DNS work fine on this machine: ; <<>> DiG 9.7.2-P3 <<>> -t SRV _sip._udp.sipnet.ru ;; global options: +cmd ;; Got answer: ;; ->>HEADER<<- opcode: QUERY, status: NOERROR, id: 53284 ;; flags: qr rd ra; QUERY: 1, ANSWER: 1, AUTHORITY: 0, ADDITIONAL: 0
;; QUESTION SECTION: ;_sip._udp.sipnet.ru. IN SRV ;; ANSWER SECTION: _sip._udp.sipnet.ru. 172800 IN SRV 10 0 5060 sipnet.ru. ;; Query time: 123 msec ;; SERVER: 172.16.11.1#53(172.16.11.1) ;; WHEN: Mon Feb 14 20:09:00 2011 ;; MSG SIZE rcvd: 66 Rushan Shaymardanov 2011/2/14 Fellipe ... <[email protected]>: > Maybe a DNS problem? > Try to ping from another machine your HOSTNAME. > Best regards, > Fellipe > >> Date: Mon, 14 Feb 2011 15:29:08 +0500 >> From: [email protected] >> To: [email protected] >> Subject: [asterisk-users] Problems with realtime sip >> >> I have a problem using asterisk 1.6 with realtime sip. >> >> When I add sip channel (my sip provider) to asterisk using realtime >> sip (http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip), >> incoming calls don't work for me. >> In asterisk CLI I get message: >> >> NOTICE[19805]: chan_sip.c:21250 handle_request_invite: Sending fake >> auth rejection for device "test" >> <sip:[email protected]>;tag=as0af02b0c. >> >> This is what happens in case I use hostname as a value of host >> parameter in sip table. When I use IP address instead of hostname, >> everything works fine. >> From the other hand, when I setup the same sip channel using sip.conf >> file, everything works fine as well, even with hostname as host >> parameter. >> >> Rushan Shaymardanov >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
