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-- Executing Dial("SIP/3000-fca3", "SIP/wakacall/001xxxxxxxxxx") in new stack
We're at xx.xx.xx.xx port 19648
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 11 lines
Reliably Transmitting (NAT) to 217.72.243.59:5060:
INVITE sip:001xxxxxxxxxx@sip.checkcdr.com SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK4a700c39;rport
From: "alen" <sip:3000@xx.xx.xx.xx>;tag=as7c9a5ec9
To: <sip:001xxxxxxxxxx@sip.checkcdr.com>
Contact: <sip:3000@xx.xx.xx.xx>
Call-ID: 5a2078725c832f0d06a6fbbf00566558 [!at] xx.xx.xx.xx (replace the [!at] with a @)
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 04 Jan 2000 23:37:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 238
v=0
o=root 24513 24513 IN IP4 xx.xx.xx.xx
s=session
c=IN IP4 xx.xx.xx.xx
t=0 0
m=audio 19648 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 - - - -
---
-- Called wakacall/001xxxxxxxxxx
Retransmitting #1 (NAT) to 217.72.243.59:5060:
INVITE sip:001xxxxxxxxxx@sip.checkcdr.com SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK4a700c39;rport
From: "alen" <sip:3000@xx.xx.xx.xx>;tag=as7c9a5ec9
To: <sip:001xxxxxxxxxx@sip.checkcdr.com>
Contact: <sip:3000@xx.xx.xx.xx>
Call-ID: 5a2078725c832f0d06a6fbbf00566558 [!at] xx.xx.xx.xx (replace the [!at] with a @)
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 04 Jan 2000 23:37:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 238
v=0
o=root 24513 24513 IN IP4 xx.xx.xx.xx
s=session
c=IN IP4 xx.xx.xx.xx
t=0 0
m=audio 19648 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 - - - -
I can't read the debug info. Any clue please? |