May 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Freeswitch Video Capture and Playback
by Esben Stien 18 Oct '10

18 Oct '10

I'm using ekiga with mod_fsv, trying to record and play back video. When I dial the record extension, it seems to record something, as the video file gets bigger. Trying then to dial the extension for play back, just hangs up, with freeswitch saying: od_fsv.c:247 File version does not match! There seems to be no information on the FSV format or the mod_fsv module on the wiki. Is this at all supposed to work?. What clients and codecs were successful?. Any pointers as to what I can try?. -- Esben Stien is b0ef@e s a http://www. s t n m irc://irc. b - i . e/%23contact sip:b0ef@ e e jid:b0ef@ n n

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Re: [Freeswitch-users] Contact Field
by Kristian Kielhofner 16 Sep '10

16 Sep '10

I assume he is talking about the contact field in the INVITE, not the REGISTER... I don't know how to accomplish this off the top of my head but I'd be interested too (quite a few carriers request this). -- Kristian Kielhofner http://blog.krisk.org ------------------------------ *From*: freeswitch-users-bounces(a)lists.freeswitch.org < freeswitch-users-bounces(a)lists.freeswitch.org> *To*: freeswitch-users(a)lists.freeswitch.org < freeswitch-users(a)lists.freeswitch.org> *Sent*: Tue May 18 11:56:28 2010 *Subject*: Re: [Freeswitch-users] Contact Field HI Ken, Add the following line <param name="extension" value="10 Digit Number"/> Lloyd On Tue, May 18, 2010 at 11:40 AM, Ken Fulmer < kenfulmer(a)icstechnologysolutions.com> wrote: > We are connecting a FS gateway to PaeTec. They need the contact field to > be the same as the calling number. Using the trunk registration 10 digit > number doesn’t work for them. > > > > I added the following parameter to remove “gw + external” in the contact > field to the left of the @ sign: > > > > <param name="extension-in-contact" value="true"/> > > > > Is there something else I should add, so the contact field shows the 10 > digit calling number? > > > > Thanks, > > > > Ken Fulmer > > > > _______________________________________________ > 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 > >

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Siptapi and Freeswitch
by Peter P GMX 10 Sep '10

10 Sep '10

Anybody tried siptapi with freeswitch? http://sourceforge.net/projects/siptapi/ This may enable Click2Dial e.g. from Outlook to Freeswitch. So anybody has experience with that solution? Best regards Peter

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error loading module 'luasql.mysql'
by Adam Wilt 02 Sep '10

02 Sep '10

Hi, I followed the instructions in the Lua documentation for setting up luasql, but when I try to run my script I get: 2010-01-27 19:08:14.799250 [ERR] mod_lua.cpp:182 error loading module 'luasql.mysql' from file '/usr/local/lib/lua/5.1/luasql/mysql.so': /usr/local/lib/lua/5.1/luasql/mysql.so: cannot restore segment prot after reloc: Permission denied stack traceback: [C]: ? [C]: in function 'require' /usr/local/freeswitch/scripts/l.lua:2: in main chunk I'm running FreeSWITCH version 1.4 and luasql version 2.1.1. I changed the permissions for mysql.so and for my script to 777, so I'm not sure where the permission problem could be. I'd appreciate any suggestions. Thanks, Adam

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Stream audio file/live to multiple SIP endpoints with IP multicast
by Sluschny, Thomas 30 Aug '10

30 Aug '10

Hi, i want to stream audio (from file and live) from FreeSwitch to multiple SIP endpoints (these are proprietary). Because of bandwith limitations we have to use IP multicast for audio data. Now the question: how can i do this smartly in FreeSwitch? I have read some Wiki entries: - mod_esf: seams to use multicast, but i couldn't find usage information - mod_shout: streaming, but from internet ? MP3, but i want wav-files to stream - mod_local_stream: seams to play files in a loop, i want to play a whole file once. - mod_conference: seams not to support IP multicast Im searching for a simple way to send a SIP INVITE to all my endpoints and then stream audio (live from soundcard or from file) over IP multicast to all these endpoints. Thanks in advance, Thomas

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[contrib] a new java ESL inbound/outbound client
by david varnes 30 Aug '10

30 Aug '10

Hi all, I had a basic java inbound ESL client kicking around that I have used in a couple of small projects over the last year. I needed something a little more complete for a new project so I dusted it off and made it less incomplete. It still needs more work and testing, but it certainly is usable right now. I have tested it against FS 1.0.4 and latest trunk. I have put it in my contrib area in svn in hopes that some may find it useful: http://svn.freeswitch.org/svn/freeswitch/trunk/contrib/dvarnes/java/esl-cli… I would be interested in any feedback ... Features * Apache License (ASL) version 2 * Standalone Inbound client * Framework classes to easily create an Outbound socket client * based on Netty [1] nio library version 3.1.5.GA (previously was using Apache MINA, but this is easier) * logging via slf4j * only dependencies are slf4j-api and netty (both Apache licensed) * single jar which is a valid OSGi bundle * built using maven * eclipse projects * reasonable level of java docs Still todo * Docs * Simple example apps * .. more in TODO.txt in project root. There is no binary jar available right now since I don't know how/if I can put files up to file.freeswitch.org. In the meantime to build you need maven [2] installed. If you are unfamiliar with maven usage, I can post a simple howto. davidv [1] http://www.jboss.org/netty/downloads.html [2] http://maven.apache.org

