- FreeSWITCH-users - FreeSWITCH Mailing Lists

Conference call is still going.. JOIN~
by Brian West 17 Apr '13

17 Apr '13

Come on... LOL -- Brian West brian(a)freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST iNUM: +883 5100 1420 9266 ISN: 410*543

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Vestec ASR example apps
by Michael Collins 17 Apr '13

17 Apr '13

Hello all! Has anyone out there been using Vestec ASR? I need to gather some basic how-to stuff to create a Vestec section on our wiki. We are gearing up for the Vestec ASR app contest and I am hoping to lower the barrier to entry for entrants by having a really nice "get-you-started" wiki page. Please email me off-list if you have done anything with Vestec and are willing to share some examples. Thanks! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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FreeSWITCH Community Conference Call Today
by Michael Collins 17 Apr '13

17 Apr '13

Hello all! Today's community conference call agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2013_04_17 Today we will be using MS Visual Studio 2012 to build rapidly FreeSWITCH applications. We look forward to hearing from Dave Kompel on this discussion. -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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auto-stack sizing on linux appears to break voicemail email notification
by Nathan Neulinger 17 Apr '13

17 Apr '13

It seems to cause the child processes to inherit the small stack size, which results in sendmail segfaulting. Haven't dug through this yet. -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

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Re: [Freeswitch-users] PRI channels suspended
by Akash Deep Verma 17 Apr '13

