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Cant compile last git on FreeBSD
by korn 01 Dec '10

01 Dec '10

Hello, I,m trying to compile last git on FreeBSD 7.1 and receive this error in mod_spandsp: making all mod_spandsp Creating mod_spandsp_la-mod_spandsp_fax.lo Compiling mod_spandsp_fax.c ... cc1: warnings being treated as errors mod_spandsp_fax.c: In function 'configure_t38': mod_spandsp_fax.c:721: warning: implicit declaration of function 't38_set_fastest_image_data_rate' mod_spandsp_fax.c: In function 't38_gateway_on_consume_media': mod_spandsp_fax.c:1637: warning: implicit declaration of function 't38_gateway_rx_fillin' gmake[4]: *** [mod_spandsp_la-mod_spandsp_fax.lo] Error 1 gmake[3]: *** [mod_spandsp-all] Error 1 gmake[2]: *** [all-recursive] Error 1 configure and gmake in ./libs/spandsp are successful can anybody tell me, why I'm getting error WBR Evgeny

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mod_fifo or mod_callcenter
by abubacker 01 Dec '10

01 Dec '10

Dear all, I want to use mod_fifo or mod_callcenter to perform queuing operation , but I dont want this application to connect the agent and the customer I have the external script to do that but it should give me the dial string of the agent where the customer is likely to connect with. Is this possible or can we do this using some work around ? Thanks in advance ! -- Best regards, N.Abubacker , Associate system engineer , bk systems pvt ltd , Ph : 9144-43902701 Disclaimer: http://www.bksystems.co.in/email-policy

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FsGui Windows build
by Dmitry Bely 01 Dec '10

01 Dec '10

Hi, Has anyone succeeded in building FsGui for Windows from the current repository sources? The older one available at http://files.freeswitch.org/fsgui/fsgui.exe suffers form at least one serious problem (100% CPU load) that makes it hardly usable. Git logs hint that it may be fixed already. - Dmitry Bely

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DynamicRTP-Type-98
by Troy Anderson 01 Dec '10

01 Dec '10

I noticed a problem with calls originating from the mod_callcenter. The order of the codecs in the Media Description portion of the SDP is different for calls from mod_callcenter than when dialing the extension from another extension. Specifically, ITU-T G.711 PCMU is coming first on calls that work, and DynamicRTP-Type-98 is coming first on those that don't. A Polycom 450 is returning a 488 Not Acceptable Here message on the callcenter calls, and an Aastra 6731i has extremely choppy audio. If I tell the Aastra to only allow G722, all works as expected. I have my codecs set up in FS as global_codec_prefs=G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM Can anyone explain what DynamicRTP-Type-98 is and why it would be in the SDP when I'm not specifying it in the codec list? Here's what the Media Description portion looks like when calling extension to extension: Media Description, name and address (m): audio 10140 RTP/AVP 0 98 99 9 8 3 101 13 Media Type: audio Media Port: 10140 Media Protocol: RTP/AVP Media Format: ITU-T G.711 PCMU Media Format: DynamicRTP-Type-98 Media Format: DynamicRTP-Type-99 Media Format: ITU-T G.722 Media Format: ITU-T G.711 PCMA Media Format: GSM 06.10 Media Format: DynamicRTP-Type-101 Media Format: Comfort noise Here's what the Media Description portion looks like when mod_callcenter initiates the call: Media Description, name and address (m): audio 12606 RTP/AVP 98 99 9 0 8 3 101 13 Media Type: audio Media Port: 12606 Media Protocol: RTP/AVP Media Format: DynamicRTP-Type-98 Media Format: DynamicRTP-Type-99 Media Format: ITU-T G.722 Media Format: ITU-T G.711 PCMU Media Format: ITU-T G.711 PCMA Media Format: GSM 06.10 Media Format: DynamicRTP-Type-101 Media Format: Comfort noise Thanks! -Troy

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ECHOCANCEL
by Raul Baldeon 30 Nov '10

