search for: df7jal23ls0d

...: CS0 verify_client : No verify_server : No And this is the relevant SIP data exchange (with public IP hidden): *CLI> <--- Received SIP request (2420 bytes) from WS:10.10.5.106:54411 ---> INVITE sip:204 at 10.10.5.49 SIP/2.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKiBw81ooU7ybSRbRqr8TOqWkMPQRdkMXo;rport From: "John Doe (101)"<sip:1060 at 10.10.5.49>;tag=heMv1HvlT7DeQxPxuqcq To: <sip:204 at 10.10.5.49> Contact: "John Doe (101)"<sip:1060 at df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=...

Hello, I'm trying to register a user using sipml5 on Asterisk 11. I followed the instructions here: http://thr3ads.net/asterisk-users/2012/08/1972342-Asterisk-Websockets I added transport=ws to my sip.conf file: [3002] username=3002 secret=XXXXXXXXX host=dynamic type=friend context=test disallow=all allow=g729 ;allow=all ; Allow codecs in order of preference allow=ilbc

...use there is no ICE support. You can see in de SIP trace below and the RTP trace below that there is no ICE support in Asterisk. [Aug 9 22:15:50] <--- SIP read from WS:178.119.146.190:36940 ---> [Aug 9 22:15:50] INVITE sip:419 at 178.18.90.230 SIP/2.0 [Aug 9 22:15:50] Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKk2KDePVLlTquEfJzIk7LCMdnOHHk4wn1;rport [Aug 9 22:15:50] From: "77"<sip:770000wrtc at 178.18.90.230>;tag=sRCvFQq3gUMqkl6TKTcl [Aug 9 22:15:50] To: <sip:419 at 178.18.90.230> [Aug 9 22:15:50] Contact: "77"<sips:770000wrtc at df7jal23ls0d.i...

...a NAT/ICE problem. Can anyone tell me then what is left that could be causing the 'no-audio' problem ?? SIP debug : [Aug 11 15:53:47] <--- SIP read from WS:178.119.146.190:60191 ---> [Aug 11 15:53:47] INVITE sip:419 at 178.18.90.230 SIP/2.0 [Aug 11 15:53:47] Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKSqKu6K3uxr3dOFdU5WAtPM5tKKA5yzAq;rport [Aug 11 15:53:47] From: <sip:770000wrtc at 178.18.90.230>;tag=SGUVL1LMdvxQrUfxprZJ [Aug 11 15:53:47] To: <sip:419 at 178.18.90.230> [Aug 11 15:53:47] Contact: <sips:770000wrtc at df7jal23ls0d.invalid;rtcweb-breaker=no;cli...

Hello thank you for your answer. I don't understand how there are many tutorials and examples on the web where every time the outcome is a working setup. Very strange I feel now after my personal experience with Asterisk 11 and webRTC. You also say Asterisk 13. How about Asterisk 12 then ?? Kind regards. On 10-08-16 21:53, Matt Fredrickson wrote: > I don't see an ice-ufrag or

...ly. I have attached the logs of Asterisk, if some one will get something useful Please reply on the same. Thanks and Regards, Anant == Using SIP VIDEO CoS mark 6 == Using SIP RTP CoS mark 5 [Oct 24 19:45:59] ERROR[3005][C-00000000]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("df7jal23ls0d.invalid", "(null)", ...): Name or service not known [Oct 24 19:45:59] WARNING[3005][C-00000000]: chan_sip.c:16067 __set_address_from_contact: Invalid host name in Contact: (can't resolve in DNS) : 'df7jal23ls0d.invalid' [Oct 24 19:45:59] ERROR[3005][C-00000000]: netsoc...

...vp=yes icesupport=yes directmedia=yes transport=wss,ws dtlsrekey=60 dtlsverify=no dtlscertfile=/etc/pki/tls/certs/rapidssl.crt dtlsprivatekey=/etc/pki/tls/private/rapidssl.key dtlssetup=actpass sip dump <--- SIP read from WS:2.2.2.2:8558 ---> INVITE sip:887 at ipbx SIP/2.0 Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKyhVsmJdAtwRvuOWz9BHiNo1DGcqp4grQ;rport From: "cervenka"<sip:vr1a882 at vhXXX.example.com>;tag=RDmpGm2Mubc5xQQ8NMli To: <sip:887 at ipbx> Contact: "cervenka"<sips:vr1a882 at df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=wss&g...

I see no indication of how to do this in sip.conf, and when I start Asterisk, it doesn't wait on port 80. Greetings, -- Juan Carlos Castro y Castro Instant Solutions - Telefonia Gerando Resultado http://www.instant.com.br Principais capitais: 4063-6100 Demais regi?es: (11)4063-6100