hi thanks again!
I have captured the asterisk log and attacehd below. Does this give you any additional information to help me out? please note that the call get through, and the $$ amt messagae and the minutes information follow. I could hear a ring and the call aborted saying, the number is unavailable.
Thanks
Joe
v=0
o=root 8766 8766 IN IP4 192.168.1.66
s=session
c=IN IP4 192.168.1.66
t=0 0
m=audio 10770 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called vbuzzer.com/14165695266
trixbox1*CLI>
<--- SIP read from 209.47.41.24:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.66:5060;branch=z9hG4bK7f53061f;rport
From: "9057921343" <sip:
[email protected]>;tag=as61a71de3
To: <sip:
[email protected]>
Call-ID:
[email protected]
CSeq: 102 INVITE
Content-Length:0
<------------->
--- (7 headers 0 lines) ---
trixbox1*CLI>
<--- SIP read from 209.47.41.24:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.66:5060;branch=z9hG4bK7f53061f;rport
From: "9057921343" <sip:
[email protected]>;tag=as61a71de3
To: <sip:
[email protected]>
Call-ID:
[email protected]
CSeq: 102 INVITE
Content-Length:0
WWW-Authenticate:Digest realm="vbuzzer.com", nonce="4d5449784d5449314e6a6b334f413d3d", algorithm=MD5
Warning:392 209.47.41.24:80 "Noisy feedback tells: pid=93190 req_src_ip=192.168.1.66 req_src_port=64205 in_uri=sip:sip:
[email protected] out_uri=sip:sip:
[email protected] via_cnt==1"
<------------->
--- (9 headers 0 lines) ---
Transmitting (no NAT) to 209.47.41.24:5060:
ACK sip:
[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.66:5060;branch=z9hG4bK7f53061f;rport
From: "9057921343" <sip:
[email protected]>;tag=as61a71de3
To: <sip:
[email protected]>
Contact: <sip:
[email protected]>
Call-ID:
[email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
-- SIP/vbuzzer.com-08362478 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
Really destroying SIP dialog '
[email protected]' Method: INVITE
a2billing.php: file:Class.RateEngine.php - line:1129 - [FAILOVER K=0]:[ANSTIME=-DIALSTATUS=CONGESTION]
a2billing.php: file:Class.RateEngine.php - line:1079 - [K=0]:[ANSWEREDTIME=0-DIALSTATUS=CONGESTION]
a2billing.php: file:Class.RateEngine.php - line:1119 - FAILOVER app_callingcard: Dialing 'SIP/vbuzzer.com/14165695266|60|HRgrL(11850000:61000:30000)' with timeout of '11850'.
<snip>