February 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

ClueCon Weekly! Feb 26th Join us!
by Ken Rice 26 Feb '14

26 Feb '14

Hey FreeSWITCHers, Join us for ClueCon Weekly today at 1PM EST (10AM PST) at sip:888@conference.freeswitch.org Todays Topic with be FreeSWITCH Infrastructure Changes Update and Following up on BSDs plus Jira! Anthony, Brian and MikeJ along with several others have been busting hump this week. Join us to find out been going on here! Ken -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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http://jira.freeswitch.org/browse/FS-6227
by Brian West 26 Feb '14

26 Feb '14

FreeSWITCHers, Here is another example of a platform we’ve supported in the past thats shriveled up an died on the vine. Anyone interested in helping reviving support for Solaris/SmartOS? -- Brian West brian(a)freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)

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NORMAL_UNSPECIFIED Disconnect
by Brian Fertig 26 Feb '14

26 Feb '14

List, I am having an interesting issue. I have active calls disconnecting at different points and it appears on the ladder graph that FS is sending BYE's to both legs of the call. I have looked at the log and cant make heads or tails as to why. I have included a snippet of the log in this email maybe someone can tell me whats going on. TIA! 2014-02-25 11:00:04.816713 [DEBUG] mod_sofia.c:87 sofia/internal/517NXXXXXX(a)216.238.XXX.XXX SOFIA INIT 2014-02-25 11:00:04.816713 [DEBUG] sofia_glue.c:2744 sofia/internal/517NXXXXXX(a)216.238.XXX.XXX sending invite version: 1.2.22 git f7d5eba 2014-02-18 21:16:05Z 64bit Local SDP: v=0 o=FreeSWITCH 1393320114 1393320115 IN IP4 144.202.XXX.XXX s=FreeSWITCH c=IN IP4 144.202.XXX.XXX t=0 0 m=audio 23890 RTP/AVP 18 0 101 13 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv 2014-02-25 11:00:04.836881 [DEBUG] switch_core_session.c:1016 Send signal sofia/internal/517NXXXXXX(a)216.238.XXX.XXX [BREAK] 2014-02-25 11:00:04.836881 [DEBUG] mod_sofia.c:127 (sofia/internal/517NXXXXXX(a)216.238.XXX.XXX) State Change CS_INIT -> CS_ROUTING 2014-02-25 11:00:04.836881 [DEBUG] switch_core_session.c:1351 Send signal sofia/internal/517NXXXXXX(a)216.238.XXX.XXX [BREAK] 2014-02-25 11:00:04.836881 [DEBUG] switch_core_state_machine.c:455 (sofia/internal/517NXXXXXX(a)216.238.XXX.XXX) State INIT going to sleep 2014-02-25 11:00:04.836881 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/517NXXXXXX(a)216.238.XXX.XXX) Running State Change CS_ROUTING 2014-02-25 11:00:04.836881 [DEBUG] sofia.c:5811 Channel sofia/internal/517NXXXXXX(a)216.238.XXX.XXX entering state [calling][0] 2014-02-25 11:00:04.836881 [DEBUG] switch_core_state_machine.c:471 (sofia/internal/517NXXXXXX(a)216.238.XXX.XXX) State ROUTING 2014-02-25 11:00:04.836881 [DEBUG] mod_sofia.c:150 sofia/internal/517NXXXXXX(a)216.238.XXX.XXX SOFIA ROUTING 2014-02-25 11:00:04.836881 [DEBUG] switch_ivr_originate.c:67 (sofia/internal/517NXXXXXX(a)216.238.XXX.XXX) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2014-02-25 11:00:04.836881 [DEBUG] switch_core_session.c:1351 Send signal sofia/internal/517NXXXXXX(a)216.238.XXX.XXX [BREAK] 2014-02-25 11:00:04.836881 [DEBUG] switch_core_state_machine.c:471 (sofia/internal/517NXXXXXX(a)216.238.XXX.XXX) State ROUTING going to sleep 2014-02-25 11:00:04.836881 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/517NXXXXXX(a)216.238.XXX.XXX) Running State Change CS_CONSUME_MEDIA 2014-02-25 11:00:04.836881 [DEBUG] switch_core_state_machine.c:490 (sofia/internal/517NXXXXXX(a)216.238.XXX.XXX) State CONSUME_MEDIA 2014-02-25 11:00:04.836881 [DEBUG] switch_core_state_machine.c:490 (sofia/internal/517NXXXXXX(a)216.238.XXX.XXX) State CONSUME_MEDIA going to sleep 2014-02-25 11:00:05.716715 [DEBUG] switch_core_session.c:1016 Send signal sofia/internal/888NXXXXXX(a)10.100.0.59 [BREAK] 2014-02-25 11:00:05.716715 [DEBUG] switch_core_session.c:1016 Send signal sofia/internal/888NXXXXXX(a)10.100.0.59 [BREAK] 2014-02-25 11:00:05.716715 [DEBUG] switch_core_session.c:1016 Send signal sofia/internal/888NXXXXXX(a)10.100.0.59 [BREAK] 2014-02-25 11:00:05.716715 [DEBUG] switch_core_session.c:1016 Send signal sofia/internal/888NXXXXXX(a)10.100.0.59 [BREAK] 2014-02-25 11:00:05.716715 [DEBUG] switch_core_session.c:1016 Send signal sofia/internal/888NXXXXXX(a)10.100.0.59 [BREAK] 2014-02-25 11:00:05.736724 [DEBUG] sofia.c:5811 Channel sofia/internal/888NXXXXXX(a)10.100.0.59 entering state [terminating][0] 2014-02-25 11:00:05.736724 [NOTICE] sofia.c:6655 Hangup sofia/internal/888NXXXXXX(a)10.100.0.59 [CS_EXECUTE] [NORMAL_UNSPECIFIED] 2014-02-25 11:00:05.736724 [DEBUG] switch_channel.c:3184 Send signal sofia/internal/888NXXXXXX(a)10.100.0.59 [KILL]

