New setup query


#1

I have a problem and hope somebody can help.

I have installed asterisk (SVN-trunk-r7814) onto a fc2 box. All installed fine with no problems and also installed AMP (Version 1.10.010).

I have setup a couple of extensions and tried to connect to them using xlite. The phones register without a problem but as soon as I try and call the other phone the call sends the SIP INVITE and then I get a SIP 603 error (declined).

I have traced with Ethereal and cannot see why its failing. Can anybody help??

Thanks in advance.

Stuart.

Here is the sip debug dump from the server.
84.146.5.89 is the (changed for security) ip address of the client
asterisk1.propco.co.uk is the asterisk server.

[code]
— (0 headers 0 lines) Nat keepalive —
asterisk1*CLI>
<-- SIP read from 84.146.5.89:5060:
INVITE sip:3001@asterisk1.propco.co.uk SIP/2.0
v: SIP/2.0/UDP 84.146.5.89:5060;rport;branch=z9hG4bKA14D7123AD104B3CA37AF7415DE0 8ECC
f: asterisk1 sip:3001@asterisk1.propco.co.uk;tag=2967433968
t: sip:3001@asterisk1.propco.co.uk
m: sip:3001@84.146.5.89:5060
i: 5585886C-73B1-489A-B928-8F4DFB57B5EA@84.146.5.89
CSeq: 36569 INVITE
Max-Forwards: 70
c: application/sdp
User-Agent: X-Lite release 1103m
l: 291

v=0
o=3001 15890265 15890281 IN IP4 84.146.5.89
s=X-Lite
c=IN IP4 84.146.5.89
t=0 0
m=audio 8000 RTP/AVP 0 8 3 98 97 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

— (11 headers 13 lines)—
Using INVITE request as basis request - 5585886C-73B1-489A-B928-8F4DFB57B5EA@84. 146.5.89
Sending to 84.146.5.89 : 5060 (NAT)
Reliably Transmitting (no NAT) to 84.146.5.89:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 84.146.5.89:5060;rport;branch=z9hG4bKA14D7123AD104B3CA37AF7415D E08ECC;received=84.146.5.89
From: asterisk1 sip:3001@asterisk1.propco.co.uk;tag=2967433968
To: sip:3001@asterisk1.propco.co.uk;tag=as5da7db9a
Call-ID: 5585886C-73B1-489A-B928-8F4DFB57B5EA@84.146.5.89
CSeq: 36569 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:3001@81.187.216.234
Proxy-Authenticate: Digest realm=“asterisk”, nonce="3391c2ce"
Content-Length: 0


Scheduling destruction of call '5585886C-73B1-489A-B928-8F4DFB57B5EA@84.146.5.89 ’ in 15000 ms
Found user '3001’
asterisk1*CLI>
<-- SIP read from 84.146.5.89:5060:
ACK sip:3001@asterisk1.propco.co.uk SIP/2.0
v: SIP/2.0/UDP 84.146.5.89:5060;rport;branch=z9hG4bKA14D7123AD104B3CA37AF7415DE0 8ECC
f: asterisk1 sip:3001@asterisk1.propco.co.uk;tag=2967433968
t: sip:3001@asterisk1.propco.co.uk;tag=as5da7db9a
m: sip:3001@84.146.5.89:5060
i: 5585886C-73B1-489A-B928-8F4DFB57B5EA@84.146.5.89
CSeq: 36569 ACK
Max-Forwards: 70
l: 0

