I upgraded to CVS Head last night to help fix my SCCP problems and now my SIP installation is having issues. If I restart Asterisk, my SIP phones may take up to an hour to register correctly so I can place calls to them. They immediately go to voicemail as being busy. If I do a "sip reload" I get: -- Got SIP response 400 "Bad Request" back from xxx.xxx.xxx.1 -- Got SIP response 400 "Bad Request" back from xxx.xxx.xxx.2 -- Got SIP response 400 "Bad Request" back from xxx.xxx.xxx.3 -- Got SIP response 400 "Bad Request" back from xxx.xxx.xxx.4 <-- snip --> Here is some sip debug info: Answering/Requesting with root capability 0x4 (ulaw) Answering with capability 0x2 (gsm) Answering with capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 12 lines Reliably Transmitting (no NAT) to xxx.xxx.xxx.xxx:5060: INVITE sip SIP/2.0 Via: SIP/2.0/UDP 10.1.1.2:5060;branch=z9hG4bK3e4409aa From: "First Last" <sip:123@xxx.xxx.xxx.xxx>;tag=as77dd1f77 To: <sip> Contact: <sip:123@xxx.xxx.xxx.xxx> Call-ID: 797dd75b005c1b0017005aeb49f9e7ac@10.1.1.2 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Tue, 03 May 2005 06:42:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 255 v=0 o=root 13863 13863 IN IP4 xxx.xxx.xxx.xxx s=session c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=audio 12338 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 - - - - --- -- Called 122 asterisk*CLI> <-- SIP read from xxx.xxx.xxx.xxx:50634: SIP/2.0 400 Bad Request Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3e4409aa From: "First Last" <sip:@xxx.xxx.xxx.xxx>;tag=as77dd1f77 To: <sip> Call-ID: 797dd75b005c1b0017005aeb49f9e7ac@xxx.xxx.xxx.xxx Date: Tue, 03 May 2005 06:42:50 GMT CSeq: 102 INVITE Content-Length: 0 --- (8 headers 0 lines)--- -- Got SIP response 400 "Bad Request" back from xxx.xxx.xxx.xxx Transmitting (no NAT) to xxx.xxx.xxx.xxx:5060: ACK sip SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3e4409aa From: "Craig Deering" <sip:122@xxx.xxx.xxx.xxx>;tag=as77dd1f77 To: <sip> Contact: <sip:122@xxx.xxx.xxx.xxx> Call-ID: 797dd75b005c1b0017005aeb49f9e7ac@10.1.1.2 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 --- -- SIP/123-3428 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) HELP!!!!!!!!!!!!! Mark
Mark Johnson wrote:> I upgraded to CVS Head last night to help fix my SCCP problems and now > my SIP installation is having issues. If I restart Asterisk, my SIP > phones may take up to an hour to register correctly so I can place > calls to them. They immediately go to voicemail as being busy. If I > do a "sip reload" I get: > > -- Got SIP response 400 "Bad Request" back from xxx.xxx.xxx.1 > -- Got SIP response 400 "Bad Request" back from xxx.xxx.xxx.2 > -- Got SIP response 400 "Bad Request" back from xxx.xxx.xxx.3 > -- Got SIP response 400 "Bad Request" back from xxx.xxx.xxx.4 > <-- snip --> > > Here is some sip debug info: > > Answering/Requesting with root capability 0x4 (ulaw) > Answering with capability 0x2 (gsm) > Answering with capability 0x8 (alaw) > Answering with non-codec capability 0x1 (telephone-event) > 12 headers, 12 lines > Reliably Transmitting (no NAT) to xxx.xxx.xxx.xxx:5060: > INVITE sip SIP/2.0 > Via: SIP/2.0/UDP 10.1.1.2:5060;branch=z9hG4bK3e4409aa > From: "First Last" <sip:123@xxx.xxx.xxx.xxx>;tag=as77dd1f77 > To: <sip> > Contact: <sip:123@xxx.xxx.xxx.xxx> > Call-ID: 797dd75b005c1b0017005aeb49f9e7ac@10.1.1.2 > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Date: Tue, 03 May 2005 06:42:54 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Content-Type: application/sdp > Content-Length: 255 > > v=0 > o=root 13863 13863 IN IP4 xxx.xxx.xxx.xxx > s=session > c=IN IP4 xxx.xxx.xxx.xxx > t=0 0 > m=audio 12338 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 - - - - > > --- > -- Called 122 > asterisk*CLI> > <-- SIP read from xxx.xxx.xxx.xxx:50634: > SIP/2.0 400 Bad Request > Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3e4409aa > From: "First Last" <sip:@xxx.xxx.xxx.xxx>;tag=as77dd1f77 > To: <sip> > Call-ID: 797dd75b005c1b0017005aeb49f9e7ac@xxx.xxx.xxx.xxx > Date: Tue, 03 May 2005 06:42:50 GMT > CSeq: 102 INVITE > Content-Length: 0 > > > --- (8 headers 0 lines)--- > -- Got SIP response 400 "Bad Request" back from xxx.xxx.xxx.xxx > Transmitting (no NAT) to xxx.xxx.xxx.xxx:5060: > ACK sip SIP/2.0 > Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3e4409aa > From: "Craig Deering" <sip:122@xxx.xxx.xxx.xxx>;tag=as77dd1f77 > To: <sip> > Contact: <sip:122@xxx.xxx.xxx.xxx> > Call-ID: 797dd75b005c1b0017005aeb49f9e7ac@10.1.1.2 > CSeq: 102 ACK > User-Agent: Asterisk PBX > Content-Length: 0 > --- > -- SIP/123-3428 is circuit-busy > == Everyone is busy/congested at this time (1:0/1/0) > > > > HELP!!!!!!!!!!!!! > > MarkAnyone?? This is killing me!!! Mark