[asterisk-users] Outside call not coming through
harry
ichverstehe at gmail.com
Sat Apr 26 10:51:05 CDT 2008
Screwed up really bad. This is the correct config and sip debug:
## 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 => _X.,Background(hello-world)
## sip debug (updated)
<--- 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=as2dae750f;lr=on>
Via: SIP/2.0/UDP 87.54.25.114;branch=z9hG4bK9588.5b1985c.0
Via: SIP/2.0/UDP 87.54.25.116:5060;branch=z9hG4bK3fdd7e37;rport=5060
From: "23864098" <sip:23864098 at 87.54.25.116>;tag=as2dae750f
To: <sip:36946811 at musimi.dk>
Contact: <sip:23864098 at 87.54.25.116>
Call-ID: 1204dee71075a8f97e6a598a4095e283 at 87.54.25.116
CSeq: 102 INVITE
User-Agent: no
Max-Forwards: 16
Date: Sat, 26 Apr 2008 15:44:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 215
v=0
o=root 19760 19760 IN IP4 87.54.25.116
s=session
c=IN IP4 87.54.25.116
t=0 0
m=audio 11114 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
<------------->
--- (15 headers 10 lines) ---
Sending to 87.54.25.114 : 5060 (no NAT)
Using INVITE request as basis request -
1204dee71075a8f97e6a598a4095e283 at 87.54.25.116
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=z9hG4bK9588.5b1985c.0;received=87.54.25.114
Via: SIP/2.0/UDP 87.54.25.116:5060;branch=z9hG4bK3fdd7e37;rport=5060
From: "23864098" <sip:23864098 at 87.54.25.116>;tag=as2dae750f
To: <sip:36946811 at musimi.dk>;tag=as3fc8c57f
Call-ID: 1204dee71075a8f97e6a598a4095e283 at 87.54.25.116
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5dc98c57"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog
'1204dee71075a8f97e6a598a4095e283 at 87.54.25.116' 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=z9hG4bK9588.5b1985c.0
From: "23864098" <sip:23864098 at 87.54.25.116>;tag=as2dae750f
Call-ID: 1204dee71075a8f97e6a598a4095e283 at 87.54.25.116
To: <sip:36946811 at musimi.dk>;tag=as3fc8c57f
CSeq: 102 ACK
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
More information about the asterisk-users
mailing list