17 Apr '13

Hi Ashish, PFA, the compiled file which I used. Thanks & Regards, - Akash Deep Verma On Wed, Apr 17, 2013 at 1:14 PM, < freeswitch-users-request(a)lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: PRI channels suspended (Ashish gautam) > 2. Re: Respect the request URI domain and To URI domain when > bridging (Jos? Luis Mill?n) > > > ---------- Forwarded message ---------- > From: Ashish gautam <ashish(a)nms.co.in> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Wed, 17 Apr 2013 12:33:23 +0530 > Subject: Re: [Freeswitch-users] PRI channels suspended > Hi Akash, > > I made these changes to the source and recompiled. Still the same this > happening. Channels 5 to 30 are suspended. > > Thanks. > > On Tue, Apr 16, 2013 at 4:39 PM, Akash Deep Verma <akashdeep.co(a)gmail.com>wrote: > >> Hi Ashish, >> >> This is a hack but you get what you need. >> >> This is because, whenever retry attempt is made 1 channel per pri get >> started, as there are 30 channels and retry attempts are 3 so the channels >> that freeswitch load is in between 3-6, so I increased retry attempt and >> complied it again. As I got rid of this error but it takes some time to >> load the pri channels (30*10 = 300seconds(5 seconds) max). >> >> Also, remember if you dial a call in between restarting phase you may >> face some error. To avoid this, dial call only after all pri channel get >> started. >> >> Here are the steps: >> >> In freeswitch source: >> Edit libs/freetdm/src/ftmod/ftmod_libpri/ftmod_libpri.h >> >> 1. Decrease timeout of channels. >> #define T316_TIMEOUT_MS_DEFAULT 30000 /* 30 sec */ >> >> change this to >> >> #define T316_TIMEOUT_MS_DEFAULT 10000 /* 10 sec */ >> >> 2. Increase attempts to 30. >> #define T316_ATTEMPT_LIMIT_DEFAULT 3 >> >> change this to >> >> #define T316_ATTEMPT_LIMIT_DEFAULT 30 >> >> 3. Increase maximum limit as according to default(set more than that of >> default) >> #define T316_ATTEMPT_LIMIT_MAX 10 >> >> change this to >> >> #define T316_ATTEMPT_LIMIT_MAX 35 >> >> Thanks & Regards, >> - >> Akash Deep Verma >> http://in.linkedin.com/in/akashdeep1 >> >> _________________________________________________________________________ >> 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 >> >> > > > ---------- Forwarded message ---------- > From: "José Luis Millán" <jmillan(a)aliax.net> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Wed, 17 Apr 2013 09:43:31 +0200 > Subject: Re: [Freeswitch-users] Respect the request URI domain and To URI > domain when bridging > Hi, > > > 2013/4/16 João Mesquita <jmesquita(a)freeswitch.org> > >> Ain't that what fs_path is for? I am not sure, maybe someone can chime-in >> here... >> > > Yes, 'fs_path' is the directive I was looking for. It specifies to SIP > proxy to which the message is being routed. > > Thank you. > > >> >> João Mesquita >> FreeSWITCH™ Solutions >> >> >> On Tue, Apr 16, 2013 at 5:10 PM, José Luis Millán <jmillan(a)aliax.net>wrote: >> >>> Hi, >>> >>> It was easy to define the sip_invite_full_to for legB given the >>> information exposed. I'm still unable to send the SIP request to a >>> different host that the one specified in the request URI. >>> >>> Can I send a request to a host that differs from the request URI domain >>> and To header URI domain of legB? If it is possible, any reference to the >>> doc will be much appreciated. >>> >>> Regards. >>> >>> >>> 2013/4/16 Michael Collins <msc(a)freeswitch.org> >>> >>>> For the most part I'd say yes. I think just having them on the wiki >>>> will be useful, even if it's just the self-explanatory part. >>>> -MC >>>> >>>> >>>> On Tue, Apr 16, 2013 at 11:56 AM, João Mesquita < >>>> jmesquita(a)freeswitch.org> wrote: >>>> >>>>> MC, will do, although they are pretty self-explanatory, ain't it? >>>>> >>>>> João Mesquita >>>>> FreeSWITCH™ Solutions >>>>> >>>>> >>>>> On Tue, Apr 16, 2013 at 2:29 PM, Michael Collins <msc(a)freeswitch.org>wrote: >>>>> >>>>>> J, >>>>>> >>>>>> Thanks for this information. It looks like none of these are on the >>>>>> wiki. Any volunteers out there willing to add these? Even if you don't know >>>>>> what they do you could add placeholders for them. See >>>>>> http://wiki.freeswitch.org/wiki/Variable_skeleton for details. >>>>>> >>>>>> Thanks, >>>>>> MC >>>>>> >>>>>> >>>>>> On Tue, Apr 16, 2013 at 9:32 AM, João Mesquita < >>>>>> jmesquita(a)freeswitch.org> wrote: >>>>>> >>>>>>> From sofia_glue.c: >>>>>>> >>>>>>> const char *invite_record_route = >>>>>>> switch_channel_get_variable(tech_pvt->channel, "sip_invite_record_route"); >>>>>>> const char *invite_route_uri = >>>>>>> switch_channel_get_variable(tech_pvt->channel, "sip_invite_route_uri"); >>>>>>> const char *invite_full_from = >>>>>>> switch_channel_get_variable(tech_pvt->channel, "sip_invite_full_from"); >>>>>>> const char *invite_full_to = >>>>>>> switch_channel_get_variable(tech_pvt->channel, "sip_invite_full_to"); >>>>>>> const char *handle_full_from = >>>>>>> switch_channel_get_variable(tech_pvt->channel, "sip_handle_full_from"); >>>>>>> const char *handle_full_to = >>>>>>> switch_channel_get_variable(tech_pvt->channel, "sip_handle_full_to"); >>>>>>> const char *force_full_from = >>>>>>> switch_channel_get_variable(tech_pvt->channel, "sip_force_full_from"); >>>>>>> const char *force_full_to = >>>>>>> switch_channel_get_variable(tech_pvt->channel, "sip_force_full_to"); >>>>>>> >>>>>>> >>>>>>> Glad to see you using FS José Luis! :-D Welcome. >>>>>>> >>>>>>> João Mesquita >>>>>>> >>>>>>> >>>>>>> On Tue, Apr 16, 2013 at 5:19 AM, José Luis Millán <jmillan(a)aliax.net >>>>>>> > wrote: >>>>>>> >>>>>>>> Let me fix an error: >>>>>>>> >>>>>>>> >>>>>>>> This is the message I would like to send to the IP 2.2.2.2 (my >>>>>>>> proxy) >>>>>>>> ``` >>>>>>>> INVITE sip:jmillan@1.2.3.4 SIP/2.0 Via: SIP/2.0/TCP >>>>>>>> bl23sl8lmku7.invalid;branch=z9hG4bK9420964 To: <sip:jmillan@1.2.3.4<sip%3Ad(a)1.2.3.4> >>>>>>>> >``` >>>>>>>> ``` >>>>>>>> >>>>>>>> Regards. >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> 2013/4/16 José Luis Millán <jmillan(a)aliax.net> >>>>>>>> >>>>>>>>> Hi all, >>>>>>>>> >>>>>>>>> Every call bridged though a specific 'gateway' does rewrite the >>>>>>>>> SIP request URI domain and To URI domain values to the gateway's 'proxy' >>>>>>>>> parameter value. >>>>>>>>> >>>>>>>>> I would like to respect such values and receive in my proxy the >>>>>>>>> SIP messages with their original request URI domain and To URI domain. >>>>>>>>> >>>>>>>>> This is, I would like to be able to send a SIP message with the >>>>>>>>> following request URI and To headers, to the IP address 2.2.2.2 >>>>>>>>> >>>>>>>>> ``` >>>>>>>>> INVITE sip:jmillan@1.2.3.4 SIP/2.0 Via: SIP/2.0/TCP >>>>>>>>> bl23sl8lmku7.invalid;branch=z9hG4bK9420964 To: <sip:d@1.2.3.4>``` >>>>>>>>> ``` >>>>>>>>> >>>>>>>>> Is it possible to handle this with FreeSwitch? >>>>>>>>> >>>>>>>>> I have followed the FS wiki but I've been unable to handle this. >>>>>>>>> >>>>>>>>> Thanks in advance. >>>>>>>>> >>>>>>>>> -- >>>>>>>>> José Luis Millán >>>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> -- >>>>>>>> José Luis Millán >>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> 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 >>>>>>>> >>>>>>>> >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> 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 >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Michael S Collins >>>>>> Twitter: @mercutioviz >>>>>> http://www.FreeSWITCH.org >>>>>> http://www.ClueCon.com >>>>>> http://www.OSTAG.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 >>>>>> >>>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> 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 >>>>> >>>>> >>>> >>>> >>>> -- >>>> Michael S Collins >>>> Twitter: @mercutioviz >>>> http://www.FreeSWITCH.org >>>> http://www.ClueCon.com >>>> http://www.OSTAG.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 >>>> >>>> >>> >>> >>> -- >>> José Luis Millán >>> >>> _________________________________________________________________________ >>> 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 >>> >>> >> >> _________________________________________________________________________ >> 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 >> >> > > > -- > José Luis Millán > > _______________________________________________ > 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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sofia profile determination
by Jimmy Chang 17 Apr '13

