September 2013 - freeswitch-dev@lists.freeswitch.org

[Freeswitch-dev] Automatically don't respond to abnormal registrations
by Sayyed Mohammad Emami Razavi 04 Oct '13

04 Oct '13

Can anyone describe a way in FS to get rid of crawlers and hackers testing your PBX second by second on the internet?! What is the problem? This is ls of my FS's sqlite db after some annoying behaviours of hackers and crawlers violently done: [root@freeswitch db]# ll total 705580 -rw-r--r-- 1 root root 7168 Sep 29 13:20 callcenter.db -rw-r--r-- 1 root root 14336 Sep 23 10:29 call_limit.db -rw-r--r-- 1 root root 257024 Sep 29 13:56 core.db -rw-r--r-- 1 root root 5120 Sep 29 13:20 fifo.db -rw-r--r-- 1 root root 98304 Sep 24 11:34 sofia_reg_sipinterface_2.db -rw-r--r-- 1 root root 367616 Sep 25 18:31 sofia_reg_sipinterface_3.db -rw-r--r-- 1 root root 500726784 Sep 29 13:56 sofia_reg_sipinterface_5.db -rw-r--r-- 1 root root 221003312 Sep 29 13:56 sofia_reg_sipinterface_5.db-journal -rw-r--r-- 1 root root 16384 Sep 23 10:29 voicemail_default.db [root@freeswitch db]# sofia_reg_sipinterface_5.db belongs to the interface listening on 5060 port. As you see sofia_reg_sipinterface_5.db exceed 500MB of size! and this db has crashed! and leads to drop port 5060 of interfaces. when i ask fs_cli about profiles it tells me: +OK log level [7] freeswitch@internal> sofia status Name Type Data State ================================================================================================= sipinterface_3 profile sip:mod_sofia@192.168.2.73:5080 RUNNING (0) voicemail_1 alias sipinterface_2 ALIASED voicemail_2 alias sipinterface_2 ALIASED sipinterface_2 profile sip:mod_sofia@192.168.2.73:5070 RUNNING (0) ================================================================================================= 2 profiles 2 aliases Which open source firewalls can handle automatically weird sip requests to ban?! Sincerely yours,

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[Freeswitch-dev] Source does not hangup
by Jason Sia 30 Sep '13

30 Sep '13

Hi, I'm using tcp, when the source calls the destination and the destination answers and then hangs up the call, the source does not hangup. Reversing the scenario wherein the source hangs up there are no problems, the destination hangs up. Did anyone had any experience on this problem? Thanks, Jason

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[Freeswitch-dev] Friday Free For All
by Ken Rice 27 Sep '13

27 Sep '13

Hey Guys Don¹t forget to join us on the FreeSWITCH Friday Free For All! SIP:888@conference.freeswitch.org or https://webrtc.freeswitch.org and call 888 Don¹t forget to keep up with what's going on with ClueCon Weekly and ClueCon with our Google+ group at https://plus.google.com/u/0/communities/108955134544785139154 and find out whats going on with FreeSWITCH via https://plus.google.com/u/0/communities/110013892794022037793 Follow us on @FreeSWITCH_wire on Twitter.com Let us know if theres something you want to see covered on ClueCon Weekly! We¹ll get someone in to talk about it! -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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[Freeswitch-dev] Freeswitch chooses the first codec and ignores the rest
by Sam Russell 25 Sep '13

25 Sep '13

I've had a couple of calls recently that have been a little weird - here's the scenario: A (local freeswitch) sends invite with PCMU, PCMA as codecs B (foreign) sends 200 OK with PCMA, PCMU as codecs Freeswitch sees the response and sets audio codec to PCMA, but other end sends us PCMU (as it was our first choice in this case - sometimes they choose a random one that both of us support). RTP is fine with this - each packet refers to the # of the codec (either standard or a reference to an rtpmap from the SDP), but freeswitch doesn't like it - it forwards the PCMA from my soft phone, but the PCMU coming in from the foreign phone gets dropped by freeswitch and doesn't get forwarded onto my soft phone (packet trace shows freeswitch being the black hole). Debug output shows freeswitch setting the audio codec, and I suspect this means it then only accepts RTP with that codec, instead of handling all codecs that it offered in its invite. Is this expected behaviour? I went trawling through RFCs yesterday and nothing jumped out at me, and I'm seeing this on a couple of call managers - also sometimes the other end will send H263+ even though both ends offered H264 as preferred, and freeswitch drops this in the same manner. I think freeswitch should be handling all codecs that it offers, especially for video when it only does pass through, so spinning up instances of extra "codecs" shouldn't create much overhead. Sent from my iPhone

