June 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

FreeSWITCH Community Conference Call - ClueCon 2013 Schedule!
by Michael Collins 19 Jun '13

19 Jun '13

Hello all! This is a reminder that we will be releasing the ClueCon 2013<http://www.cluecon.com>schedule on today's community conference call <http://wiki.freeswitch.org/wiki/FS_weekly_2013_06_19>. We also have the whole FreeSWITCH core development team joining us today. The call starts at 1PM Eastern, 10AM Pacific. We look forward to having a large crowd and some stimulating conversation on today's call! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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Call dropping while using TLS in freeswitch
by Richa Richa 19 Jun '13

19 Jun '13

Hi All, I am new to freeswitch. And I want to secure my calls using TLS. I have two Aastra 6753i phones registered as Transport Protocol setting "UDP and TCP". When I try to call one extension from another. It rings a little, then got disconnected. Freeswitch log at: http://pastebin.freeswitch.org/21077 Regards, Richa.

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Re: [Freeswitch-users] tls SIP Trunk in Freeswitch
by mehroz 19 Jun '13

19 Jun '13

It looks as if , it does not decrypt itself over the 2nd server. The encrypted SIP message must decrypt on the other end, to understand what it is about. I am also confused , which certs and from which server need to be imported to the other server. Either cafile.pem of FS1 has to go to FS2, or cafile.pem of FS2 has to move over FS1. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/tls-SIP-Trunk-in-Freeswitch-t… Sent from the freeswitch-users mailing list archive at Nabble.com.

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CHANGED Skype gw to Weekly Conference Call
by Giovanni Maruzzelli 19 Jun '13

19 Jun '13

Hey FreeSWITCHers, while I was testing performance related innovation I invalidated by mistake the Skype account you called to connect to the Weekly Call. So, the "skypiax5" account you used to call is no more. Please call skype "freeswitch_conf_call" to be automatically connected to the Weekly Call. Sorry for the inconvenience, -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618

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Best practice for centralized user registration database
by Emrah 19 Jun '13

19 Jun '13

Hi all, What is the best practice today to handle users directory dynamically through a database, ideally PostgreSQL or MySQL? I'm kind of concerned by what I've seen so far in the available documentation. Do I really have to write an API to serve data from a DB to FreeSwitch? How about monitoring registration states? My goal is simple. I would like to have multiple FS instances share the same users directory and be able to tell on which server the client has registered. Your help would be greatly appreciated. Thanks, E

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having problem build 1.2.10 stable with VS2010
by Vincent Xia 19 Jun '13

19 Jun '13

i downloaded the FS 1.2.10 stable tarball and was trying to build it in VS2010, after several attempts to "build all", the libeay32 still cannot be built, resulting fail of other dependencies includeing FreeSwitchCore, VS was reporting “netinet/sctp.h”: No such file or directory then i added preprocess definitions OPENSSL_NO_SCTP and OPENSSL_NO_EC_NISTP_64_GCC_128, now it reports several unresolved external symbols, as BUF_memdup, BUF_strdup, BUF_strlcat, Camellia_set_key, EC_POINT_point2oct, RSA_flags, etc. so do i still missed any external libs or something was wrong with my settings?

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tls SIP Trunk in Freeswitch
by Abdullah 19 Jun '13

19 Jun '13

hi ,all please help me how to configure SIP (Tls) TRUNK in free-switch SIP Server. any idea. thanks in advance -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/tls-SIP-Trunk-in-Freeswitch-t… Sent from the freeswitch-users mailing list archive at Nabble.com.

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freeswitch suddenly very slow
by devel＠omninet.eu 19 Jun '13