17 Apr '13

Hi, I have defined two SIP profile, external and internal. The external profile listen port 5080, and the internal profile listen port 5060. When a user in the same domain with FS dials telephone number using port 5080, FS determines to use internal profile. I want to use external profile when user dials port 5080 no matter where he or she is. Is it possible and what to do? Thanks. Jimmy

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Re: [Freeswitch-users] PRI channels suspended
by Akash Deep Verma 17 Apr '13

17 Apr '13

Hi Ashish, This is a hack but you get what you need. This is because, whenever retry attempt is made 1 channel per pri get started, as there are 30 channels and retry attempts are 3 so the channels that freeswitch load is in between 3-6, so I increased retry attempt and complied it again. As I got rid of this error but it takes some time to load the pri channels (30*10 = 300seconds(5 seconds) max). Also, remember if you dial a call in between restarting phase you may face some error. To avoid this, dial call only after all pri channel get started. Here are the steps: In freeswitch source: Edit libs/freetdm/src/ftmod/ftmod_libpri/ftmod_libpri.h 1. Decrease timeout of channels. #define T316_TIMEOUT_MS_DEFAULT 30000 /* 30 sec */ change this to #define T316_TIMEOUT_MS_DEFAULT 10000 /* 10 sec */ 2. Increase attempts to 30. #define T316_ATTEMPT_LIMIT_DEFAULT 3 change this to #define T316_ATTEMPT_LIMIT_DEFAULT 30 3. Increase maximum limit as according to default(set more than that of default) #define T316_ATTEMPT_LIMIT_MAX 10 change this to #define T316_ATTEMPT_LIMIT_MAX 35 Thanks & Regards, - Akash Deep Verma http://in.linkedin.com/in/akashdeep1

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plan break
by Jimmy Chang 17 Apr '13

17 Apr '13

Hi, I have a chatplan like this: <?xml version="1.0" encoding="utf-8"?> <include> <context name="default"> <extension name="testnumber"> <condition field="to" expression="^testnumber"/> <condition field="${cond(${_body} < 10 ? YES : NO)}" expression="^YES$"> <action application="reply" data="OK"/> </condition> </extension> <extension name="demo"> <condition field="to" expression="^(.*)$"> <action application="send" /> </condition> </extension> </context> </include> No matter what result the {cond(${_body} is, I don't want the plan flow go down to demo extension. What should I do? Thanks. Jimmy