30 Nov '10

Hello, I've connected a Sangoma E1 card without Echo Cancel. I'm using FreeTDM (Wanpipe Mode). My card does't have echo cancel by hardware. Is there some way to configure soft echo cancel in my current configuration (Wanpipe Mode)? Something like Oslec or MG2. Thanks, This is my current config: *conf/freetdm.conf* [span wanpipe wp1] trunk_type => e1 b-channel => 1:1-15 d-channel => 1:16 *autoload_configs/freetdm.conf.xml* <configuration name="freetdm.conf" description="Freetdm Configuration"> <settings> <param name="debug" value="0"/>    </settings> <config_profiles> <profile name="telco_1"> <param name="switchtype" value="euroisdn" /> <param name="signalling" value="cpe" /> <param name="overlap" value="yes" /> <param name="facility" value="yes" /> </profile> </config_profiles> <sangoma_pri_spans> <span name="wp1" cfgprofile="telco_1"> <param name="dialplan" value="XML"/> <param name="context" value="from-pstn"/> </span> </sangoma_pri_spans> </configuration> -- -- Raúl Baldeón Area de Soporte - Infobox Latinoamérica Tel. +511 7129130 Anexo 41 -- -- Raúl Baldeón Area de Soporte - Infobox Latinoamérica Tel. +511 7129130 Anexo 41

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Tutorial: FreeSwitch as Media Server and SBC for Kamailio 3.1
by Daniel-Constantin Mierla 30 Nov '10

30 Nov '10

Hey guys, a short note to inform that I updated my tutorial about using FreeSwitch and Kamailio together for large VoIP platforms. Besides upgrade to use latest Kamailio major stable release, v3.1.0, there are couple of new features added in the architecture of the VoIP platform: - along with providing media services (voicemail, conferencing, a.s.o.), FreeSwitch is used now also as SBC for topology hiding and media relay (this helps for transconding needs, palying audio messages during the early session or NAT traversal) - Kamailio config has added features to detect DoS/DDoS and scanning attacks, secure SIP communication over TLS, IP authentication and a bunch of neat things that help for an easier maintenance and update of parameters such as database connectivity details, local IPs, etc. Hope it will be useful for many of you. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Jan 24-26, 2011, Irvine, CA, USA http://www.asipto.com

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Siemenst C470IP and G.722
by Thomas Mueller 30 Nov '10

30 Nov '10

hi wanted to use G.722 with a Siemens Gigaset C470IP. It was all the time using G.729 so i've removed all codecs but G.722 in codec list. which lead me to INCOMPATIBLE_DESTINATION. output of "/log 7" on fs_cli 2010-11-30 17:41:26.216025 [DEBUG] sofia.c:4579 Remote SDP: v=0 o=1001 5012 72 IN IP4 192.168.236.161 s=Mapping c=IN IP4 192.168.236.161 t=0 0 m=audio 5012 RTP/AVP 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 2010-11-30 17:41:26.216025 [DEBUG] sofia_glue.c:4367 Audio Codec Compare [telephone-event:101:8000:0:0]/[G729:18:8000:20:8000] This leads me to the maybe inaccurate conclusion that Siemens Gigaset sends 101 for G.722 what freeswitch translates to telephone-event (according to vars.xml 101 is telephone-event). Reading further http://wiki.freeswitch.org/wiki/RTP_payload_list G.722 should have id 9. Am I right, that Siemens is wrong? - Thomas

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DTMF handling changed from 1.0.6 to GIT?
by Yehavi Bourvine 30 Nov '10

30 Nov '10

Hello, I've downloaded the snapshot from today and tried it with our current 1.0.6 configuration. It seems that DTMF is not relayed anymore between the profiles... We have the phones on one profile and a SIP gateway on another one. To make it simple, I've set my all parties to use INFO so I can trace it with SIPtrace. I see that the Polycom generating the SIP INFO commands for each DTMF, but they are not relayed to the other party. The local voicemail on FS detects them correctly. Both profiles have: <param name="dtmf-type" value="info"/> Any idea? Which debug to enable in order to trace it? Thanks! __Yehavi:

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VAD + session record
by Tamas Cseke 30 Nov '10

30 Nov '10

Hello, We have a problem with session recording if VAD is turned on. if there is no RTP there is no data written to the file so the it becomes shorter then the speech. I tried brigde_generate_comfort_noise=yes to generate the noise, but i think it doesn't work. I don't hear the noise only the total silence if VAD is turned on. Should it generate noise if there is no RTP packets (as fas as I understand, it should) or only if FS receives CN RTP packets (audio_bridge_thread check SFF_CNG flag on frame, but I don't really understand what does it mean. It must be CN packet? ) Should FS send CN packets when VAD is turned on, or not? Thanks, Tamas

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voipabuse blacklist
by François Delawarde 30 Nov '10

30 Nov '10

I've got new attacks on multiple servers every day that slow down FreeSWITCH a lot. When under REGISTER attack, a real call can take more than 30 seconds to establish. Anyone knows how to propose new IPs to the voipabuse blacklist? I can't seem to find a submit link in the http://www.infiltrated.net/voipabuse/ website, and currently have to maintain my own blacklist. Thanks, François.

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