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Have someone zrtp with correct sas ?
by J.B. BERLIN 29 Aug '13

29 Aug '13

Dear users, I try run FS last git build from today 19. June 2011 with ZRTP. Zfone show me green but the sas are everytime different! Tested with 2x Notebooks on Fedora 14 Twinkle(via yum) & with two desktops with Win7 x64 / Zfone 0.92.build.219 and X-Lite on both sides. FS is build with libzrtp-0.81.514 zrtp.lua is enabled ! global_setvar zrtp_secure_media=true" is set Or have someone a real good step by step guide ? Many thanks and best regards Jacek

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CONFIG_HZ=1000 vs mod_timerfd -- preferred approach
by Jayanth Acharya 29 Aug '13

29 Aug '13

Assuming that one has full control over the kernel configuration and FS build / config options, is there a preferred mechanism between the 2 for letting FS have access to high resolution timers, i.e. - CONFIG_HZ=1000 - mod_timerfd This was one step I missed in my FS setup, and I think it might explain the extremely choppy, broken, garbled announcement I hear when I call into my gsmopen EP via phone. J

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Implications of [inbound|outbound]-use-callid-as-uuid in sofia.xml
by Kim Rostgaard Christensen 29 Aug '13

29 Aug '13

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi list First post, so I would just like to give a big hat tip to the people involved - extremely nice project. Well documented, nice community We recently moved from Asterisk to FreeSWITCH, and have not looked back since. Asterisk is a very nice project, but AMI is less suited our needs than ESL. Today, while mucking about with PJSUA, I discovered the call_id to be (entirely) different from the uuid assigned to the channel in FS. While this appears to remedied by the *-use-callid-as-uuid options in sofia.xml I still wonder if there are any potential complications/slowdown associated with using the option? Best - -- Kim Rostgaard Christensen | krc(a)retrospekt.dk | http://retrospekt.dk -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.11 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://www.enigmail.net/ iQEcBAEBAgAGBQJSHcgfAAoJEKRZG9Ehk6ybKZQH/3MvKC1QBmgX+6mH3fwnZWcI ezJ2XxL6EX+09uqBTbXnk2i8AJHMhZbMAokKJuMcyZHfJvOcQiBnj66fK7njPSBX A5/6ieXEzsxI+QHSu26t1pnhFLltgvkOGD2wVVf0e+bQhu027zDM6WW46IJtx3oa NBK3gC/oFOw/by0ejeEHsWOwpUvtimEdFRgSJFOYA03L+SEQWQn34uoiCC0fMHmS yKt1axCU04GDeN9nIEhrJTC+n/B5BiK4FcwbeMeUvj/VrK/+i57XCF+Wnr2qWJjy kBo4Rg1nDURw3nfQiBOhJ27ja5G4kUFAfiJzHs0/yjEsl233Ovakl6GIiwvlkpk= =+QMf -----END PGP SIGNATURE-----

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sip-force-contact problem in FS 1.2.12
by Juan Antonio Ibañez Santorum 29 Aug '13

29 Aug '13

Hello, Do you know if there are any changes in the way of manage sip-force-contact var in version FS 1.2.12? I have one NAT test endpoint which registers as 'Registered(UDP-NAT)' in version 1.2.12 leaving contact IP intact while registers as ''Registered(AUTO-NAT-2.0)' in version 1.2.10 forcing contact IP to its public IP. Tests done within the same machine and same configuration. Regards

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Recording Duration on EC2
by Byron Clark 29 Aug '13

29 Aug '13

I'm using FreeSWITCH on EC2 to call into a conference bridge and record the audio from the conference. It works well except for one thing: the recording contains all the audio from the call, but the duration of the file is shorter than the call. The duration is typically 3-5 seconds shorter than the call for each 5 minutes of call. On a recent test, wall clock showed a duration of 15:32.93 but the duration of the recording file was 15:19.96. The ugly part is that this only happens on EC2 instances where there is some CPU steal time (<5%) occurring. Here's my setup: Operating System: Ubuntu 12.04 FreeSWITCH: 1.2.12 Command I'm using to start the call: originate {record_waste_resources=true}sofia/external/SIPADDR &record(/tmp/record1.wav) default default Any ideas on how to make the recording duration actually match up with the call duration? Or even what's really going wrong so I can work on fixing that? -- Byron Clark