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Testing WebRTC with JsSIP and OverSIP
by Henry Huang 17 Apr '13

17 Apr '13

Hi, Like the rest of the world, I am crazy about getting webrtc to work. Since FS does not yet do websocket. I have used JsSIP on Chrome and OverSIP as the sip proxy for websocket. I was able to call from Chrome with extension 1001 to another regular extension 1004 on FS with 2 way audio and dtmf working fine. The media is transporting directly between Chrome and FS. That is awesome. But when I try to call ext. 1001(JsSIP on Chrome) back from 1004. I am getting the following error from the javascript console, but I believe it's just reporting what the Chrome is not happy about. And if I try to pick up the call on Chrome, it would exit that session and kept the calling ringing. 1. JsSIP | RTC SESSION | unable to set local description jssip-devel.js:3313 <http://jssip.net/download/jssip-devel.js> 1. Failed to parse SessionDescription. m=audio 1 RTP/SAVPF Expects at least 4 fields. I have checked specifically about the SAVPF field I am sending from FS, and it has 4 field if I am not mistaken: INVITE sip:id2uaa1s@ehg4bjsdusqf.invalid;transport=ws SIP/2.0 Record-Route: <sip:08b1bd9007@72.1.46.750ddc3f-2196-1231-13a0-5254004b917b :10080;transport=ws;lr;ovid=f96ad63d> Record-Route: <sip:_NzIuMS40Ni4xMjJfNTA2MA--@oversip-public-ip :5060;transport=udp;lr;ovid=f96ad63d> Via: SIP/2.0/WS oversip-public-ip:10080;branch=z9hG4bK2b52650d8dfffc0bec5baf62029018cbb31b2e6c;rport Via: SIP/2.0/UDP freeswich-public-ip;branch=z9hG4bKme6eNg3jBvHpm;received=freeswich-public-ip;rport=5060 Max-Forwards: 10 From: "Extension 1004" <sip:1004@freeswich-public-ip>;tag=2SNgp6mXyc0Nr To: <sip:id2uaa1s@ehg4bjsdusqf.invalid;transport=ws;ov-ob=08b1bd9007> Call-ID: 750ddc3f-2196-1231-13a0-5254004b917b CSeq: 42742737 INVITE Contact: <sip:mod_sofia@freeswich-public-ip:5060> User-Agent: FreeSWITCH-mod_sofia/1.5.1b+git~20130408T160028Z~d50d43f788 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 886 X-Fs-Support: update_display,send_info Remote-Party-Id: "Extension 1004" <sip:1004@freeswich-public-ip >;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1366127964 1366127965 IN IP4 freeswich-public-ip s=FreeSWITCH c=IN IP4 freeswich-public-ip t=0 0 a=msid-semantic: WMS XbFAafgwqma0yucP2swUgWG29xewft33 *m=audio 29638 RTP/SAVPF 0 8 3 13* a=fingerprint:sha-256 3A:E4:E2:49:05:98:07:F6:A4:50:EA:8E:E8:04:C7:85:9E:AF:9E:42:E0:42:D2:17:96:EA:59:B6:FA:17:C4:24 a=rtcp-mux a=rtcp:29638 IN IP4 freeswich-public-ip a=ssrc:3312676386 cname:vzuvEASUNTLceOS7 a=ssrc:3312676386 msid:XbFAafgwqma0yucP2swUgWG29xewft33 a0 a=ssrc:3312676386 mslabel:XbFAafgwqma0yucP2swUgWG29xewft33 a=ssrc:3312676386 label:XbFAafgwqma0yucP2swUgWG29xewft33a0 a=ice-ufrag:6eSd1o51L9nRthLO a=ice-pwd:G5abwTWaDycHcUF9 a=candidate:7283942946 1 udp 659136 freeswich-public-ip 29638 typ host generation 0 a=candidate:7283942946 2 udp 659134 freeswich-public-ip 29638 typ host generation 0 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:p0HhxqXHnO/K86u3Qu1StqefUKtbbGVKA+Y8KSj1 a=ptime:20 At this point, I don't know how to troubleshoot further. I am posting this here to see if there is anyone running into similar issue and have a solution to this? Thanks, Henry

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Javadoc for esl.jar
by Marwan Idriss 16 Apr '13

16 Apr '13

Greeting, Is their any Javadoc for esl.jar ? Regards Marwan

4 3