Configuring the Cisco SPA8800 IP Telephony Gateway in an Asterisk Environment
© 2009 Cisco Systems, Inc. All rights reserved.
Page 27 of 69
M2: [1:0]AUD Rel Call
M0: CC:Ended
M2: DLG Terminated 345e80
M2: Sess Terminated
M0: [0]FM Alert Stop RxTx (c=002b02d8;a=0)
M0: [0:0]AUD Rel Call
M0: [0]On Hook
SPA8800 SIP Debugging
The SPA8800 can also supply SIP debug information to assist with troubleshooting. Enable SIP
debugging as follows: SPA8800 web-ui > Voice tab > Line
N
tab > SIP Settings > SIP Debug
Option:
Following is an example of syslog information produced at Debug Level 3 with SIP Debug Option
set to full [numbers changed]. This example shows a SIP INVITE, the 100 trying, a report that an
unlisted codec type is being requested, and the BYE:
M1: [1]<<192.168.2.20:5060(826)
INVITE sip:[email protected]:5161 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.20:5060;branch=z9hG4bK58d12649;rport
From: "SPA8k8Phone1" <sip:[email protected]>;tag=as16cfeb3c
To: <sip:[email protected]:5161>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 28 May 2009 13:29:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 285
v=0
o=root 7298 7298 IN IP4 192.168.2.20
s=session
c=IN IP4 192.168.2.20
t=0 0
m=audio 10114 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
