-- Executing [
[email protected]] Dial("SIP/270008-00000000", "SIP/DG/044NNNNNNN") in new stack
== Using SIP RTP CoS mark 5
Audio is at 2968
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 80.91.169.4:5060:
INVITE sip:
[email protected] SIP/2.0
Via: SIP/2.0/UDP 172.20.20.57:5060;branch=z9hG4bK31549707;rport
Max-Forwards: 70
From: "270008" <sip:
[email protected]>;tag=as5ca2e908
To: <sip:
[email protected]>
Contact: <sip:
[email protected]:5060>
Call-ID:
[email protected]
CSeq: 102 INVITE
User-Agent: Asterisk
Date: Wed, 24 Oct 2012 18:22:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 283
v=0
o=root 1905072958 1905072958 IN IP4 172.20.20.57
s=Asterisk PBX 1.8.17.0
c=IN IP4 172.20.20.57
t=0 0
m=audio 2968 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 SIP/DG/044NNNNNNN
<--- SIP read from UDP:80.91.169.4:5060 --->
SIP/2.0 407 authentication required
Allow: UPDATE,REFER,INFO
Call-ID:
[email protected]
Contact: <sip:
[email protected]:5060;user=phone>
CSeq: 102 INVITE
From: "270008" <sip:
[email protected]>;tag=as5ca2e908
Proxy-Authenticate: Digest realm="natsip.datagroup.com.ua",nonce="00558826104339a3040765271d48dd61",opaque="0054f7f356e1fa1",stale=false,algorithm=MD5
Server: Cirpack/v4.56 (gw_sip)
To: <sip:
[email protected]>;tag=03-08188-005588fe-5a23406d2
Via: SIP/2.0/UDP 172.20.20.57:5060;received=172.20.20.57;rport=1024;branch=z9hG4bK31549707
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Transmitting (NAT) to 80.91.169.4:5060:
ACK sip:
[email protected] SIP/2.0
Via: SIP/2.0/UDP 172.20.20.57:5060;branch=z9hG4bK31549707;rport
Max-Forwards: 70
From: "270008" <sip:
[email protected]>;tag=as5ca2e908
To: <sip:
[email protected]>;tag=03-08188-005588fe-5a23406d2
ontact: <sip:
[email protected]:5060>
Call-ID:
[email protected]
CSeq: 102 ACK
User-Agent: Asterisk
Content-Length: 0
---
[Oct 24 21] NOTICE[15557]: chan_sip.c:20492 handle_response_invite: Failed to authenticate on INVITE to '"270008" <sip:
[email protected]>;tag=as5ca2e908'
-- SIP/DG-00000001 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Auto fallthrough, channel 'SIP/270008-00000000' status is 'CONGESTION'
Really destroying SIP dialog '
[email protected]' Method: INVITE
