June 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] FreeSWITCH adds WebRTC support to new 1.4 BETA.
by Ashley B 10 Jul '13

10 Jul '13

Gerald Were you able to resolve this issue? I am experiencing the exact same problem. Rgds Ashley. > > > ---------- Forwarded message ---------- > From: Gerald Weber <gerald.weber(a)besharp.at> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Thu, 27 Jun 2013 16:12:35 +0000 > Subject: Re: [Freeswitch-users] FreeSWITCH adds WebRTC support to new 1.4 > BETA. > > I've played a little more and tried the demo at www.webrtc.org, which > works to chrome should be fine.**** > > ** ** > > I called another ext (snom) instead of funny prompts in the switch:**** > > Audio from the headset mic -> snom is ok but not the other way round. The > headset dont get any audio...**** > > ** ** > > Fsctl debug is here: **** > > http://pastebin.freeswitch.org/21120**** > > ** ** > > Pcap trace done on the fs server with pcapsipdump is here (hope this is > correct):**** > > > https://dl.dropboxusercontent.com/u/21597352/20130627-175955-2005-2002-7363… > **** > > ** ** > > ** ** > > *Von:* freeswitch-users-bounces(a)lists.freeswitch.org [mailto: > freeswitch-users-bounces(a)lists.freeswitch.org] *Im Auftrag von *Anthony > Minessale > *Gesendet:* Donnerstag, 27. Juni 2013 17:27 > *An:* FreeSWITCH Users Help > *Betreff:* Re: [Freeswitch-users] FreeSWITCH adds WebRTC support to new > 1.4 BETA.**** > > ** ** > > try fsctl debug_level 10 and get a pcap for more info.**** > > ** ** > > It looks like before it was hairpinning then figuring out the local addr > and after you added localnet.auto it was right from the start, either way > it should have worked. **** > > ** ** > > On Thu, Jun 27, 2013 at 9:51 AM, Gerald Weber <gerald.weber(a)besharp.at> > wrote:**** > > Nope, windows firewall is disabled, also iptables on FreeSWITCH Testserver > is disabled. > Every other client is working (phonerlite, bria, xlite,etc) > > -----Ursprüngliche Nachricht----- > Von: freeswitch-users-bounces(a)lists.freeswitch.org [mailto: > freeswitch-users-bounces(a)lists.freeswitch.org] Im Auftrag von Kristian > Kielhofner > Gesendet: Donnerstag, 27. Juni 2013 16:34**** > > An: FreeSWITCH Users Help > Betreff: Re: [Freeswitch-users] FreeSWITCH adds WebRTC support to new 1.4 > BETA. > > The traces you provided look fine... Do you have a firewall running on > your test PC, by chance? > > On Thu, Jun 27, 2013 at 10:07 AM, Gerald Weber <gerald.weber(a)besharp.at> > wrote: > > Ps: i also included the output from jssip in the pastebin link > > > > > > > > Von: freeswitch-users-bounces(a)lists.freeswitch.org > > [mailto:freeswitch-users-bounces@lists.freeswitch.org] Im Auftrag von > > Anthony Minessale > > > > > > Gesendet: Donnerstag, 27. Juni 2013 15:16 > > An: FreeSWITCH Users Help > > Betreff: Re: [Freeswitch-users] FreeSWITCH adds WebRTC support to new > > 1.4 BETA. > > > > > > > > try adding the sofia profile param > > > > > > > > apply-candidate-acl=localnet.auto > > > > > > > > > > > > > > > > On Thu, Jun 27, 2013 at 4:45 AM, Gerald Weber > > <gerald.weber(a)besharp.at> > > wrote: > > > > I got rid of sip_secure_media and tried again, here is the trace: > > > > http://pastebin.freeswitch.org/21116 > > > > > > > > SIP User 2020 on Chrome calls Ext 20000 (the funny prompts thing) > > > > I'm no expert, but i guess there is something wrong with this**** > > > stun/run/ice thing.**** > > > > > > > > > Thanks, > > > > gw > > > > > > > > Von: freeswitch-users-bounces(a)lists.freeswitch.org > > [mailto:freeswitch-users-bounces@lists.freeswitch.org] Im Auftrag von > > Anthony Minessale > > Gesendet: Mittwoch, 26. Juni 2013 17:24 > > > > > > An: FreeSWITCH Users Help > > Betreff: Re: [Freeswitch-users] FreeSWITCH adds WebRTC support to new > > 1.4 BETA. > > > > > > > > You don't have to set anything in the dialplan. > > > > Get a debug console trace and post it to pastebin. > > > > > > > > > > > > On Wed, Jun 26, 2013 at 9:41 AM, Gerald Weber > > <gerald.weber(a)besharp.at> > > wrote: > > > > Hi, > > > > > > > > how did you get audio working ? > > > > I've also tried jssip and latest chrome and sip works, but there is no > > audio. Codecs like opus, ilbc and isac are installed. > > > > In the dialplan i set sip_secure_media=true but the speaker stays silent. > > I've tried ws and wss bindings but no luck. > > > > > > > > Thanks for any suggestions, > > > > gw > > > > > > > > Von: freeswitch-users-bounces(a)lists.freeswitch.org > > [mailto:freeswitch-users-bounces@lists.freeswitch.org] Im Auftrag von > > João Mesquita > > Gesendet: Mittwoch, 26. Juni 2013 15:28 > > An: FreeSWITCH Users Help > > Betreff: Re: [Freeswitch-users] FreeSWITCH adds WebRTC support to new > > 1.4 BETA. > > > > > > > > Whooray!!!! I've installed