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Secure Websocket Setup
by Ricardo Saavedra 29 Aug '13

29 Aug '13

Hello, The webrtc setup instructions are:      Now I have a test server for the Websocket and is working. ): For the secure websocket I have some questions: Freeswitch have any tool to generate a self-signed wss.pem file? Can I use the files: usr/local/freeswitch/certs/dtls-srtp.crt and usr/local/freeswitch/certs/ dtls-srtp.key to generate the wss.pem that I need? Thank you, Ricardo Saavedra

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mod_gsmopen - extremely choppy announcement on placing call
by Jayanth Acharya 28 Aug '13

28 Aug '13

Hi, For the very first call I place using my mobile phone, to the UMTS dongle that is used as a mod_gsmopen endpoint, I distinctly hear an announcement, but that is extremely choppy, and I am unable to understand what it says. Soon after, I get a SEGV -- repeatable and already have a JIRA filed for same. What could be the reason for the extremely chopping announcements? Is it the rather low-end Celeron 1.5GHz w/ 512MB RAM. Thanks in anticipation for any hints / pointers. cheers, Jay

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Fw: SQL error on 'show calls count'
by Georgi Stefanov 28 Aug '13

28 Aug '13

Hello all, I have just found something strange Please see the following freeswitch@internal> status UP 0 years, 0 days, 0 hours, 0 minutes, 20 seconds, 669 milliseconds, 247 microseconds FreeSWITCH (Version 1.4.0 ) is ready 0 session(s) since startup 0 session(s) - 0 out of max 2200 per sec 3000 session(s) max min idle cpu 0.00/94.00 Current Stack Size/Max 240K/8192K freeswitch@internal> version FreeSWITCH Version 1.4.0 () freeswitch@internal> show calls count 0 total. 2013-08-28 13:40:28.554849 [ERR] switch_core_sqldb.c:1126 SQL ERR: [select count(*) from basic_calls where hostname='my-domain-name-as-example'] database disk image is malformed freeswitch@internal> How this happened? 1.I have two freeswitches (same version 1.4.0 installed yesterday) 2.2000 calls from FS1-> FS2 3.FS3 just died 4.I have launched FS2 again (freeswitch -nc) 5.Open fs_cli 6.Execute the commands shown in listing above Is this bug? Am I missing something? p.s. Forgive me if this is not the right list to post this message

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RFC4579- Adding a Participant by Focus- Invite with contact ; isfocus
by Jonathan Hunter 28 Aug '13

28 Aug '13

Hi Guys, I previously posted questions about getting RFC4575/4579 working with Freeswitch, and in terms of SUBSCRIBE/NOTIFY this is working fine so thanks very much for the help. I just wondered if it had been tested, in particular around RFC4579, and section 5.2; 5.2. INVITE: Adding a Participant by the Focus - Dial-Out To directly add a participant to a conference, a focus SHOULD send an INVITE to the participant containing a Contact header field with the conference URI and the 'isfocus' feature parameter. I have tested this, I cant see ;isfocus on the contact header field, when getting freeswitch to add a participant using originate, be it manually or via api. Is there again any particular syntax required for this to work, or has it been tested, as I just see the standard contact header of <sip:mod_sofia@xx.xx.xx.xx:5080> and no ;isfocus. Many thanks Jon

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ZRTP : Identical SAS with default media mode?
by mehroz 28 Aug '13

28 Aug '13

HI All, Is it possible to get same SAS with default media mode i,e when media is processed ? Assuming action application="set" data="zrtp_enrollment=true" , should set the FS as trusted MiTM, and pass the same SAS to called party! But I don't see that behaviour ! -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/ZRTP-Identical-SAS-with-defau… Sent from the freeswitch-users mailing list archive at Nabble.com.

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