December 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Lenny's killing it.
by Gabriel Gunderson 26 Jan '14

26 Jan '14

No way! Lenny is on a roll! It's too perfect (listen in order)! 1) http://itslenny.com/recording.php?file=4d8cde0cc5a4d86ccf221aa5b2903e95 2) http://itslenny.com/recording.php?file=4fdd3f445fff8e1b73456af602367b6e Classic! Gabe

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How to refuse T.38 requests
by gegetel 24 Jan '14

24 Jan '14

Hello, My locale UAs do not know anything about T.38 protocol and I want to send a fax to some PSTN subscriber via OVH, my french VoIP provider, in an RTP/PCMA session. This session begins normally but, as soon as OVH's gateway recognizes fax tones from remote and/or locale side, an undesirable UPDATE with SDP for a T.38 session is sent to my side. The problem is that, immediately, FreeSwitch says "200 OK" instead of something like "488 Not acceptable here" and this is the beginning of a dialogue of the deaf which only takes end when remote fax-machine hangs up. I have done several things that I found in the WIKI: In my dialplan added these lines: <action application="set" data="sip_ignore_reinvites=true"/> <action application="set" data="fax_enable_t38=false"/> <action application="set" data="fax_enable_t38_request=false"/> In ~/conf/vars.xml: <X-PRE-PROCESS cmd="set" data="refuse_t38=true"/> In ~/conf/sip_profiles/external/sip.ovh.fr.xml: <param name="t38-passthru" value="false"/> But without better result. This is an example of the UPDATE request: UPDATE sip:gw+sip.ovh.fr@92.146.52.82:5080;transport=udp;gw=sip.ovh.fr SIP/2.0 Call-ID: 9686bdc8-e997-1231-27aa-001fc61584f4 Contact: <sip:10.7.1.60:5060> Content-Type: application/sdp Cseq: 29265730 UPDATE From: <sip:0556ZZZZZZ@sip.ovh.fr>;tag=00-08184-01c44a83-0e1305300 Max-Forwards: 30 Record-Route: <sip:91.121.129.20:5060;lr> To: \"0033972XXXXXX\" <sip:0033972XXXXXX@sip.ovh.fr>;tag=75761tHD7541g Via: SIP/2.0/UDP 91.121.129.20:5060;branch=z9hG4bK-DIVG-481b17bf-03a84e77 User-Agent: Cirpack/v4.56 (gw_sip) Content-Length: 299 v=0 o=cp10 138814832168 138814832170 IN IP4 10.7.1.132 s=SIP Call c=IN IP4 91.121.128.144 t=0 0 m=image 35996 udptl t38 a=sendrecv a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxRateManagement:TransferredTCF a=T38FaxMaxBuffer:1000 a=T38FaxMaxDatagram:200 a=T38FaxUdpEC:t38UDPRedundancy And that is the OK answer: SIP/2.0 200 OK Via: SIP/2.0/UDP 91.121.129.20:5060;branch=z9hG4bK-DIVG-481b17bf-03a84e77 Record-Route: <sip:91.121.129.20:5060;lr> From: <sip:0556ZZZZZZ@sip.ovh.fr>;tag=00-08184-01c44a83-0e1305300 To: \"0033972XXXXXX\" <sip:0033972XXXXXX@sip.ovh.fr>;tag=75761tHD7541g Call-ID: 9686bdc8-e997-1231-27aa-001fc61584f4 Cseq: 29265730 UPDATE Contact: <sip:gw+sip.ovh.fr@92.146.52.82:5080>;transport=udp;gw=sip.ovh.fr User-Agent: FreeSWITCH-mod_sofia/1.5.8b+git~20131213T181356Z~87751f9eaf~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 201 v=0 o=FreeSWITCH 1388128642 1388128643 IN IP4 92.146.52.82 s=FreeSWITCH CH c=IN IP4 92.146.52.82 t=0 0 m=audio 19678 RTP/AVP 8 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 Any help much appreciated, thanks in advance...

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Wanpipe with recent versions of Linux
by Steve Underwood 23 Jan '14

23 Jan '14

Hi all, Does anyone run wanpipe with any recent version of Linux? Wanpipe 7.0.8 seems fine when compiled for Centos 6.4. However, If I try any recent kernel (e.g. the ones in recent versions if Fedora) wanpipe fails to build for a variety of reasons (i.e. if I work around one blockage I just keeping hitting another one). Regards, Steve

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uuid_setvar prefixing
by Callum Guy 08 Jan '14

08 Jan '14

Hi All, When using inbound ESL method uuid_setvar I understand the variables will get prefixed with "variable_" such that... uuid_setvar chan123456 MyVar MyVal ...reports in the next event as: [variable_MyVar] => MyVal My issue is that i'm building an event based system that stores custom channel data using this operation and i don't want user entered variable names overwriting exiting channel data. I am counteracting the above concern by simply adding my own prefix to all set/get operations but I am still interested in finding out the significance of these variables and if its only FreeSWITCH modules (i.e. "playback_terminators" for playback application) that populate those in the "variable_" list or if these are used deeper in the core as well? Any further information would be really useful, thanks, Callum ______________________________ Callum Guy Developer X-on Framlingham Technology Centre Station Road, Framlingham, Suffolk, IP13 9EZ T 0333 332 0116 E callum.guy(a)x-on.co.uk X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD Company Registration No. 2578478 This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. Please consider the environment before printing this email.

