rendered paste bodydebian*CLI> sip debug
SIP Debugging enabled
The 'sip debug' command is deprecated and will be removed in a future release. Please use 'sip set debug' instead.
-- Accepting AUTHENTICATED call from 10.1.99.102:
> requested format = ulaw,
> requested prefs = (ulaw|gsm),
> actual format = ulaw,
> host prefs = (ulaw|gsm),
> priority = mine
-- Executing [6044842871@from-totalconnect:1] Dial("IAX2/totalconnect-2", "SIP/6044842871") in new stack
Audio is at 10.1.99.111 port 17912
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 10.1.99.107:5060:
INVITE sip:6044842871@10.1.99.107;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.1.99.111:5060;branch=z9hG4bK6d35094f;rport
From: ""TOTAL CONNECT" <sip:6044840151@10.1.99.111>;tag=as4b899486
To: <sip:6044842871@10.1.99.107;transport=udp>
Contact: <sip:6044840151@10.1.99.111>
Call-ID: 7503ab36682b7b336e612e3a2bbe7750@10.1.99.111
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 12 May 2008 20:05:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 260
v=0
o=root 7042 7042 IN IP4 10.1.99.111
s=session
c=IN IP4 10.1.99.111
t=0 0
m=audio 17912 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 6044842871
Retransmitting #1 (NAT) to 10.1.99.107:5060:
INVITE sip:6044842871@10.1.99.107;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.1.99.111:5060;branch=z9hG4bK6d35094f;rport
From: ""TOTAL CONNECT" <sip:6044840151@10.1.99.111>;tag=as4b899486
To: <sip:6044842871@10.1.99.107;transport=udp>
Contact: <sip:6044840151@10.1.99.111>
Call-ID: 7503ab36682b7b336e612e3a2bbe7750@10.1.99.111
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 12 May 2008 20:05:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 260
v=0
o=root 7042 7042 IN IP4 10.1.99.111
s=session
c=IN IP4 10.1.99.111
t=0 0
m=audio 17912 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
---
Retransmitting #2 (NAT) to 10.1.99.107:5060:
INVITE sip:6044842871@10.1.99.107;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.1.99.111:5060;branch=z9hG4bK6d35094f;rport
From: ""TOTAL CONNECT" <sip:6044840151@10.1.99.111>;tag=as4b899486
To: <sip:6044842871@10.1.99.107;transport=udp>
Contact: <sip:6044840151@10.1.99.111>
Call-ID: 7503ab36682b7b336e612e3a2bbe7750@10.1.99.111
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 12 May 2008 20:05:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 260
v=0
o=root 7042 7042 IN IP4 10.1.99.111
s=session
c=IN IP4 10.1.99.111
t=0 0
m=audio 17912 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
---
[May 12 13:05:16] NOTICE[7050]: chan_sip.c:2887 auto_congest: Auto-congesting SIP/6044842871-081c9000
-- SIP/6044842871-081c9000 is circuit-busy
Scheduling destruction of SIP dialog '7503ab36682b7b336e612e3a2bbe7750@10.1.99.111' in 6400 ms (Method: INVITE)
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel 'IAX2/totalconnect-2' status is 'CONGESTION'