December 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Issues with bidirectional early media
by Max Bridgewater 31 Dec '10

31 Dec '10

Hi Folks, A while ago, I tried the following code and it indeed worked with my carriers here in Canada: session.preAnswer(); dtmf = session.getDigits(4, "#", 10000); console_log("got total digits: " + dtmf + "\n"); Eight months ago, I could read DTMF entered by the users while in early media. It seems it's not working anymore. Has this behavior changed in Freeswitch or all of the carriers turned it off? I tried to test it with Xlite but the newest version seem not to send DTMF at all. I would highly appreciate if somebody could test this to confirm that it's indeed related to my carriers. Things work fine when I replace preAnswer() with answer(); but that's not what i want to do. Thanks, Max.

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Freeswitch with TLS
by Michel Freiha 31 Dec '10

31 Dec '10

Dear Sir, I'm planning to use Freeswitch server as a proxy server before OpenSIPS...The Job of freeswitch is only when I need to use SIP over TLS or SRTP when there is a severe firewall on user side...The Client will send SIP over TLS to freeswitch and the freeswitch will forward these packets to openSIPS or kamailio or whatever Registrar server... Is that possible? How many concurrent TLS connection a freeswitch server can handle? Regards

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CPS
by Sam 31 Dec '10

31 Dec '10

Hello, Was testing the environment with media bypass, could see the CPU for FS application shooting to 200% on quadracore 64 bits debian for CPS of 70. Is it true for the environment or can be scaled further ? Regards Sam

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Group / channel limit
by Sameer 31 Dec '10

31 Dec '10

Hi guys, Is there a way to check limit status or number of active calls on a number from XML curl PHP And then return the dial plan commands?

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scaling
by Sam 31 Dec '10

31 Dec '10

Hi, Can it be done such that 2 servers having identical configuration for conference , and treating those 2 servers as one whole server . Such that when the conference room is initiated on one server ,the person on the second server can join the same conference room, how such could be synchronized. Such that any one logs in on either server can initiate the same conference room and talk to each other. Regards Sam

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FreeSWITCH SBC Setup (HA mode)
by Ghulam Mustafa 31 Dec '10

31 Dec '10

Hi everyone, i am trying to setup FreeSWITCH as a SBC in ha mode; for this purpose i have setup two FS boxes(identical) in HA mode using heartbeat, mysql as odbc backend and enabled track-calls in both sip profiles. Everything is working as expected i.e sip sessions successfully recovered during failover in both directions. i am just curious and would like to know should i go for odbc or is it possible to avoid odbc and use drbd for /opt/switch/db directory to keep .db files in sync between two fs boxes. is it likely to work? another question is related to performance when track-calls is enabled, we are supposed to handle ~300 sip concurrent calls around the clock in proxy_media mode, coming from our sip provider (on external profile) and forwarding to our call-center server (configured as internal sip gateway), we have plenty of good hardware (quad-core Xeon servers with 8Gigs ram), is it good to go :) or can raise alarms? Thanks and best regards, -- Ghulam Mustafa cell: +92 333.611.7681 sip: cyrenity(a)ekiga.net mail: mustafa.pk(a)gmail.com web: cyrenity.wordpress.com

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CentOS 5.5 - FreeSWITCH - Audio Quality issues
by Aloysius Lloyd 30 Dec '10

30 Dec '10

Hi All, *FreeSWITCH environment* Hardware - Intel Core 2 Quad 2.4GHz , 4GB RAM OS - CentOS 5.5 FreeSWITCH Version 1.0.head (git-fcd6c54 2010-12-19 00-13-08 -0500) Ethernet : Public IP + 10Mbps port Calls Between Phones and outside calls perfect. But Voice Quality is not great in the following scenarios 1. Voice Mails ... most of the voicemails breaking voices .. lots of static 2. IVR ... breaking voices 3. System Automatically Hangup ... while people leaving messages.. Do I need to adjust any parameters ? Any help is appreciated. Thanks Lloyd

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ftmod_libpri: support for overlap dial
by Nagalenoj H. 30 Dec '10

30 Dec '10

Dear Friends, When I tried ftmod_libpri to check whether it supports overlap dial. It didn't work. So, is there anyone trying to implement overlap dial in ftmod_libpri? -- Regards, Nagalenoj H.

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group_call - ignore the sip_from_user
by Aloysius Lloyd 30 Dec '10

30 Dec '10

Hi All, If a member of a group calling to the group , I want to ignore the member from the group. Is there any way to do this ? Thanks Lloyd

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skypopen load problem
by Seven Du 30 Dec '10

30 Dec '10

Hi, So finally I get a chance to move to CentOS 64bit. git HEAD on CentOS 5.5 64bit. skype 2.0.0.72 static. Load error: http://pastebin.freeswitch.org/14888 Seems sk_2 read extra AUTOAWAY OFF and ERROR 68, however, client.c never shows those extra lines on my test against all my 20 instances. I use multi-instances-same-username btw. ./client :302 Initialized XInitThreads! PROTOCOL 7 #ciapalino PONG CONNSTATUS ONLINE CURRENTUSERHANDLE idapted_voip_10 USERSTATUS ONLINE Sometimes I can load till sk_8, sometimes sk_1 doesn't load. Any clue? Thanks. -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn

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