[Asterisk-Users] SIP and CVS Head

[Mark Johnson]

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 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

Anyone??  This is killing me!!!

Mark