19 Jun '13

Hello, I am facing a very strange problem: my fs server suddenly performs very very slow. Here a description of the system: Debian 7, odbc, mysql is on another server, xml_curl for the users Suddenly the server stops responding to registration requests or answers with a very big delay. Sometimes I see there a message “503 System Busy”. When this behavior starts, the active calls can continue, without any issue. The problem is with new requests of any type. Once, during this phase, I’ve tried to start manually (cli) the watchdog. It crashed the fs session immediately. I thought I found the problem and next time I started the watchdog while the system was running well. The watchdog did not recognize the problem when fs was running again very slow. Sofia siptrace shows the registration attempts. Almost no answers are shown. Tracing on the network side shows the registration attempts arriving, but usually there is no response from the server. No other activity towards mysql. Restarting freeswitch can solve the problem for a few minutes, sometimes for several hours, but it suddenly happens again. Lately, even after the restart the speed is not coming back to normal. This happened also during the night, with almost no calls and only registration requests coming in. In the morning there was no user registered. We restarted fs and the registrations started over. I’ve tried to restart also Apache, MySQL, without noticing any difference. Also a reboot of the server did not help. A workaround now is to increase the re-registration timer for the registration. Now I can see that sometimes the response to the registration attempt is immediately, sometimes there is a bigger delay, but enough for the client to stay registered. With some clients the registration has a bigger delay and they get un-registered. It looks like suddenly we have a bottleneck, but I can’t find it. I forgot to mention, that the server hardware has plenty of resources (24 GB RAM, 4 cores, cpu running very low). In the past I have seen similar behavior, when we had some attacks from “bad guys”, trying to place a lot of calls at the same time… We can’t see such an attack now. Does anybody have an idea into which direction to continue investigating? Thank you A

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2-party video call is not working
by Vincent Xia 19 Jun '13

19 Jun '13

hi all, im testing video features of FS including video record/playback, 2-party video call and video conference, everything works fine except the 2-party video call, after the call get connected i clicked "start" button of my eyebeam softphone to send video, the FS log displays 2013-06-18 17:54:16.346026 [DEBUG] switch_rtp.c:4686 Correct ip/port confirmed. 2013-06-18 17:54:16.346026 [DEBUG] switch_ivr_bridge.c:72 Send signal sofia/internal/sip:1006@192.168.0.100:58734 [BREAK] 2013-06-18 17:54:16.346026 [DEBUG] switch_ivr_bridge.c:73 sofia/internal/ 1002(a)192.168.0.100 video thread ended. while in video conference (which works fine), it's just 2013-06-18 17:54:16.346026 [DEBUG] switch_rtp.c:4686 Correct ip/port confirmed. i've had the following lines in vars.xml: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G722,PCMU,PCMA,GSM,H261,H263,H263-1998"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM,H261,H263,H263-1998"/> is there something wrong with my config?

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Re: [Freeswitch-users] Sangoma D500 mod_sangoma_codec
by Andre Demattia 18 Jun '13

18 Jun '13

Ds3 or T1. Anything to terminate sip traffic. Pretty much something that I can be my own sip provider. I'm not sure what equipment to get. I need 1000's if ports and 1000 cps. -----Original Message----- From: "Guillermo Ruiz Camauer" <grcamauer(a)gmail.com> Sent: ‎6/‎17/‎2013 9:56 PM To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> Subject: Re: [Freeswitch-users] Sangoma D500 mod_sangoma_codec By dial tone do you mean an analog line? Guillermo Sent from my iPhone On 17/06/2013, at 22:51, Ken Rice <krice(a)freeswitch.org> wrote: You mean something to gate SIP to the traditional TDM PSTN? (like T1s or E1s?) if so then you are looking for something like an A104DE (4 port T1 card) or something like an Adtran 900 series gateway if you want a hardware device On 6/17/13 8:40 PM, "Andre Demattia" <andretodd(a)verizon.net> wrote: Hmm I was looking for something to turn sip traffic to dial tone. I guess this isn't what I need. From: Guillermo Ruiz Camauer <mailto:grcamauer@gmail.com> Sent: ?6/?17/?2013 9:07 PM To: FreeSWITCH Users Help <mailto:freeswitch-users@lists.freeswitch.org> Subject: Re: [Freeswitch-users] Sangoma D500 mod_sangoma_codec The simple andwer is NO. The Sangoma D500 is a transcoding card. It converts an RTP stream in g729 to PCMU for example. It does absolutely nothing with SIP signaling. It is recommended for high traffic systems or for a large number of G729 licenses. Note that the D500 can be shared among several servers. There are smaller cards like the D100 which is the one I have which run up to 120 simultaneous transcoding streams. A similar card is the Digium TCE400B. Guillermo Sent from my iPhone On 17/06/2013, at 20:07, Andre <andretodd(a)verizon.net> wrote: HI, has anyone used Sangoma D500 with freeswitch and if so how does it work? Does this card connected to a Gig E become a SIP Trunk? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@list [The entire original message is not included.]

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