source (Stash) on a root server with (temporarily) disabled firewall. There I added an external sip profile in conf/sip_profiles/external/sipgate.xml with the following content: <include> <gateway name="sipgate.de"> <param name="proxy" value="sipgate.de"/> <param name="username" value="215****"/> <param name="password" value="TOPSECRET"/> <param name="extension" value="1000"/> </gateway> <param name="rfc-5626" value="true"/> <param name="reg-id" value="1"/> </include> If I start freeswitch and type 'sofia status gateway' I get the following result: Profile::Gateway-Name Data State IB Calls(F/T) OB Calls(F/T) ================================================================================================= external-ipv6::sipgate.de sip:215****@sipgate.de REGED 0/0 0/0 external::sipgate.de sip:215****@sipgate.de TRYING 0/0 0/0 (retry: 45s) ================================================================================================= 2 gateways: Inbound(Failed/Total): 0/0,Outbound(Failed/Total):0/0 The second sipgate account does not register with error 908 (timeout) and sometimes with error 503 dns error. Can anyone help me to fix this? Thanks M. Lehmann

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Codec Negotiation
by Stephen Wilde 01 Aug '14

01 Aug '14

I have a "no audio" issue with FreeSWITCH with a specific sip supplier, no issue with all others. What I see is a strange codec negotiation: FreeSWITCH sends the SDP in the INVITE with: m=audio 22026 RTP/AVP 18 8 101 The Supplier sends its SDP in the 200 OK with: m=audio 34020 RTP/AVP 8 0 18 4 101 After that, FreeSWITCH uses the G729 and the supplier uses the PCMA so something is wrong in the codec negotiation. Who is in error FreeSWITCH or the supplier and why? Thanks in advance. Stephen

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mod_blacklist
by Terry Barnum 01 Aug '14

01 Aug '14

I started playing around this weekend with blacklisting certain telemarketers that seem to call like clockwork. I got it working using <https://confluence.freeswitch.org/display/FREESWITCH/mod_blacklist> as a starting point. For now I'm just using the console to add numbers, not a dialplan extension. If I have two numbers in the blocked list and delete one of them using the console, then save the list, it still saves both numbers to the blocked file. I have to manually edit blocked.list to remove the number. FreeSWITCH Version 1.4.6+git~20140603T193516Z~9479729340~64bit (git 9479729 2014-06-03 19:35:16Z 64bit) on a 2.3GHz i7 MacMini. From the console: freeswitch(a)mini.local> blacklist add blocked 1234567890 +OK 2014-07-28 18:09:26.556327 [INFO] mod_blacklist.c:228 Added [1234567890] to list [blocked] freeswitch(a)mini.local> blacklist check blocked 1234567890 true freeswitch(a)mini.local> blacklist del blocked 1234567890 +OK 2014-07-28 18:09:50.776324 [INFO] mod_blacklist.c:250 Removed [1234567890] from list [blocked] freeswitch(a)mini.local> blacklist check blocked 1234567890 false freeswitch(a)mini.local> blacklist save blocked 2014-07-28 18:10:11.056373 [INFO] mod_blacklist.c:280 Saving blocked to /usr/local/freeswitch/conf/blacklists/blocked.list 2014-07-28 18:10:11.056373 [DEBUG] mod_blacklist.c:288 adding 15129160420 to the dump file +OK 2014-07-28 18:10:11.056373 [DEBUG] mod_blacklist.c:288 adding 1234567890 to the dump file The last line shows fs saving the deleted number to the blocked list. Are the docs incorrect? Is there another way to do this? Thanks, -Terry

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Server having two Sangoma cards and can't handle full capacity
by Sambit Jaysingh 01 Aug '14

01 Aug '14

We are having two (Sangoma) Voice Cards, one with 8-ports and other with 16-ports. I have integrated ESL library with java in INBOUND MODE and making outbound calls. The issue what we are facing are 1. When I use all 720 channels or more than 450 channels, we see the events processed by ESL is very slow and we are not getting the events properly from ESL after placing the calls, even after so many minutes, so we have to forcefully close the connection. 2. And I see error mod_event_socket.c:375 Lost 502 events! And the java app also crashes with error ESL connection timeout. 3. And I see segmentation_fault in Freeswitch and big dump file is written and which would crash the system, because of high CPU\IO. But if we reduce the channels say, 450 we don't see any issue at all. What could be the reason for this issue. Appreciate any Help... -- Regards Sambit +91-9916211211

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