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