[Asterisk-Users] iconnecthere 480 error: is there a workaround?

Sun Mar 23 19:38:25 MST 2003

Mark,

I believe there is: Here is the exchange using sip debug.

Gregg

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

bigcat*CLI> sip debug
SIP Debugging Enabled
    -- Executing Dial("Phone/phone0",
"SIP/777716176210060 at iconnecthere") in new stack
Interface is eth0
IP Address is 192.168.4.3
We're at 192.168.4.3 port 39998
Answering with preferred capability 4
Answering with preferred capability 8
Answering with preferred capability 1
Answering with preferred capability 2
10 headers, 10 lines
XXX Need to handle Retransmitting XXX:
INVITE sip:777716176210060 at 213.137.73.178 SIP/2.0
Via: SIP/2.0/UDP 192.168.4.3:5060;branch=0ed5cff0
From: "asterisk" <sip:asterisk at 192.168.4.3>;tag=67ff402e
Contact: <sip:asterisk at 192.168.4.3>
To: <sip:777716176210060 at 213.137.73.178>
Call-ID: 02193d28008cf0ac35ca524333abe63b at 192.168.4.3
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Content-Type: application/sdp
Content-Length: 202

v=0
o=root 13858 13858 IN IP4 192.168.4.3
s=session
c=IN IP4 192.168.4.3
t=0 0
m=audio 39998 RTP/AVP 0 8 4 3
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
 (no NAT) to 213.137.73.178:5060
    -- Called 777716176210060 at iconnecthere
Sip read: >
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.4.3:5060;branch=0ed5cff0
Call-ID: 02193d28008cf0ac35ca524333abe63b at 192.168.4.3
From: "asterisk" <sip:asterisk at 192.168.4.3>;tag=67ff402e
To: <sip:777716176210060 at 213.137.73.178>
CSeq: 102 INVITE
Content-Length: 0

7 headers, 0 lines
Sip read:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.4.3:5060;branch=0ed5cff0
Call-ID: 02193d28008cf0ac35ca524333abe63b at 192.168.4.3
From: "asterisk" <sip:asterisk at 192.168.4.3>;tag=67ff402e
To: <sip:777716176210060 at 213.137.73.178>;tag=4b857f0d-5ab788dc
CSeq: 102 INVITE
Proxy-Authenticate: DIGEST realm="deltathree.com", nonce="3e7e6ed6",
algorithm=MD5
Content-Length: 0

8 headers, 0 lines
XXX Need to handle Retransmitting XXX:
ACK sip:777716176210060 at 213.137.73.178 SIP/2.0
Via: SIP/2.0/UDP 192.168.4.3:5060;branch=0ed5cff0
From: "asterisk" <sip:asterisk at 192.168.4.3>;tag=67ff402e
To: <sip:777716176210060 at 213.137.73.178>;tag=4b857f0d-5ab788dc
Call-ID: 02193d28008cf0ac35ca524333abe63b at 192.168.4.3
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

 (no NAT) to 213.137.73.178:5060
We're at 192.168.4.3 port 39998
Answering with preferred capability 4
Answering with preferred capability 8
Answering with preferred capability 1
Answering with preferred capability 2
XXX Need to handle Retransmitting XXX:
INVITE sip:777716176210060 at 213.137.73.178 SIP/2.0
Via: SIP/2.0/UDP 192.168.4.3:5060;branch=0ed5cff0
From: "asterisk" <sip:asterisk at 192.168.4.3>;tag=67ff402e
Contact: <sip:asterisk at 192.168.4.3>
To: <sip:777716176210060 at 213.137.73.178>
Call-ID: 02193d28008cf0ac35ca524333abe63b at 192.168.4.3
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Proxy-Authorization: Digest username="85904362", realm="deltathree.com",
algorithm="MD5", uri="sip:777716176210060 at 213.137.73.178",
nonce="3e7e6ed6", response="b6bab0a7e409d10496cd6140e6d1e063"
Content-Type: application/sdp
Content-Length: 202

v=0
o=root 13837 13837 IN IP4 192.168.4.3
s=session
c=IN IP4 192.168.4.3
t=0 0
m=audio 39998 RTP/AVP 0 8 4 3
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
 (no NAT) to 213.137.73.178:5060
Sip read: >
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.4.3:5060;branch=0ed5cff0
Call-ID: 02193d28008cf0ac35ca524333abe63b at 192.168.4.3
From: "asterisk" <sip:asterisk at 192.168.4.3>;tag=67ff402e
To: <sip:777716176210060 at 213.137.73.178>
CSeq: 103 INVITE
Content-Length: 0