the new beta yesterday and tried it out > > with > > Chrome+JsSIP and it works fantastic!!! > > > > > > > > One question tho, is it possible/desirable to have a sofia profile to > > work only with webrtc and not with udp or tcp bindings? > > > > > > > > I haven't tried and I guess I should've before asking dumb questions > but.... > > > > > > João Mesquita**** > > > FreeSWITCHT Solutions**** > > > > > > > > > On Tue, Jun 25, 2013 at 7:42 PM, Anthony Minessale > > <anthony.minessale(a)gmail.com> wrote: > > > > Now also working in latest FireFox 22 > > > > > > > > sipml5 works with both chrome and FF > > > > > > > > https://webrtc.freeswitch.org/sipml5/ > > > > > > > > > > > > On Tue, Jun 25, 2013 at 2:04 AM, Michael Jerris <mike(a)jerris.com> wrote: > > > > Media is not handled by websockets. Web-sockets are a new transport > > for sip signaling. If an endpoint registers from a websocket > > transport, when we send it a call, we automatically send it an sdp > > appropriate for webrtc. If you want to send a call with an sdp for > > webrtc that isn't for a registration over websockets, just add > > media_webrtc=true var to the origination vars. If we receive an sdp > from webrtc, we will automatically handle it. > > > > Mike > > > > > > On Jun 25, 2013, at 12:37 AM, Jeff Leung <jleung(a)v10networks.ca> wrote: > > > >> Media wise, how would that be handled with web sockets? > >> > >>> -----Original Message----- > >>> From: freeswitch-users-bounces(a)lists.freeswitch.org > >>> [mailto:freeswitch- users-bounces(a)lists.freeswitch.org] On Behalf Of > >>> Dan Barua > >>> Sent: Monday, June 24, 2013 3:10 PM > >>> To: freeswitch-users(a)lists.freeswitch.org > >>> Subject: Re: [Freeswitch-users] FreeSWITCH adds WebRTC support to > >>> new > >>> 1.4 BETA. > >>> > >>> I asked the same question on Google+, Michael Jerris helpfully replied: > >>> > >>> you just need to setup FreeSWITCH as normal, and add 2 configuration > >>> params for the binding addresses for the websockets and secure > >>> websockets. the params are ws-binding and wss-binding format is > >>> ip:port or just :port if your using the same ip as sip-ip > >>> > >>> > >>> I haven't got round to trying it myself yet. > >>> > >>> -Dan > >>> > >>> > >>> > >>> > >>> -- > >>> View this message in context: http://freeswitch- > >>> users.2379917.n2.nabble.com/FreeSWITCH-adds-WebRTC-support-to-new- > >>> 1-4-BETA-tp7592138p7592141.html > >>> Sent from the freeswitch-users mailing list archive at Nabble.com. > >>> > >>> __________________________________________________________ > >>> _______________ > >>> 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-u > >>> sers > >>> 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-us > >> ers > >> 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-use > > rs**** > > > http://www.freeswitch.org > > > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale@hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888@conference.freeswitch.org > > googletalk:conf+888@conference.freeswitch.org > > pstn:+19193869900 > > > >**** > > > ______________________________________________________________________ > > ___ 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-use > > rs > > 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-use > > rs**** > > > http://www.freeswitch.org > > > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale@hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888@conference.freeswitch.org > > googletalk:conf+888@conference.freeswitch.org > > pstn:+19193869900 > > > >**** > > > ______________________________________________________________________ > > ___ 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-use > > rs**** > > > http://www.freeswitch.org > > > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale@hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888@conference.freeswitch.org > > googletalk:conf+888@conference.freeswitch.org > > pstn:+19193869900 > > > >**** > > > ______________________________________________________________________ > > ___ 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-use > > rs**** > > > http://www.freeswitch.org > > > > > > -- > Kristian Kielhofner > > _________________________________________________________________________ > 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**** > > > > **** > > ** ** > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale@hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org > googletalk:conf+888@conference.freeswitch.org > pstn:+19193869900 **** > > _______________________________________________ > 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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Cisco 7970G MWI + External SIP Voicemail Server
by Jeremy Moschner 09 Jul '13

