On Friday 30 March 2007 04:02, Matt Putnam wrote: > I dont know if you have done this but run a sip show peers and make > sure that its registered with asterisk. Sounds like it is not > registering with asterisk which would allow you to call out but when > it tries to call you it dosent have an ip to contact you at.
ship show peers shows it thus roo*CLI> sip show peers Name/username Host Dyn Nat ACL Port Status chandler/chandler 192.168.0.10 D 5060 OK (6 ms) 1 sip peers [1 online , 0 offline] sip show registry shows nothing. With sip debug on I get the following as I start to make the call 13 headers, 12 lines Reliably Transmitting (no NAT) to 192.168.0.10:5060: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK1ff726a0;rport From: "Alan" <sip:[EMAIL PROTECTED]>;tag=as3e6d82c9 To: <sip:[EMAIL PROTECTED]:5060> Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 30 Mar 2007 06:07:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 263 v=0 o=root 29004 29004 IN IP4 192.168.0.20 s=session c=IN IP4 192.168.0.20 t=0 0 m=audio 19854 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- roo*CLI> <-- SIP read from 192.168.0.10:5060: SIP/2.0 100 Trying To: <sip:[EMAIL PROTECTED]:5060> From: "Alan" <sip:[EMAIL PROTECTED]>;tag=as3e6d82c9 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK1ff726a0 Server: roo.home Content-Length: 0 --- (8 headers 0 lines) --- roo*CLI> <-- SIP read from 192.168.0.10:5060: SIP/2.0 486 Busy Here To: <sip:[EMAIL PROTECTED]:5060>;tag=cf4213264eacc5ei0 From: "Alan" <sip:[EMAIL PROTECTED]>;tag=as3e6d82c9 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK1ff726a0 Server: roo.home Content-Length: 0 --- (8 headers 0 lines) --- Transmitting (no NAT) to 192.168.0.10:5060: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK1ff726a0;rport From: "Alan" <sip:[EMAIL PROTECTED]>;tag=as3e6d82c9 To: <sip:[EMAIL PROTECTED]:5060>;tag=cf4213264eacc5ei0 Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- dial status is BUSY Destroying call '[EMAIL PROTECTED]' > > On 3/29/07, Alan Chandler <[EMAIL PROTECTED]> wrote: > > I have a linksys SPA 3102 with a DECT phone connected into its > > Telephone port. > > > > It has been working, but something I've done (and I don't know > > what) means that now everytime asterisk tries to dial it, it says > > it is busy. > > > > I can make calls from it through asterisk > > > > I am at a complete loss to know what to try next to fix it. Any > > ideas? > > > > > > -- > > Alan Chandler > > http://www.chandlerfamily.org.uk > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users -- Alan Chandler http://www.chandlerfamily.org.uk _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