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EventConsumer and multi-threading
by babak yakhchali 16 Jul '10

16 Jul '10

Hi I'm using mod_managed and I've used EventConsumer on a thread to listen for incomming events but whenever I wanna reload the .dll file of my script it complains about thread aborting. I used a dummy custom:dead event to end the thread but next time my dll loaded the custom:dead event is still in events and the tread ends. Is there any workaround to flush EventConsumer?? Thanks

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Diversion header
by Jonas Gauffin 23 Jun '10

23 Jun '10

Hello, I got a incoming call from my provider which goes to a javascript. The javascript transfers the call to an external destination (goes back through my provider). A diversion headers is added by FreeSWITCH on the outbound call. How can I remove it (or if not possible, how do I change it)? javascript: session.setCallerData("caller_id_number", aNumber); session.setVariable("gate_caller_id_number", aNumber); session.setVariable("gate_site_id", SITE_ID); session.execute("transfer", this.dtmfBuffer + " XML internal"); send 1179 bytes to udp/[130.244.Y.YY]:5060 at 16:13:16.752200: ------------------------------------------------------------------------ INVITE sip:024390510@sip-corporate.provider.com<sip%3A024390510(a)sip-corporate.provider.com>SIP/2.0 Via: SIP/2.0/UDP 212.247.XX.XX;rport;branch=z9hG4bKaa59XjFt2N2rS Max-Forwards: 67 From: "0236661XXX" <sip:0243795362@212.247.XX.XX>;tag=BHvytKtv47r2a To: <sip:024390510@sip-corporate.provider.com<sip%3A024390510(a)sip-corporate.provider.com> > Call-ID: 5a022f8f-a0b9-122d-779a-dbdeced2caa5 CSeq: 127654862 INVITE Contact: <sip:gw+dalaconnecttele2@212.247.XX.XX :5060;transport=udp;gw=dalaconnecttele2> User-Agent: Gateon Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 273 Diversion: 0 <sip:0@212.151.Y.Y ;user=phone>;reason=unconditional;counter=1;privacy=off X-FS-Support: update_display Remote-Party-ID: "0236661XXX" <sip:0243795362@212.247.XX.XX >;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1267513684 1267513685 IN IP4 212.247.XX.XX s=FreeSWITCH c=IN IP4 212.247.XX.XX t=0 0 m=audio 32712 RTP/AVP 8 3 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20

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span not defined error
by Neil Patel 21 Jun '10

21 Jun '10

Hi All, I am trying to dial out over my PRI, and am getting this error: 2010-05-16 01:01:21.452392 [CRIT] zap_io.c:1139 SPAN NOT DEFINED! 2010-05-16 01:01:21.452392 [ERR] mod_openzap.c:1154 No channels available 2010-05-16 01:01:21.452392 [ERR] switch_ivr_originate.c:2249 Cannot create outgoing channel of type [OpenZAP] cause: [NORMAL_CIRCUIT_CONGESTION] This is my openzap.conf: [span wanpipe smg_prid] name => smg_prid trunk_type =>e1 b-channel => 1:1-15 b-channel => 1:17-31 trunk_type =>e1 b-channel => 2:1-15 b-channel => 2:17-31 This is my openzap.conf.xml: <configuration name="openzap.conf" description="OpenZAP Configuration"> <settings> <param name="debug" value="0"/> </settings> <boost_spans> <span id="smg_prid"> <param name="dialplan" value="XML"/> <param name="context" value="default"/> </span> </boost_spans> <analog_spans> <span id="smg_prid"> <param name="dialplan" value="XML"/> <param name="context" value="default"/> </span> </analog_spans> </configuration> And here is the lua code I'm using to dial out: sessiondata = "OpenZAP/smg_prid/<num>" new_session = freeswitch.Session(sessiondata) What am I missing here? Thanks, Neil

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FS as Media Gateway Only
by Code Ghar 18 Jun '10

18 Jun '10

Is it possible -- and are there any case studies, practical experience, etc -- on deploying FreeSWITCH (FS) in this architecture: one server (FSSIP) handles SIP signaling only, and multiple servers (FSRTP1, FSRTP2, ..., FSRTPn) handle all media responsibilities? So when a call comes in, the SDP contains IP of, say FSRTP1, as media handler. For this to work, FSSIP would request FSRTPx for media resources for each new call and add its IP and port in SDP. The media servers/gateways would play IVR, etc.; collect DTMF and forward as appropriate to FSSIP; perform transcoding; etc.; all while FSSIP only deals with signaling. This way multiple servers could be deployed to handle media responsibilities and only a handful would be required for signaling. In future if there's a greater need for transcoding, etc. all you need to do is deploy a media server and not have to add servers for signaling. This idea came to me because I have come across two proprietary applications that do it this way. They have a SIP component and a media component. You can run both on the same physical machine or you can separate them out into multiple machines. Another way for this could be to integrate FS as a media component to another application's SIP component. A mix-and-match, so to speak. On the flip side, deploy FS as a SIP server and use media capabilities of some other hardware or software application. For example, FS handles signaling and use dedicated hardware for media. A good example of this is illustrated (somewhat) by an image on Sangoma's website: http://www.sangoma.com/assets/images/content/transcoding_diagram.jpg. Look at the "pooled transcoding". Is FS even designed to be this modular? If so, how can the aforementioned scenario(s) be achieved?

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