[Asterisk-Users] SIP and CVS Head

[Mark Johnson]

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 at xxx.xxx.xxx.xxx>;tag=as77dd1f77
To: <sip>
Contact: <sip:123 at xxx.xxx.xxx.xxx>
Call-ID: 797dd75b005c1b0017005aeb49f9e7ac at 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 at 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 at xxx.xxx.xxx.xxx>;tag=as77dd1f77
To: <sip>
Contact: <sip:122 at xxx.xxx.xxx.xxx>
Call-ID: 797dd75b005c1b0017005aeb49f9e7ac at 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