July 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Cause: ALLOTTED_TIMEOUT
by Lloyd Aloysius 10 Jul '13

10 Jul '13

Hi All When originate fail . The cli shows mod_dptools.c:3123 Originate Failed. Cause: ALLOTTED_TIMEOUT Which channel variables gives this cause value. I try several no luck bridge_hangup_cause last_bridge_hangup_cause hangup_cause_q850 Thanks LLoyd * *

2 2

outgoing INVITE without SDP in proxy_media mode
by 최현국 10 Jul '13

10 Jul '13

Hello freeswitch-users, I've encountered the following problem. Outgoing INVITE message from FS has no SDP in proxy_media mode. (in default mode(hard proxy?), everything is working fine.) test enviorment is the following: SIP client (CSipSample) ---> FS ---> other SIP server I've configured the following configuration at FS for proxy_media mode. [sip_profile] <param name="inbound-late-negotiation" value="true"/> [dialplan] <extension name="public_extensions"> <condition field="destination_number" expression="(.+)$"> <action application="set" data="proxy_media=true"/> <action application="export" data="codec_string=${ep_codec_string}"/> <action application="bridge" data="sofia/trunk/$1(a)yyy.yyy.yyy.yyy"/> </condition> </extension> [freeswitch version] ./freeswitch -version FreeSWITCH version: 1.5.1b+git~20130624T105435Z~275f2cf092 this is INVITE message from SIP client. INVITE sip:2002@test.com SIP/2.0 Via: SIP/2.0/UDP 192.168.2.19:41089;rport;branch=z9hG4bKPjs.6qQMadZveqMBGhgPt4VOA20eJmYm4- Max-Forwards: 70 From: "2004" <sip:2004@test.com>;tag=TfVNZT3HS7yRsl8UhZfcv4zYjaHHgFfq To: <sip:2002@test.com> Contact: "2004" <sip:2004@192.168.2.19:41089> Call-ID: DJBhgeByGZJ7WJLzwLNtPliL1mc5vM8A CSeq: 32048 INVITE Route: <sip:xxx.xxx.xxx.xxx:5080;transport=udp;lr> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: CSipSimple_SC-02C-10/r2272 Content-Type: application/sdp Content-Length: 340 v=0 o=- 3582370009 3582370009 IN IP4 192.168.2.19 s=pjmedia c=IN IP4 192.168.2.19 t=0 0 m=audio 4000 RTP/AVP 99 0 8 101 c=IN IP4 192.168.2.19 a=rtcp:4001 IN IP4 192.168.2.19 a=sendrecv a=rtpmap:99 SILK/24000 a=fmtp:99 useinbandfec=0 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 this is the outgoing INVITE message from FS. the message has no SDP. INVITE sip:2002@yyy.yyy.yyy.yyy SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5090;rport;branch=z9hG4bK1Djm23v8c17vF Route: <sip:cmf.nhn.jp> Max-Forwards: 69 From: "2004" <sip:2004@xxx.xxx.xxx.xxx>;tag=1K848K75U26Ne To: <sip:2002@yyy.yyy.yyy.yyy> Call-ID: 3b9cae19-6349-1231-80a1-3640b59986cf CSeq: 46354541 INVITE Contact: <sip:mod_sofia@xxx.xxx.xxx.xxx:5090> User-Agent: FreeSWITCH-mod_sofia/1.5.1b+git~20130624T105435Z~275f2cf092 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Privacy: none Content-Length: 0 X-FS-Support: update_display,send_info P-Asserted-Identity: "2004" <sip:2004@xxx.xxx.xxx.xxx> Am is missing some configuration for proxy_media? Thank you in advance. Regards, hkchoi

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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 > >

11 51

FS as SBC
by Nikolay Kondratyev 09 Jul '13

09 Jul '13

Hi All, I'm new to freeswitch and I'm investigating if i can use FS as SBC. I see that one can use FS as SBC for incoming and outgoing calls. But I was not able to find out if FS can be used for registration. I mean the subscriber from internet should send Register to FS, and FS should send Register to internal sip server. Is this possible? Another question is: what will happen with "contact parameters" in Invite and Register messages? Will FS pass unknown parameters through? I'm interested in particular in the following Contact field: Contact: <sip:+79980000004@10.169.10.64:5070;transport=udp>;+g.3gpp.icsi-ref="urn%3Au rn-7%3A3gpp-service.ims.icsi.mmtel,urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel .gsma.videocall";audio;video;q=0.8;+g.oma.sip-im;+g.oma.sip-im.large-message ;expires=3600 If FS will get Invite or Register from Internet with such Contact, is it possible to preserve parameters (namely +g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel,urn%3Aurn-7%3A3g pp-service.ims.icsi.mmtel.gsma.videocall";audio;video;q=0.8;+g.oma.sip-im;+g .oma.sip-im.large-message;expires=3600) on the internal call leg? Thanks in advance, Nikolay.