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NetBSD Status
by Brian West 26 Feb '14

26 Feb '14

FreeSWITCHers, I’ve taken NetBSD as far as I can without some assistance from someone that is more familiar with NetBSD and various build issues that you may encounter. Here are some details if someone wishes to take this over the finish line: Configure detects the SWITCH_SIZE_T_FMT as “d”, It should be “zd” If you manually cane src/include/switch_am_config.h after configure it will compile. Here are a few of the patches that I’ve tweaked so far not sure if they are all that correct: http://www.bkw.org/netbsd.diff The presence of AC_FUNC_MALLOC in configure.in breaks malloc on NetBSD and rpl_malloc isn’t defined. NetBSD also suffers the same as a few other other BSD’s with a broken tgmath.h, thats the change to src/mod/applications/mod_spandsp/mod_spandsp_modem.h Also —disable-srtp will need to be used until the rest of the issues are resolved properly. Commit 378caebc9a1b67aa16ac69fc6b48452078ae19f1 solves the issue with disabling srtp. Thanks, -- Brian West brian(a)freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)

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group call shared line
by Jason Holden 26 Feb '14

26 Feb '14

All, I have shared line working properly when dialing an extension directly, even if there are multiple registered extensions. The remaining problem is when group calling shared line does not functioning. Any recommendations to address this? Internal profile <param name="manage-presence" value="true"/> <param name="manage-shared-appearance" value="true"/> <param name="presence-hosts" value="$${domain},$${internal_sip_ip},$${external_sip_ip}"/> <param name="force-register-domain" value="$${domain}"/> <param name="force-register-db-domain" value="$${domain}"/>

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Brainstorming load balancing feature - dispatcher & FS
by jay binks 26 Feb '14

26 Feb '14

So I want to bash out a feature I was thinking of, please can anyone chime in some ideas on this. So its easy to have OpenSIP / Kamailio ( dispatcher ) send SIP Options to FS to see if its responding .. this is great ... My request would be to have some kind of control over the response FS Sends back. What I would love is to respond with idle CPU , current SPS / Max SPS or channels / Max Channels. This is only a high level "this would be nice" , so I havnt totally thought this through but... we would have to provide an ACL for who to provide this back to, coz I wouldnt really want any old guy on the internet sending me options and me respond going ... oh yea bro, Im at 90% ... dont have to try hard to DDOS Me.. :) but yea... if we could give this back to the load balancers, then we could easily instruct them to back traffic off to this gateway. the next part is , how do you back off the traffic. one option is to have FS respond differently if any of these are close to thresholds. or to provide these values in custom headers, but that would require changes in dispatcher ( which is ok I Guess ) anyways... I wanted some input on peoples thoughts here.. let me know what you think. Jay

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Segmentation fault while starting freeswitch.
by vinay saini 26 Feb '14

26 Feb '14

Hi, I have installed freeswitch in centos 6.4 from the rpm , and the freeswitch version is freeswitch-1.4.2-0 but i face problem when starting it. It is giving Segmentation fault while staring Starting freeswitch: chown: cannot access `/var/log/freeswitch/*': No such > file or directory > /etc/init.d/functions: line 181: 20622 Segmentation fault (core > dumped) $cgroup $nice runuser -s /bin/bash $user -c "$corelimit >/dev/null > 2>&1 ; $*" > [FAILED] > > can any one help me to resolve this problem. thanks, vinay

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Re: [Freeswitch-users] performance
by Keith 26 Feb '14

26 Feb '14

Hi Brian, Thanks for your reply, not sure I follow what you mean by git bisect?? Sorry I am a bit new to a lot of this? Are you saying to compare the GIT versions? Cheers Keith

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Re: [Freeswitch-users] Performance
by Keith 26 Feb '14

26 Feb '14

Well I did have FS bridging to an asterisk box behind it, but wanted to check it wasn't the FS box which was the problem to did a simple dial plan with some music playing back. The interesting thing is, I used one of the latest GIT versions and managed to get 1000 calls, but I would expect with the hardware i have to get even more. I would rather use a stable version of the software, it's just bizare that newer hardware with much higher spec doesn't perform as well. Thanks, Keith

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Call Hanging up on receiving 302 from SIP endpoint
by Akhil Chandra 25 Feb '14

25 Feb '14

Hello everyone, Looks like I have a strange problem at hand. A SIP user/ extension has forwarded his phone to his mobile number. 1. When another user( ie extension) calls the SIP user, the call is forwarded to forwarded number after getting 302 redirect. 2. However, if I call from PSTN into the SIP user, the call hangs up after getting the 302 message. Not sure whats going on. I have the <param name="manual-redirect" value="true"/> in the SIP profile for the extension. NB: I have 2 profiles, one for extensions and one for SIP trunk. Just wanted to separate the ports. Nothing else. But the SIP profile for trunk does not have the manual-redirect set to true. Regards, Akhil

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