July 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Server having two sangoma cards and can't handle full capacity
by Shaik Bawajan 22 Jul '14

22 Jul '14

We are having two (Sangoma) Voice Cards, one with 8-ports and other with 16-ports. We use Freeswitch with ESL library to connect and process the Outbound calls. The issue what we are facing is when we use all 690 channels, we see segmentation_fault in Freeswitch and big dump file is written (ie., > 10gb) and which would crash the system, because of high CPU\IO. But if we reduce the channels say, 450 we don't see any issue at all. What could be the reason for this issue. Appreciate any Help...

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(no subject)
by Nick Zaitsev 22 Jul '14

22 Jul '14

Good day to you. I set the freeswitch to work with skinny endpoints,i can make calls from sip devic3e to skinny device but i can not vice versa. I can only see pushing digits in fs debug?but no call processing. Could you help please? -- Nick Zaitsev

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Conference auto outcall questions
by Chris Martineau 22 Jul '14

22 Jul '14

Hi, Just setting up conferencing to use auto outcall. Q1. Is it possible to not hear anything from the called user until connected and then just hear the entry sound? Currently if I outcall to more than 1 person I get a mess of ringing until each person answers. Q2. If you auto outcall and it goes into voicemail is there anyway to kick the offending conference leg? This is especially a pain with mobiles as invariably the only way to stop the voicemail from just sitting there recording your conference is to hang up? Thanks Chris

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ClueCon is less then 2 weeks away! Have you registered?
by Ken Rice 21 Jul '14

21 Jul '14

Quick Reminder! If you haven¹t registered for ClueCon you¹ll want to do that today! Visit ClueCon.com for registration, schedule and speaker information! If you have not yet made your reservations at the hotel, rooms are in short supply. Get those reservations in now! And don¹t forget our Sponsors: Sangoma Technologies, Hardware and Software for your IP Communications Systems <http://sangoma.com> Range Networks, Flexible Cellular Networks using commercial opensource software. <http://www.rangenetworks.com> Corvisa Cloud, Contact Center Solutions <http://www.corvisacloud.com> FlowRoute, Provider of Business SIP Trunking, DIDs, and wholesale VoIP <http://flowroute.com> TruPhone, Global Mobile Provider. Need mobile service in more then one country and don¹t want more then one phone? <http://truphone.com> Simwood, European PSTN Operator, and Wholesale VoIP Provider <http://simwood.com> Fairwares, Re-inventing GSM with OpenSource <https://fairwaves.co> OnSip, Hosted VoIP and Unified Communications <http://www.onsip.com> 2600hz, Makers of Kazoo, A scalable hosted solutions platform <http://2600hz.com> Ziron, UK based wholesale wireline, VoIP and SMS provider. <http://www.ziron.com/> Yealink, Hardphones for your VoIP deployments <http://yealink.com> Ooma, VoIP Provider for Home and Small Business <http://ooma.com> OrecX, OpenSource Call Recording <http://orecx.com> VoiceTel, Wholesale VoIP Origination Services. <http://www.voicetel.com> ThinkGeek, Need toys for your inner geek or for the geek in your family? Check them out! <http://thinkgeek.com> -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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Follow us on Twitter @freeswitch & @cluecon for a chance to win $100.00 amazon gift card
by Brian West 21 Jul '14

21 Jul '14

Winner picked at ClueCon, You must follow us and we'll randomly pick someone you'll have to reply to claim the prize! ;) NO FOLLOW BOTS PLEASE! I posted a $50.00 discount code on twitter, find it, register and get 50.00 off, first person to use it gets it. ;) -- *Brian West* brian(a)freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest

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Windows installation problem!
by farzad jahangiri sisakht 21 Jul '14

21 Jul '14

Hello everybody I've been trying to test the Freeswitch on windows and as I need the mod_skypopen , so I need to build it from source. I followed the wiki page but it's very unclear at some points. I have built the solution but with many warnings and errors like files that couldn't fine and so on, below is the link to the txt file of output messages. Please kindly see if all are right here? And also after building the solution where is executable files and output, How can I run the Freeeswitch? Do I need to install it then or is it ready? This part is not clear for people who are not familiar with Visual studio. the installer package is more clear as it copies the files in the program files and ready to run. Do you know any installer with skypopen and gtalk module maybe? output file: http://www.aisoftworld.com/build.txt Many thanks for your help in advance.

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ISDN Module - ftmod_libpri
by Fred Schulz - privat 21 Jul '14