09 Jul '13

Hi Everyone I am running an embedded FreeSWITCH installation (on OpenWRT, opkg version 1.0.7-2) and have a single Cisco 7970G that registers fine and can make and recieve calls (most of the time). However, one of my SIP providers (Internode) also offers a voicemail system (useful if I have my phone turned off and means I don't have to run a voicemail module on my installation). What I would like to find out is if it is possible for the 7970G (or indeed any other endpoint) to query/receive notification from Internode's voicemail system and display a MWI (or some sort of notification on the phone) when a message is there. Now, from my reading, I believe that this is accomplished through either SIP NOTIFY or SUBSCRIBE messages, but I am still quite confused about how it all works. I am unsure if FreeSWITCH needs to "relay" this information to the endpoint, or if is configurable via the MWI-Account parameter (ie. the directory user registers to receive these updates directly from the SIP server). I imagine I'll need to provide a bit more detail, so please let me know where to start.

2 4

FreeSWITCH not replying to SIP INVITE
by Christopher Hall 07 Jul '13

07 Jul '13

Hi, I'm new to FreeSWITCH lists and I'm not sure if this is the correct list for my post, any help will be much appreciated. I have a FreeSWITCH server configured to connect to a Draytel (draytel.org) SIP service. This works correctly and I can dial in and out using the service. I have an application running on a Windows Server that interfaces using event socket to the FreeSWITCH server. The intended use is that calls dialling in through the draytel service are automatically handled by this application. Again this works correctly. However, after a period of time which does not yet appear to be deterministic the FreeSWITCH server stops responding to INVITE requests from the draytel server. The FreeSWITCH server can still respond as I can dial using a local network VOIP phone and the application handles that call, but on the draytel connection there is no response. When FreeSWITCH stops responding I see absolutely nothing in the output logging information. I have run WireShark on the server that is running FreeSWITCH and I can see SIP INVITE requests being sent from draytel to the FreeSWITCH host but I don't see any kind of response. Help! Thanks Chris.

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Is FreeSwitch run on Windows 2008 R2 64?
by Albert Nguyen 05 Jul '13

05 Jul '13

Hi, I am trying to load freeswitch on to Windows 2008 R2 64 bit server but so far I can not get any windows versions of freeswitch to work on this os (latest 32 bits and 64 bits version from freeswitch web site). It give error message saying that it could not load the ..v8.dll and another dll file within the switch_loadable_module.c. Is Freeswitch not compatible to win2008R2 or am I missing something? Cheers, Albert

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att_xfer question
by Stuart Mills 05 Jul '13

05 Jul '13

Hi all, I have a question which is hopefully not too complicated. I’m playing with mid call transfer features, I have an extension in features.conf to do the MCT which works fine, a_leg connects to the b_leg, then b_leg transfer to destination c_leg and can talk. My question is this, is it then easy for b_leg to toggle the call between a_leg and c_leg, I take it it’s just another feature extension that bridges calls between the 2 legs, could it be that simple? Kind Regards, Stuart Mills

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Faxing problems if B leg offers T.38
by Rob Morin 04 Jul '13

04 Jul '13

Im having problems faxing if the B leg offers T.38. Several scenarios, but always the same result. First, the architecture FAX à ATA (T.38 optional) à FS à Carrier à Destination (Possibly another carrier (PSTN) and the fax machine) On my ATA, T.38 is enabled (Auto Detect). If the destination supports T.38, the Carrier will offer it to Freeswitch. If the Destination doesnt support T.38, the B leg doesnt offer it and the faxes go through. Ive set Freeswitch as: <action application="set" data="fax_enable_t38=true"/> <action application="export" data="t38_passthru=true"/> When thats the case, and the B leg offers T.38, Freeswitch passes through the offer and the ATA responds, 3 times. But the response never gets sent to the B leg, so it terminates the call after about 30 seconds. When I set Freeswitch as: <action application="set" data="fax_enable_t38=false"/> The B leg still offers T.38 if its other side is capable. Freeswitch doesnt pass the offer through and the B leg terminates the call after about 30 seconds. Im running on a recent git ( FreeSWITCH Version 1.0.head (git-86d757d 2011-05-17 22-51-47 -0500) ). Ive had no success attempting to configure FS as a gateway. Frankly, T.38 passthrough has had the most success, mostly when the other ends arent T.38 capable and it stays in G711u. Is there something else I need to do to enable T.38 in passthrough mode? Thank you, Rob Morin PS I can provide tcpdumps of this, or whatever else is necessary.