2 1

Video not secured on B-Leg
by Asier Benito 09 Jul '13

09 Jul '13

Hi, I'm new to Freeswitch and I'm trying to secure B-leg with SRTP from incoming call with video and audio via RTMP. To secure it, i'm using nolocal:sip_secure_media=true, the problem is that freeswitch only secures audio but not video, I'm using Freeswitch 1.2.11 and Linphone as B leg, here you have the SDP: 2013-07-09 08:35:07.210255 [DEBUG] switch_channel.c:1176 sofia/internal/ 1006(a)5.56.58.79 EXPORTING[export_vars] [sip_secure_media]=[true] to event 2013-07-09 08:35:07.210255 [DEBUG] switch_channel.c:1176 sofia/internal/ 1006(a)5.56.58.79 EXPORTING[export_vars] [sdp_secure_savp_only]=[true] to event 2013-07-09 08:35:07.210255 [DEBUG] sofia_glue.c:2678 Local SDP: v=0 o=FreeSWITCH 1373328169 1373328170 IN IP4 XX.XX.XX.XX s=FreeSWITCH c=IN IP4 XX.XX.XX.XX t=0 0 m=audio 30738 RTP/SAVP 99 100 101 13 a=rtpmap:99 SPEEX/16000 a=rtpmap:100 SPEEX/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:u9q805XwSn8pIzKZZbfxxGoOOQU6ztTbr7zCUx+q a=ptime:20 a=sendrecv m=video 24388 RTP/AVP 98 a=rtpmap:98 H264/90000 a=fmtp:98 packetization-mode=0; profile-level-id=42000a If I capture packets with tcpdump I can see how audio is secured with SRTP but not the video. Any idea about what is going wrong? Thanks in advance. Regards.

1 0

effective_caller_id_number / freetdm / libpri - not working
by Peter Waldheim 09 Jul '13

09 Jul '13

Hi, I have seen some posts about caller-id-problems recently, but not many resolutions. Also as this uses a PRI line, I'll post it extra. I'm trying to get user-specific caller-ids (or any caller-id for now really) to work on a PRI line. This has been working with a (very) old version of freeswitch and libpri, so if there is a change in the required configuration, a pointer to that would be very welcome of course. In the dialplan I set effective_caller_id_number (and _name just to be sure) right before bridging (also nowhere else): <action application="export" data="effective_caller_id_number=0xxxxxxx21"/> <action application="export" data="effective_caller_id_name=0xxxxxxx21"/> <action application="info"/> <action application="bridge" data="freetdm/1/a/$1" /> Info displays accordingly: variable_effective_caller_id_number: [0xxxxxxx21] variable_effective_caller_id_name: [0xxxxxxx21] variable_export_vars: [effective_caller_id_number,effective_caller_id_name] variable_current_application: [info] But the other side still sees the base-number (0xxxxxxx20) of the pri-trunk. Any ideas on what is going wrong or required to make this work would be greatly appreciated (or how to further analyze this). Thanks and regards Peter

1 0

Re: [Freeswitch-users] LRN DIP 302 Dialplan
by Andre Demattia 09 Jul '13

09 Jul '13

Hi, I'm checking to see if that is the best way to do the dip. So calling the dialplan then waiting for the on hang up complete then dialing again is the right way? Andre -----Original Message----- From: "Avi Marcus" <avi(a)avimarcus.net> Sent: ‎7/‎8/‎2013 5:07 PM To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> Subject: Re: [Freeswitch-users] LRN DIP 302 Dialplan Sure, do the dip first then do your LCR. Maybe I'm just not getting the question? -Avi On Mon, Jul 8, 2013 at 6:54 PM, <andretodd(a)verizon.net> wrote: HI, I need to look up a number to see if it's ported/ LCR before I bridge the call. What is the best way to go about doing this. Call comes in, I need to look up the LRN data via SIP 302. The results of that I need to hit my database for the LCR then I need to bridge the number. Would it be wise to dial the LRN server then on Hangup complete hit the database then dial the number? Thanks _________________________________________________________________________ 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

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Loss of DTMF in latest GIT with SRTP enabled?
by Yehavi Bourvine 09 Jul '13

09 Jul '13

Hello, I've noticed that DTMF does not work with the recent latest GIT versions when using SRTP. When I disable SRTP it works ok. Has anyone else noticed it? BTW, I've opened a Jira FS-5571 on it. Thanks, __Yehavi:

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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.

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LRN DIP 302 Dialplan
by andretodd＠verizon.net 08 Jul '13

08 Jul '13

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