21 Jul '14

Hi all, i just try to setup an ISDN connection between a freeswitch and a cisco cue... I am trying to load libpri with bri support. But it allways shows: 2014-07-21 15:14:56.233050 [ERR] ftmod_libpri.c:2952 Unsupported trunk type: 'BRI', libpri too old Here are my configs: Freetdm.conf [general] cpu_monitor => no cpu_monitoring_interval => 2000 ; monitor usage every 2 seconds cpu_set_alarm_threshold => 90 ; whenever 90% of global CPU usage is reached, trigger the alarm. cpu_reset_alarm_threshold => 80 ; when the CPU usage decreases at 80%, clear the alarm. cpu_alarm_action => reject,warn ; Start rejecting calls when the CPU alarm is triggered and also print warnings. [span zt trunk1] trunk_type => bri b-channel => 1-2 d-channel => 3 [span zt trunk2] trunk_type => bri b-channel => 4-5 d-channel => 6 [span zt trunk3] trunk_type => bri b-channel => 7-8 d-channel => 9 [span zt trunk4] trunk_type => bri b-channel => 10-11 d-channel => 12 freetdm.conf.xml <configuration name="freetdm.conf" description="FreeTDM Configuration"> <settings> <param name="debug" value="1"/> <param name="sip-headers" value="true"/> </settings> <libpri_spans> <param name="node" value="cpe"/> <param name="switch" value="euroisdn"/> <param name="opts" value="omit_redirecting_number"/> <param name="dp" value="unknown"/> <param name="debug" value="none"/> <param name="dialplan" value="xml"/> <param name="context" value="public"/> <param name="node" value="cpe"/> <param name="switch" value="euroisdn"/> <param name="opts" value="omit_redirecting_number"/> <param name="dp" value="unknown"/> <param name="debug" value="none"/> <param name="dialplan" value="xml"/> <param name="context" value="public"/> <param name="node" value="cpe"/> <param name="switch" value="euroisdn"/> <param name="opts" value="omit_redirecting_number"/> <param name="dp" value="unknown"/> <param name="debug" value="none"/> <param name="dialplan" value="xml"/> <param name="context" value="public"/> <param name="node" value="cpe"/> <param name="switch" value="euroisdn"/> <param name="opts" value="omit_redirecting_number"/> <param name="dp" value="unknown"/> <param name="debug" value="none"/> <param name="dialplan" value="xml"/> <param name="context" value="public"/> </libpri_spans> </configuration> Tones.conf is default ... Thank you for any help

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Determine the database schema in freeswitch
by Nick Zaitsev 21 Jul '14

21 Jul '14

Good day to you, I am using database(postgres) to share mod_sofia wit multiple fs servers. I would like to use a special schema for the tables. Where shall I set it in odbc-dsn parameter? Best regards, Thank you for your time. Nick

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Problem about ZRTP call
by Zhang, Yu (Floyd) 21 Jul '14

21 Jul '14

Hi all, I'm not sure about the right FS expert to my problem so I sent mail to all users, hope someone could help me, thanks in advance! We are doing a small test with amazing freeswitch. Our goal is to use two linphone clients with FS to have an E2E call with ZRTP encrypted. Our environment is: Freeswitch-1.2.17 (compiled with zrtp-enabled)+Linphone+ubuntu12.04LTS We can do a non-zrtp call with FS successfully, but when we doing the zrtp, we encountered problem (before that, I have configure the environment with issuing global_setvar zrtp_secure_media=true global_setvar zrtp_enrollment=true The problem is, after serveral seconds good voice, there will be silence but call not drop, we can see the log info: 2014-07-21 11:30:41.904140 [DEBUG] switch_rtp.c:927 [ zrtp utils]: Send <SASRELAY> ssrc=1338834216 seq=51924 size=92. Stream 20:DH:SASRELAY 2014-07-21 11:30:41.904140 [DEBUG] switch_rtp.c:927 [zrtp protoco]: ERROR! Decrypt failed. ID=20:DH s=SRTP authentication failure (RTP size=172 ssrc=2087558422 seq=13/13 pt=0) 2014-07-21 11:30:41.924173 [DEBUG] switch_rtp.c:927 [zrtp protoco]: ERROR! Decrypt failed. ID=20:DH s=SRTP authentication failure (RTP size=172 ssrc=2087558422 seq=14/14 pt=0) .... 2014-07-21 11:30:51.384177 [DEBUG] switch_rtp.c:927 [ zrtp mitm]: WARNING! SASRELAY Max retransmissions count reached. ID=20 2014-07-21 11:30:51.384177 [DEBUG] switch_rtp.c:927 [ zrtp engine]: Enter InitiatingError State with ERROR:<Protocol Packets Retries Timeout>, notification Disabled. ID=20 2014-07-21 11:30:51.384177 [DEBUG] switch_rtp.c:927 [ zrtp]: Stream ID=20 DH switching <SASRELAY> ---> <ERROR>. The full log is in the attachment. Could you help me solve some of my questions about that: 1. How did this happen? Is there any additional configurations that I missed for ZRTP/SAS configure? If yes, which file I need to configure them in and is there an example for it? 2. I want to know the mechanism about the ZRTP running in FS, if the media flow has already encrypted in linphone side, then what FS side did with the voip call flow regarding ZRTP? I am looking into the code yet I still want to know the specific docs to clarify that 3. What role does the FS play in this kind of VoIP call? Does it do operations towards the encrypted traffic or it will encrypt that? Is there any method that FS only played the role of signaling server and traffic is flowed between point-to-point, i.e., by pass the freeswitch server. I would highly appreciate any of your help about my question, hope it could be solved under your great help. Look forward to your reply! Thanks, Floyd

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problum with Skypopen simultaneous incoming calls
by Shabbir abbasi 21 Jul '14

21 Jul '14

i have setup two skype interfaces with same skype user name when i calling from another skype to freeswitch this error is coming [WARNINGA 2193 ][interface2 ][UP,INPROGRS] no tech_pvt->session_uuid_str after INPROGRESS, let's hangup and call hangsup immediately if i setup two skype with diffrent skype usernames all works fine os = windows xp FreeSWITCH Version 1.5.12b~32bit ( 32bit) skype v 4.2 also tried skype v 4.2 v 5.2 v6.18 same result Could anyone help me ?.

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