<------------>
-- Executing Dial("SIP/0001-00000010", "SIP/sapo/289dddddd,50")
== Using SIP RTP CoS mark 5
Audio is at 82.154.203.14 port 11096
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 213.13.89.67:5070:
INVITE sip:
[email protected]:5070 SIP/2.0
Via: SIP/2.0/UDP 82.154.203.14:5060;branch=z9hG4bK688b6bbd;rport
Max-Forwards: 70
From: "0001" <sip:
[email protected]>;tag=as3d0bdbd6
To: <sip:
[email protected]:5070>
Contact: <sip:
[email protected]>
Call-ID:
[email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.9-2ubuntu2
Date: Fri, 08 Jul 2011 10:11:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 294
v=0
o=root 1142191950 1142191950 IN IP4 82.154.203.14
s=Asterisk PBX 1.6.2.9-2ubuntu2
c=IN IP4 82.154.203.14
t=0 0
m=audio 11096 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
-- Called sapo/289dddddd
<--- SIP read from UDP:213.13.89.67:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 82.154.203.14:5060;received=82.154.203.14;branch=z9hG4bK688b6bbd;rport=5060
From: "0001" <sip:
[email protected]>;tag=as3d0bdbd6
To: <sip:
[email protected]:5070>
Call-ID:
[email protected]
CSeq: 102 INVITE
<------------->
--- (6 headers 0 lines) ---
<--- SIP read from UDP:213.13.89.67:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 82.154.203.14:5060;received=82.154.203.14;branch=z9hG4bK688b6bbd;rport=5060
From: "0001" <sip:
[email protected]>;tag=as3d0bdbd6
To: <sip:
[email protected]:5070>;tag=aprqngfrt-0asfdk10000c6
Call-ID:
[email protected]
CSeq: 102 INVITE
<------------->
--- (6 headers 0 lines) ---
Transmitting (NAT) to 213.13.89.67:5060:
ACK sip:
[email protected]:5070 SIP/2.0
Via: SIP/2.0/UDP 82.154.203.14:5060;branch=z9hG4bK688b6bbd;rport
Max-Forwards: 70
From: "0001" <sip:
[email protected]>;tag=as3d0bdbd6
To: <sip:
[email protected]:5070>;tag=aprqngfrt-0asfdk10000c6
Contact: <sip:
[email protected]>
Call-ID:
[email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.9-2ubuntu2
Content-Length: 0
---
[Jul 8 11:11:46] WARNING[3426]: chan_sip.c:17865 handle_response_invite: Received response: "Forbidden" from '"0001" <sip:
[email protected]>;tag=as3d0bdbd6'
-- SIP/sapo-00000011 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Auto fallthrough, channel 'SIP/0001-00000010' status is 'CONGESTION'
<--- Reliably Transmitting (NAT) to 192.168.3.2:57574 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.3.2:57574;branch=z9hG4bK-d8754z-4d64327a32318571-1---d8754z-;received=192.168.3.2;rport=57574
From: "0001"<sip:
[email protected]:5060>;tag=f14c5d0b
To: <sip:
[email protected]:5060>;tag=as3e0ea8b7
Call-ID: Y2FmOWUzZGQ2OTA2YzkxZDRiMjE3ODgwOGQwMWVmNTA.
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.9-2ubuntu2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
X-Asterisk-HangupCause: Call Rejected
X-Asterisk-HangupCauseCode: 21