When i try to call '36946811' from the outside the call gets through,
but is rejected and the sound file is not played, this is my conf and
sip debug output:
## sip.conf
[general]
context=incoming
register => 36946811:L0sebitch at musimi.dk/1234
port=5060
bindaddr=0.0.0.0
srvlookup=yes
## extensions.conf
[incoming]
exten => 36946811,1,Background(hello-world)
## sip debug
*CLI>
<--- SIP read from 87.54.25.114:5060 --->
INVITE sip:1234 at 67.207.147.205 SIP/2.0
Record-Route: <sip:87.54.25.114;ftag=688c7f1d;lr=on>
Via: SIP/2.0/UDP 87.54.25.114;branch=z9hG4bK70b1.34ff9f57.0
Via: SIP/2.0/UDP
192.168.2.5:5060;received=62.107.1.48;branch=z9hG4bK-d8754z-c59c0dcc191940e4-1---d8754z-;rport=5060
Max-Forwards: 16
Contact: <sip:36946731 at 62.107.1.48:5060>
To: <sip:36946811 at musimi.dk>
From: "Harry"<sip:36946731 at
musimi.dk>;transport=UDP;tag=688c7f1d
Call-ID: NTRmMjg4ZGFlZDU2MDUxOTAxZjQ0OGNiYTFiOTMxOGQ.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Type: application/sdp
User-Agent: Zoiper rev.417
Content-Length: 311
v=0
o=Z 0 0 IN IP4 192.168.2.5
s=Z
c=IN IP4 192.168.2.5
t=0 0
m=audio 8000 RTP/AVP 3 110 97 8 0 101
a=fmtp:97 mode=30
a=fmtp:101 0-15
a=rtpmap:3 GSM/8000
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=direction:active
<------------->
--- (14 headers 15 lines) ---
Sending to 87.54.25.114 : 5060 (no NAT)
Using INVITE request as basis request -
NTRmMjg4ZGFlZDU2MDUxOTAxZjQ0OGNiYTFiOTMxOGQ.
Found peer 'musimi'
<--- Reliably Transmitting (NAT) to 87.54.25.114:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
87.54.25.114;branch=z9hG4bK70b1.34ff9f57.0;received=87.54.25.114
Via: SIP/2.0/UDP
192.168.2.5:5060;received=62.107.1.48;branch=z9hG4bK-d8754z-c59c0dcc191940e4-1---d8754z-;rport=5060
From: "Harry"<sip:36946731 at
musimi.dk>;transport=UDP;tag=688c7f1d
To: <sip:36946811 at musimi.dk>;tag=as0f99b309
Call-ID: NTRmMjg4ZGFlZDU2MDUxOTAxZjQ0OGNiYTFiOTMxOGQ.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="35c07307"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog
'NTRmMjg4ZGFlZDU2MDUxOTAxZjQ0OGNiYTFiOTMxOGQ.' in 32000 ms (Method:
INVITE)
<--- SIP read from 87.54.25.114:5060 --->
ACK sip:1234 at 67.207.147.205 SIP/2.0
Via: SIP/2.0/UDP 87.54.25.114;branch=z9hG4bK70b1.34ff9f57.0
From: "Harry"<sip:36946731 at
musimi.dk>;transport=UDP;tag=688c7f1d
Call-ID: NTRmMjg4ZGFlZDU2MDUxOTAxZjQ0OGNiYTFiOTMxOGQ.
To: <sip:36946811 at musimi.dk>;tag=as0f99b309
CSeq: 2 ACK
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Reliably Transmitting (NAT) to 62.107.1.48:5060:
OPTIONS sip:lolz at 192.168.2.5:5060;rinstance=d815b062f3a40a5e SIP/2.0
Via: SIP/2.0/UDP 67.207.147.205:5060;branch=z9hG4bK1945b111;rport
From: "asterisk" <sip:asterisk at 67.207.147.205>;tag=as00c1a604
To: <sip:lolz at 192.168.2.5:5060;rinstance=d815b062f3a40a5e>
Contact: <sip:asterisk at 67.207.147.205>
Call-ID: 4e2f2293727eb63c6d175d8726450a84 at 67.207.147.205
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 26 Apr 2008 14:46:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
---
<--- SIP read from 62.107.1.48:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 67.207.147.205:5060;branch=z9hG4bK1945b111;rport=5060
Contact: <sip:192.168.2.5:5060>
To: <sip:lolz at 192.168.2.5:5060;rinstance=d815b062f3a40a5e>;tag=d4846e53
From: "asterisk"<sip:asterisk at 67.207.147.205>;tag=as00c1a604
Call-ID: 4e2f2293727eb63c6d175d8726450a84 at 67.207.147.205
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, REFER, MESSAGE, OPTIONS
User-Agent: Zoiper rev.417
Allow-Events: message-summary
Content-Length: 0
<------------->
--- (13 headers 0 lines) ---
Really destroying SIP dialog
'4e2f2293727eb63c6d175d8726450a84 at 67.207.147.205' Method: OPTIONS