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dynamic change conference profile in conference.conf.xml
by Shijun Bian 06 Jan '14

06 Jan '14

Hi all, I just be in trouble in the conference setting. I am building up a freeswitch cluster to server >100,000 users and all these users will be in thousands conference room. because the user base is growing day by day , so I have to dynamically add freeswtich node if necessary. But the ip address of the conference room is specified in the profile as the "domain" field in the conference.conf.xml which is loaded when the freeswitch started . now the problem is I can not restart all the freeswitch nodes the add a new profile into the conference.conf.xml when a new IP is be added into the cluster. so I wonder is it possible that I could add a new conference profile dynamiclly? I have been in trouble and google around for days but still can't find a way to solve it. Please help, thanks in advance. -- *Best Regards* *ShiJun Bian*

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iLBC and spandsp codecs not available in stock install
by Francis 02 Jan '14

02 Jan '14

Hi, When I do a show codecs in fs_cli, none of the spandsp codecs show up, even though GSM and G722 are enabled in vars.xml and mod_spandsp is enabled in modules.conf.xml. The module was compiled and installed. I'm having the same issue with iLBC. This isn't consistent with all modules however, since speex works just fine. Any ideas? Francis

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strange behavior in setting origination_caller_id_number
by Jun Sun 02 Jan '14

02 Jan '14

I have a very simple call plan to allow internal extension to call external PSTN numbers. See below. <extension name="domestic.example.com"> <condition field="${toll_allow}" expression="domestic"/> <condition field="destination_number" expression="^(\d{11})$"> <action application="set" data="origination_caller_id_number=11111111"/> <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/> <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/> <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/>  <=== NOT WORK  <=== NOT WORK <action application="bridge" data="[origination_caller_id_number=111111]sofia/gateway/idt/$1"/> <=== WORKS!! </condition> </extension> There a couple of things strange here: 1. I have to put origination_caller_id_number inside data field of "bridge" application in order for the dialing out to happen. Why putting it outside as channel variable does not work? 2. Actually because bridge command is for leg-B, really it should be "effective_caller_id_number" not "origination_caller_id_number". Is that right? I'm using IDT express as SIP carrier. I don't know if they changed something or I changed configuration in freeswitch. Now it seems I have to set origination_caller_id_number if I want to use IDT for outbound calls. I'd really appreciate your help. Cheers. Jun

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Playback in a loop
by Ali Pey 02 Jan '14

02 Jan '14

Hello, Is there a way to playback a file in a loop? I'm trying to find a way to put the a-leg on hold with music while b-leg gets some prompts to do a initiate a transfer. I have looked at endless_playback, but the problem seems to be that there is no way to stop the play back and bridge the call again. Is that still true with endless_playback? Regards, Ali Pey

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Cross compiling freeswitch for raspberry pi
by Dušan Dragić 01 Jan '14

01 Jan '14

Hi all, This weekend I tried my luck at cross-compiling freeswitch for raspberry pi (armv6). After a few bumps along the way (opened jiras for those) I managed to build it, but when running freeswitch on the target system it segfaults during startup. I'm building on Fedora 19 x86_64 using the prebuilt toolchain from https://github.com/raspberrypi/tools (I can build a toolchain from source if anyone thinks it might be better). Raspbian debootstrap created base system is used as sysroot (also tried loop mounted raspbian image as sysroot). My simple build environment setup script: http://pastebin.freeswitch.org/21698 The build process (source the script, configure and compile): ./bootstrap.sh . ~/fs-cc-env.sh ./configure --build=x86_64-redhat-linux --host=arm-linux-gnueabihf --with-sysroot=$SYSROOT make make install didn't work (I guess it's kinda expected, didn't really look into it) so I just copied the binaries, libraries and modules to rpi. Also mod_spidermonkey and mod_xml_rpc were disabled. Didn't even try to build mod_spidermonkey and mod_xml_rpc failed to build (well it builds but makefile tries to run libs/xmlrpc-c/lib/expat/gennmtab/gennmtab which was compiled for arm and should have been x86, probably should create jira for this as well). Console log when FS is started: http://pastebin.freeswitch.org/21699 and backtrace: http://pastebin.freeswitch.org/21700 Is my build environment sane? What am I missing, any hints? Has anyone successfully cross-compiled FS for raspberry pi? Thanks, -- Dušan Dragić

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Parallela board - interesting for FreeSWITCH?
by Bob Hartwig 01 Jan '14

01 Jan '14

Does anyone here have an opinion on the Parallela board - http://www.adapteva.com/products/parallella/parallella/ - as a platform for FreeSWITCH? Could it be appropriate for applications that need to be small and inexpensive, but also capable of doing a fair amount of real-time transcoding? Bob

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