Hi All,
I’ve installed asterisk on opensuse 11.1. It’s up and running. I wanted to create sip test dial plan but I’m unable to register local softphone - so asterisk and softphone are on the same machine. I was wondering if this is a special case - is there something particular to be done on the both sides?
Relevant sip.conf lines:
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
context = from-sip ; Default for incoming calls
callerid=NoCallID
srvlookup=no
dtmfmode=rfc2833
allow=all
nat=no
[1052]
type=friend
username=1052
secret=ruka
host=dynamic
mailbox=1052
canreinvite=yes
dtmfmode=rfc2833
group=1
callgroup=1
pickupgroup=1
and here’s sip debug log:
<--- SIP read from 127.0.0.1:5061 --->
REGISTER sip:127.0.0.1 SIP/2.0
CSeq: 1 REGISTER
Via: SIP/2.0/UDP 127.0.0.1:5061;branch=z9hG4bK42352c3d-1c50-de11-9bb8-0015f201d7a2;rport
User-Agent: Ekiga/3.0.1
From: <sip:1052@127.0.0.1>
Call-ID: 329b2b3d-1c50-de11-9bb8-0015f201d7a2@linux
To: <sip:1052@127.0.0.1>
Contact: <sip:1052@127.0.0.1:5061>;q=1, <sip:1052@192.168.1.2>;q=0.500
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING
Expires: 36000
Content-Length: 0
Max-Forwards: 70
<------------->
--- (12 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 127.0.0.1 : 5061 (NAT)
<--- Transmitting (no NAT) to 127.0.0.1:5061 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 127.0.0.1:5061;branch=z9hG4bK42352c3d-1c50-de11-9bb8-0015f201d7a2;received=127.0.0.1;rport=5061
From: <sip:1052@127.0.0.1>
To: <sip:1052@127.0.0.1>
Call-ID: 329b2b3d-1c50-de11-9bb8-0015f201d7a2@linux
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
<--- Transmitting (no NAT) to 127.0.0.1:5061 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 127.0.0.1:5061;branch=z9hG4bK42352c3d-1c50-de11-9bb8-0015f201d7a2;received=127.0.0.1;rport=5061
From: <sip:1052@127.0.0.1>
To: <sip:1052@127.0.0.1>;tag=as1a677423
Call-ID: 329b2b3d-1c50-de11-9bb8-0015f201d7a2@linux
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3008d53a"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '329b2b3d-1c50-de11-9bb8-0015f201d7a2@linux' in 32000 ms (Method: REGISTER)
linux*CLI>
<--- SIP read from 127.0.0.1:5061 --->
REGISTER sip:127.0.0.1 SIP/2.0
CSeq: 1 REGISTER
Via: SIP/2.0/UDP 127.0.0.1:5061;branch=z9hG4bK42352c3d-1c50-de11-9bb8-0015f201d7a2;rport
User-Agent: Ekiga/3.0.1
From: <sip:1052@127.0.0.1>
Call-ID: 329b2b3d-1c50-de11-9bb8-0015f201d7a2@linux
To: <sip:1052@127.0.0.1>
Contact: <sip:1052@127.0.0.1:5061>;q=1, <sip:1052@192.168.1.2>;q=0.500
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING
Expires: 36000
Content-Length: 0
Max-Forwards: 70
<------------->
--- (12 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 127.0.0.1 : 5061 (NAT)
<--- Transmitting (no NAT) to 127.0.0.1:5061 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 127.0.0.1:5061;branch=z9hG4bK42352c3d-1c50-de11-9bb8-0015f201d7a2;received=127.0.0.1;rport=5061
From: <sip:1052@127.0.0.1>
To: <sip:1052@127.0.0.1>
Call-ID: 329b2b3d-1c50-de11-9bb8-0015f201d7a2@linux
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
<--- Transmitting (no NAT) to 127.0.0.1:5061 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 127.0.0.1:5061;branch=z9hG4bK42352c3d-1c50-de11-9bb8-0015f201d7a2;received=127.0.0.1;rport=5061
From: <sip:1052@127.0.0.1>
To: <sip:1052@127.0.0.1>;tag=as1a677423
Call-ID: 329b2b3d-1c50-de11-9bb8-0015f201d7a2@linux
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3008d53a"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '329b2b3d-1c50-de11-9bb8-0015f201d7a2@linux' in 32000 ms (Method: REGISTER)
linux*CLI>
<--- SIP read from 127.0.0.1:5061 --->
REGISTER sip:127.0.0.1 SIP/2.0
CSeq: 1 REGISTER
Via: SIP/2.0/UDP 127.0.0.1:5061;branch=z9hG4bK42352c3d-1c50-de11-9bb8-0015f201d7a2;rport
User-Agent: Ekiga/3.0.1
From: <sip:1052@127.0.0.1>
Call-ID: 329b2b3d-1c50-de11-9bb8-0015f201d7a2@linux
To: <sip:1052@127.0.0.1>
Contact: <sip:1052@127.0.0.1:5061>;q=1, <sip:1052@192.168.1.2>;q=0.500
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING
Expires: 36000
Content-Length: 0
Max-Forwards: 70
<------------->
--- (12 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 127.0.0.1 : 5061 (NAT)
<--- Transmitting (no NAT) to 127.0.0.1:5061 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 127.0.0.1:5061;branch=z9hG4bK42352c3d-1c50-de11-9bb8-0015f201d7a2;received=127.0.0.1;rport=5061
From: <sip:1052@127.0.0.1>
To: <sip:1052@127.0.0.1>
Call-ID: 329b2b3d-1c50-de11-9bb8-0015f201d7a2@linux
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
<--- Transmitting (no NAT) to 127.0.0.1:5061 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 127.0.0.1:5061;branch=z9hG4bK42352c3d-1c50-de11-9bb8-0015f201d7a2;received=127.0.0.1;rport=5061
From: <sip:1052@127.0.0.1>
To: <sip:1052@127.0.0.1>;tag=as1a677423
Call-ID: 329b2b3d-1c50-de11-9bb8-0015f201d7a2@linux
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3008d53a"
Content-Length: 0
...
Seems like softphone (Ekiga) is not getting responses from asterisk - since it doesn’t send authorization credentials but keeps repeating REGISTER and ignores 401.
Any ideas what could be wrong?
Thanks,
Milan