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Caller ID and Caller Name Issue
by Jagadish Thoutam 04 Jul '13

04 Jul '13

HI, i have a freeswitch as a VOIP Gateway that will handle the Inbound and Outbound calls with media proxy, here issue comes My Freeswith is Sending FreeSwitch as a callid Number.. and callid name as a Number i wondered why it is sending like that i even comment the effective acller id name and caller id but still same issue, can any one help on this. Version : FreeSWITCH Version 1.5.1b+git~20130423T194907Z~e1c325dcb5 (git e1c325d 2013-04-23 19:49:07Z) Thanks Jagan

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ptime transcoding does not pass rfc2833
by Chris Martineau 04 Jul '13

04 Jul '13

Hi, Does anyone know if this is a bug or a setting? If I use an incoming call ptime of say 40 on gsm to save client bandwidth and output to a carrier at 20 the rtp is transcoded to the smaller ptime fine but any rfc2833 dtmf signalling is not passed through. I have tried the pass_rfc2833 setting both on and off but it has no effect. I have done some web searches but can find nothing refering to this issue. Any help you could offer would be greatly appreciated. FreeSWITCH version: 1.0.head (git-58c3c3a 2011-11-22 18-22-57 -0600) Regards Chris

2 3

Secure B-Leg from PSTN call - how to?
by Peter Waldheim 03 Jul '13

03 Jul '13

Hi, I'm trying to secure (SRTP) the b-leg for a call coming in via pstn (pri/freetdm). For this I'm setting sip_secure_media and also trying sdp_secure_savp_only, but it seems freeswitch does not offer SAVP to my client (which in turn will refuse). SIP-to-SIP this works fine. Here the the portion of the log where I would have expected to see SAVP in the sdp: EXECUTE FreeTDM/1:5/21 bridge(sofia/external5090/21%10.1.1.12) 2013-06-28 09:20:10.800816 [DEBUG] switch_channel.c:1176 FreeTDM/1:5/21 EXPORTING[export_vars] [sip_secure_media]=[true] to event 2013-06-28 09:20:10.800816 [DEBUG] switch_channel.c:1176 FreeTDM/1:5/21 EXPORTING[export_vars] [dialed_extension]=[21] to event 2013-06-28 09:20:10.800816 [DEBUG] switch_channel.c:1176 FreeTDM/1:5/21 EXPORTING[export_vars] [sip_secure_media]=[true] to event 2013-06-28 09:20:10.800816 [DEBUG] switch_channel.c:1176 FreeTDM/1:5/21 EXPORTING[export_vars] [sdp_secure_savp_only]=[true] to event 2013-06-28 09:20:10.800816 [DEBUG] switch_ivr_originate.c:2050 Parsing global variables 2013-06-28 09:20:10.800816 [NOTICE] switch_channel.c:1030 New Channel sofia/external5090/21 [2b0bff7e-dfc3-11e2-b111-c96542f7174a] 2013-06-28 09:20:10.800816 [DEBUG] mod_sofia.c:4420 (sofia/external5090/21) State Change CS_NEW -> CS_INIT 2013-06-28 09:20:10.800816 [DEBUG] switch_core_session.c:1341 Send signal sofia/external5090/21 [BREAK] 2013-06-28 09:20:10.800816 [DEBUG] switch_core_state_machine.c:416 (sofia/external5090/21) Running State Change CS_INIT 2013-06-28 09:20:10.800816 [DEBUG] switch_core_state_machine.c:455 (sofia/external5090/21) State INIT 2013-06-28 09:20:10.800816 [DEBUG] mod_sofia.c:87 sofia/external5090/21 SOFIA INIT 2013-06-28 09:20:10.800816 [DEBUG] sofia_glue.c:1191 sip:21@client.ip.is.secret:49915;rinstance=e177370cb4131e9f;transport=tls Setting proxy route to sofia/external5090/21 2013-06-28 09:20:10.800816 [DEBUG] sofia_glue.c:1220 Local SDP: v=0 o=FreeSWITCH 1372384350 1372384351 IN IP4 my.ip.is.secret s=FreeSWITCH c=IN IP4 my.ip.is.secret t=0 0 m=audio 19660 RTP/AVP 8 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 20590 RTP/AVP 34 98 a=rtpmap:34 H263/90000 a=rtpmap:98 H264/90000 Any help with this would be greatly appreciated. Regards Peter

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Regarding Freeswicth Nat
by johnthan123 03 Jul '13

03 Jul '13

Hi All, I am really having Hard time with NAT, Can any one give me the steps witch already works for them, its Help for Many people who is having issue with NAT. Thanks in Advance.

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