7 headers, 0 lines
Sip read: >
SIP/2.0 183 Session Progress
Via:  SIP/2.0/UDP 192.168.4.3:5060;branch=0ed5cff0;received=66.30.28.60
From: "asterisk" <sip:asterisk at 192.168.4.3>;tag=67ff402e
To: sip:777716176210060 at 213.137.73.178;tag=1FD16250-8D
Date: Mon, 24 Mar 2003 02:35:05 GMT
Call-ID: 02193d28008cf0ac35ca524333abe63b at 192.168.4.3
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 103 INVITE
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 180

v=0
o=CiscoSystemsSIP-GW-UserAgent 3626 3613 IN IP4 213.137.65.239
s=SIP Call
c=IN IP4 213.137.65.239
t=0 0
m=audio 18358 RTP/AVP 4
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no

12 headers, 8 lines
Sip read: >
SIP/2.0 200 OK
Via:  SIP/2.0/UDP 192.168.4.3:5060;branch=0ed5cff0;received=66.30.28.60
From: "asterisk" <sip:asterisk at 192.168.4.3>;tag=67ff402e
To: sip:777716176210060 at 213.137.73.178;tag=1FD16250-8D
Date: Mon, 24 Mar 2003 02:35:05 GMT
Call-ID: 02193d28008cf0ac35ca524333abe63b at 192.168.4.3
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 103 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER,
SUBSCRIBE, NOTIFY, INFO
Allow-Events: telephone-event
Contact: sip:777716176210060 at 213.137.65.239:5060
Record-Route: <sip:213.137.79.80>, <sip:213.137.79.78>,
<sip:777716176210060 at 213.137.73.178:5060;maddr=213.137.73.176>
Content-Type: application/sdp
Content-Length: 180

v=0
o=CiscoSystemsSIP-GW-UserAgent 3626 3613 IN IP4 213.137.65.239
s=SIP Call
c=IN IP4 213.137.65.239
t=0 0
m=audio 18358 RTP/AVP 4
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no

14 headers, 8 lines
XXX Need to handle Retransmitting XXX:
ACK sip:777716176210060 at 213.137.73.178 SIP/2.0
Via: SIP/2.0/UDP 192.168.4.3:5060;branch=0ed5cff0
From: "asterisk" <sip:asterisk at 192.168.4.3>;tag=67ff402e
To: <sip:777716176210060 at 213.137.73.178>;tag=4b857f0d-5ab788dc
Call-ID: 02193d28008cf0ac35ca524333abe63b at 192.168.4.3
CSeq: 103 ACK
User-Agent: Asterisk PBX
Content-Length: 0

 (no NAT) to 213.137.73.178:5060
    -- SIP/iconnecthere-08b0 answered Phone/phone0
XXX Need to handle Retransmitting XXX:
BYE sip:777716176210060 at 213.137.73.178 SIP/2.0
Via: SIP/2.0/UDP 192.168.4.3:5060;branch=0ed5cff0
From: "asterisk" <sip:asterisk at 192.168.4.3>;tag=67ff402e
To: <sip:777716176210060 at 213.137.73.178>;tag=4b857f0d-5ab788dc
Call-ID: 02193d28008cf0ac35ca524333abe63b at 192.168.4.3
CSeq: 104 BYE
User-Agent: Asterisk PBX
Content-Length: 0

 (no NAT) to 213.137.73.178:5060
  == Spawn extension (default, 16176210060, 1) exited non-zero on
'Phone/phone0'
Sip read: >
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.4.3:5060;branch=0ed5cff0
Call-ID: 02193d28008cf0ac35ca524333abe63b at 192.168.4.3
From: "asterisk" <sip:asterisk at 192.168.4.3>;tag=67ff402e
To: <sip:777716176210060 at 213.137.73.178>;tag=4b857f0d-5ab788dc
CSeq: 104 BYE
Content-Length: 0

7 headers, 0 lines
    -- Hungup 'Phone/phone0'
Sip read: >
SIP/2.0 408 Request Timeout
Call-ID: 02193d28008cf0ac35ca524333abe63b at 192.168.4.3
Via: SIP/2.0/UDP 192.168.4.3:5060;branch=0ed5cff0
From: "asterisk" <sip:asterisk at 192.168.4.3>;tag=67ff402e
To: <sip:777716176210060 at 213.137.73.178>;tag=4b857f0d-5ab788dc
CSeq: 104 BYE
Content-Length: 0

7 headers, 0 lines
Interface is eth0
IP Address is 192.168.4.3
bigcat*CLI>