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[Freeswitch-dev] Debugging DTLS for WebRTC
by Sam Russell 25 Sep '13

25 Sep '13

Hi, I've spent a few hours digging through captures and RFCs and I think I found where I'm getting caught - having the server key isn't enough - you need the client key (i.e. from your browser). I've got a couple of issues I'm tracking down with WebRTC support for FreeSwitch, does anybody know how to export private keys from Chrome/Firefox that they use for WebRTC? If I can get the browser private key AND the key from Freeswitch (dtls-srtp.key by the looks) then I can decrypt a DTLS stream and figure out what's going on. Cheers Sam

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[Freeswitch-dev] About the voice response, when the user can't find
by sparklezou 24 Sep '13

24 Sep '13

Hi develop team, When the sip user account is NOT create in Freeswtich, and the number is in the dial plan, freeswitch will respone "Goodbye", NO more info. Suggest to response, "NO such user, Goodbye". 2013-09-24 11:19:22.163081 [NOTICE] mod_loopback.c:945 Pre-Answer loopback/voicemail-a! 2013-09-24 11:19:22.163081 [NOTICE] mod_dptools.c:1253 Pre-Answer loopback/voicemail-b! 2013-09-24 11:19:22.163081 [WARNING] mod_voicemail.c:3452 Can't find user [3666(a)172.0.0.10] Thanks! 2013-09-24 sparklezou

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[Freeswitch-dev] how to encrypt the sip message?
by 彭勇 20 Sep '13

20 Sep '13

one of my sip gateway using a encrypted SIP message. char * sip_encrypt(char* str) { ... return encrypted_str; } i should send encrypted to the gateway. the gateway also send me a encrypted SIP message, and i should decrypt it. then how can i do it in Freeswitch? thanks -- Peng Yong

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[Freeswitch-dev] Freeswitch rejecting streams when a=rtpmap:96 /0
by Sam Russell 19 Sep '13

19 Sep '13

Hi, I've been using Freeswitch as a PBX, and I'm having problems when making video calls when the video stream is rejected. The other endpoint (happens on 4 distinct endpoints) sends back a m= line with port 0, and an a= line with codec type as /0. When this happens, I get the error "sofia.c:1431 nua_i_media_error: unknown event 22: 988 Incomplete offer/answer" and the call is rejected. I think Freeswitch is getting upset with the a= line after the "m=video 0" line because there's no codec name. The RFC says that any a= lines after an m= line with port=0 must be ignored, is this a bug? t=0 0 m=audio 31720 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 m=video 0 RTP/AVP 96 a=rtpmap:96 /0

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Re: [Freeswitch-dev] [freeswitch-sec] FreeSWITCH selinux policy sponsorship
by Brian West 19 Sep '13

19 Sep '13

Kristian, I personally do not have any experience in this area, Maybe someone on the Dev/Users list would be interested in this topic? /b -- Brian West brian(a)freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire , @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) On Sep 9, 2013, at 10:10 AM, Kristian Kielhofner <kris(a)kriskinc.com> wrote: > Hello all, > > My company would like to sponsor the development of a proper selinux > policy for FreeSWITCH. How should we get this going? > > -- > Kristian Kielhofner > > _______________________________________________ > freeswitch-sec mailing list > freeswitch-sec(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-sec

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[Freeswitch-dev] freeswitch.Session questions
by Wolfgang Pichler 18 Sep '13

18 Sep '13

Hi all, i do have the following scenario with a freeswitch system call comes in -> lua script gets executed -> lua script does create new session with freeswitch.Session('dialStr') <- without the original session as parameter - because i want to be able to route to the next destination when the first does not work lua script does check if new session is ready lua script does check if old session is still ready lua script does bridge the two sessions So far so good - but it does happen from time to time - that the original session does get a hangup - and the ready function does still return ready - or the hangup is after the ready check - and before the bridge function. And freeswitch does in this case still create the bridge - so there is b-leg with one way audio... This does sound strange to me Shouldn't the bridge function check again leg-a and leg-b - then bridge - and if one of those legs do hangup - stop the bridge ? Or did i understand the freeswitch.Session function wrong ? can i specify the original session there - without hanging up the original session when the new session won't get ready ? br, Wolfgang

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