[asterisk-users] SIP registration problem

[Dovid B]

I have seen this issue where there were internet connectivity issues. Asterisk registers every so often with the ITS. For some reason or another (it can be many reasons such as DNS, internet, ISP has issue etc). asterisk cant re-register so it keeps trying.
As far as the so context if you have a simple register line in sip.conf (such as register=> axe:by at sip_provider) then asterisk will tell the server that it is registering it with to send all calls to the s extension in your default context.

  ----- Original Message ----- 
  From: Michelle Dupuis 
  To: asterisk-users at lists.digium.com 
  Sent: Saturday, May 05, 2007 4:08 PM
  Subject: [asterisk-users] SIP registration problem


  I've reposted with a more meaningful subject - hopefully someone will reply....We have an Asterisk v1.2.16 box registering with an ITSP using SIP.  The registration succeeds, and is confirmed with SIP SHOW REGISTER.   However, we frequently (every few minutes) see this on our console:

  REGISTER attempt 1 to 999 at pbx.itsp.com 
  REGISTER attempt 2 to 999 at pbx.itsp.com 

  Any ideas what is going on?  In particular
  1.  What causes the two register attempt messages above?
  2.  Why is our asterisk box being associated with the "entryunauthorized" context, not the "entryinternal" context?  (See below)
  3.  Why is the contact "<sip:s at 123.183.86.231:5060>" in our SIP messages, why s@ anything?

  Thanks
  MD

  ------------------------------------------

  Contents of sip.conf at ITSP:

  [999]
  context=entryinternal   ; I know this context exists! This is the right context.
  type=friend
  username=999
  secret=1111
  callerid="Test" <999>
  host=dynamic                    
  nat=no                        
  canreinvite=no                
  allow=ulaw
  allow=alaw
  dtmfmode=rfc2833

  -------------------------------------------

  Console log from ITSP show strange SIP traffic:

  ---
  Scheduling destruction of call '3ec8bba250eab701464d5b1f4d2c51b9 at 127.0.0.1' in 15000 ms
  pbx*CLI> 
  pbx*CLI> 
  <-- SIP read from 123.183.86.231:5060: 
  REGISTER sip:pbx.itsp.com SIP/2.0
  Via: SIP/2.0/UDP 123.183.86.231:5060;branch=z9hG4bK1a8cee82;rport
  From: <sip:999 at pbx.itsp.com>;tag=as3218ff14
  To: <sip:999 at pbx.itsp.com>
  Call-ID: 3ec8bba250eab701464d5b1f4d2c51b9 at 127.0.0.1
  CSeq: 103 REGISTER
  User-Agent: Asterisk PBX
  Max-Forwards: 70
  Authorization: Digest username="999", realm="pbx.itsp.com", algorithm=MD5, uri="sip:pbx.itsp.com", nonce="5cec66c0", response="6451967016fc38f896efeb7247523fe1", opaque=""
  Expires: 120
  Contact: <sip:s at 123.183.86.231:5060>
  Event: registration
  Content-Length: 0

  --- (13 headers 0 lines) ---
  Using latest REGISTER request as basis request
  Sending to 123.183.86.231 : 5060 (NAT)
  Transmitting (no NAT) to 123.183.86.231:5060:
  SIP/2.0 100 Trying
  Via: SIP/2.0/UDP 123.183.86.231:5060;branch=z9hG4bK1a8cee82;received=123.183.86.231;rport=5060
  From: <sip:999 at pbx.itsp.com>;tag=as3218ff14
  To: <sip:999 at pbx.itsp.com>
  Call-ID: 3ec8bba250eab701464d5b1f4d2c51b9 at 127.0.0.1
  CSeq: 103 REGISTER
  User-Agent: Asterisk PBX
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  Contact: <sip:999 at 74.110.57.25>
  Content-Length: 0


  ---
  Transmitting (no NAT) to 123.183.86.231:5060:
  SIP/2.0 200 OK
  Via: SIP/2.0/UDP 123.183.86.231:5060;branch=z9hG4bK1a8cee82;received=123.183.86.231;rport=5060
  From: <sip:999 at pbx.itsp.com>;tag=as3218ff14
  To: <sip:999 at pbx.itsp.com>;tag=as7d680d48
  Call-ID: 3ec8bba250eab701464d5b1f4d2c51b9 at 127.0.0.1
  CSeq: 103 REGISTER
  User-Agent: Asterisk PBX
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  Expires: 120
  Contact: <sip:s at 123.183.86.231:5060>;expires=120
  Date: Fri, 04 May 2007 19:27:58 GMT
  ontent-Length: 0

  <-- SIP read from 123.183.86.231:5060: 
  OPTIONS sip:pbx.itsp.com SIP/2.0
  Via: SIP/2.0/UDP 123.183.86.231:5060;branch=z9hG4bK36c1df86;rport
  From: "asterisk" <sip:asterisk at 123.183.86.231>;tag=as6e5334cf
  To: <sip:pbx.itsp.com>
  Contact: <sip:asterisk at 123.183.86.231:5060>
  Call-ID: 5f9dafd128057f97406f0a0736c0d878 at 123.183.86.231
  CSeq: 102 OPTIONS
  User-Agent: Asterisk PBX
  Max-Forwards: 70
  Date: Fri, 04 May 2007 19:38:36 GMT
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  Content-Length: 0

  --- (12 headers 0 lines) ---
  Looking for s in entryunauthorized (domain pbx.itsp.com)
  Transmitting (no NAT) to 123.183.86.231:5060:
  SIP/2.0 200 OK
  Via: SIP/2.0/UDP 123.183.86.231:5060;branch=z9hG4bK36c1df86;received=123.183.86.231;rport=5060
  From: "asterisk" <sip:asterisk at 123.183.86.231>;tag=as6e5334cf
  To: <sip:pbx.itsp.com>;tag=as51d476cd
  Call-ID: 5f9dafd128057f97406f0a0736c0d878 at 123.183.86.231
  CSeq: 102 OPTIONS
  User-Agent: Asterisk PBX
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  Contact: <sip:74.110.57.25>
  Accept: application/sdp
  Content-Length: 0


   



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