— (9 headers 0 lines)—
asterisk1*CLI>
<-- SIP read from 84.146.5.89:5060:
INVITE sip:3001@asterisk1.propco.co.uk SIP/2.0
v: SIP/2.0/UDP 84.146.5.89:5060;rport;branch=z9hG4bKE50A85F2B297408FAC771B5F2FF1 6DA2
f: asterisk1 sip:3001@asterisk1.propco.co.uk;tag=2967433968
t: sip:3001@asterisk1.propco.co.uk
m: sip:3001@84.146.5.89:5060
i: 5585886C-73B1-489A-B928-8F4DFB57B5EA@84.146.5.89
CSeq: 36570 INVITE
Proxy-Authorization: Digest username=“3001”,realm=“asterisk”,nonce=“3391c2ce”,re sponse=“be773de03cf1ff908a411bb848471dd9”,uri="sip:3001@asterisk1.propco.co.uk"
Max-Forwards: 70
c: application/sdp
User-Agent: X-Lite release 1103m
l: 291

v=0
o=3001 15890265 15890281 IN IP4 84.146.5.89
s=X-Lite
c=IN IP4 84.146.5.89
t=0 0
m=audio 8000 RTP/AVP 0 8 3 98 97 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

— (12 headers 13 lines)—
Using INVITE request as basis request - 5585886C-73B1-489A-B928-8F4DFB57B5EA@84. 146.5.89
Sending to 84.146.5.89 : 5060 (NAT)
Found user '3001’
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 98
Found RTP audio format 97
Found RTP audio format 101
Peer audio RTP is at port 84.146.5.89:8000
Found description format pcmu
Found description format pcma
Found description format gsm
Found description format iLBC
Found description format speex
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc )/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event) , combined - 0x1 (telephone-event)
Looking for 3001 in from-internal (domain asterisk1.propco.co.uk)
list_route: hop: sip:3001@84.146.5.89:5060
Transmitting (no NAT) to 84.146.5.89:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 84.146.5.89:5060;rport;branch=z9hG4bKE50A85F2B297408FAC771B5F2F F16DA2;received=84.146.5.89
From: asterisk1 sip:3001@asterisk1.propco.co.uk;tag=2967433968
To: sip:3001@asterisk1.propco.co.uk
Call-ID: 5585886C-73B1-489A-B928-8F4DFB57B5EA@84.146.5.89
CSeq: 36570 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:3001@81.187.216.234
Content-Length: 0


-- Executing Macro("SIP/3001-fca4", "exten-vm|3001|3001") in new stack
-- Executing Macro("SIP/3001-fca4", "user-callerid") in new stack
-- Executing Macro("SIP/3001-fca4", "hangupcall") in new stack
-- Executing ResetCDR("SIP/3001-fca4", "w") in new stack
-- Executing NoCDR("SIP/3001-fca4", "") in new stack
-- Executing Wait("SIP/3001-fca4", "5") in new stack
-- Executing Hangup("SIP/3001-fca4", "") in new stack

Reliably Transmitting (no NAT) to 84.146.5.89:5060:
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 84.146.5.89:5060;rport;branch=z9hG4bKE50A85F2B297408FAC771B5F2F F16DA2;received=84.146.5.89
From: asterisk1 sip:3001@asterisk1.propco.co.uk;tag=2967433968
To: sip:3001@asterisk1.propco.co.uk;tag=as1bf7d14f
Call-ID: 5585886C-73B1-489A-B928-8F4DFB57B5EA@84.146.5.89
CSeq: 36570 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:3001@81.187.216.234
Content-Length: 0


asterisk1*CLI>
<-- SIP read from 84.146.5.89:5060:
ACK sip:3001@asterisk1.propco.co.uk SIP/2.0
v: SIP/2.0/UDP 84.146.5.89:5060;rport;branch=z9hG4bKE50A85F2B297408FAC771B5F2FF1 6DA2
f: asterisk1 sip:3001@asterisk1.propco.co.uk;tag=2967433968
t: sip:3001@asterisk1.propco.co.uk;tag=as1bf7d14f
m: sip:3001@84.146.5.89:5060
i: 5585886C-73B1-489A-B928-8F4DFB57B5EA@84.146.5.89
CSeq: 36570 ACK
Max-Forwards: 70
l: 0

— (9 headers 0 lines)—
Destroying call ‘5585886C-73B1-489A-B928-8F4DFB57B5EA@84.146.5.89’

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