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@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@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@pbx.itsp.com
REGISTER attempt 2 to 999@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@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@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@pbx.itsp.com>;tag=as3218ff14
To: <sip:999@pbx.itsp.com>
Call-ID: 3ec8bba250eab701464d5b1f4d2c51b9@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@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@pbx.itsp.com>;tag=as3218ff14
To: <sip:999@pbx.itsp.com>
Call-ID: 3ec8bba250eab701464d5b1f4d2c51b9@127.0.0.1
CSeq: 103 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:999@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@pbx.itsp.com>;tag=as3218ff14
To: <sip:999@pbx.itsp.com>;tag=as7d680d48
Call-ID: 3ec8bba250eab701464d5b1f4d2c51b9@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@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@123.183.86.231>;tag=as6e5334cf
To: <sip:pbx.itsp.com>
Contact: <sip:asterisk@123.183.86.231:5060>
Call-ID: 5f9dafd128057f97406f0a0736c0d878@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@123.183.86.231>;tag=as6e5334cf
To: <sip:pbx.itsp.com>;tag=as51d476cd
Call-ID: 5f9dafd128057f97406f0a0736c0d878@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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