[Asterisk-Users] Re: Cisco 7960 International

[Shaun]

Well looks like the phone is sending some data...  I was unable to debug the 
problem however..

~Shaun

<-- SIP read from 68.5.xxx.xxx:1025:
INVITE sip:9011905326471222 at 204.10.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK396e6066
From: "302" <sip:302 at 204.10.xxx.xxx>;tag=00115cd9d037086a148db8bd-5c06b810
To: <sip:9011905326471222 at 204.10.xxx.xxx>
Call-ID: 00115cd9-d0370007-64a30013-349be63b at 192.168.1.101
Max-Forwards: 70
CSeq: 101 INVITE
User-Agent: Cisco-CP7960G/8.0
Contact: <sip:302 at 192.168.1.101:5060;transport=udp>
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Supported: replaces,join,norefersub
Content-Length: 274
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 6585 0 IN IP4 192.168.1.101
s=SIP Call
t=0 0
m=audio 16396 RTP/AVP 0 8 18 101
c=IN IP4 192.168.1.101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/0
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

--- (16 headers 13 lines)---
Using INVITE request as basis request - 
00115cd9-d0370007-64a30013-349be63b at 192.168.1.101
Reliably Transmitting (NAT) to 68.5.xxx.xxx:1025:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 
192.168.1.101:5060;branch=z9hG4bK396e6066;received=68.5.xxx.xxx
From: "302" <sip:302 at 204.10.xxx.xxx>;tag=00115cd9d037086a148db8bd-5c06b810
To: <sip:9011905326471222 at 204.10.xxx.xxx>;tag=as10ca5f54
Call-ID: 00115cd9-d0370007-64a30013-349be63b at 192.168.1.101
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9011905326471222 at 204.10.xxx.xxx>
Proxy-Authenticate: Digest realm="asterisk", nonce="334c26ca"
Content-Length: 0


---
Scheduling destruction of call 
'00115cd9-d0370007-64a30013-349be63b at 192.168.1.101' in 15000 ms
Found user '302'
<-- SIP read from 68.5.xxx.xxx:1025:
ACK sip:9011905326471222 at 204.10.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK396e6066
From: "302" <sip:302 at 204.10.xxx.xxx>;tag=00115cd9d037086a148db8bd-5c06b810
To: <sip:9011905326471222 at 204.10.xxx.xxx>;tag=as10ca5f54
Call-ID: 00115cd9-d0370007-64a30013-349be63b at 192.168.1.101
CSeq: 101 ACK
Content-Length: 0


--- (7 headers 0 lines)---
<-- SIP read from 68.5.xxx.xxx:1025:
INVITE sip:9011905326471222 at 204.10.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK4716b54a
From: "302" <sip:302 at 204.10.xxx.xxx>;tag=00115cd9d037086a148db8bd-5c06b810
To: <sip:9011905326471222 at 204.10.xxx.xxx>
Call-ID: 00115cd9-d0370007-64a30013-349be63b at 192.168.1.101
Max-Forwards: 70
CSeq: 102 INVITE
User-Agent: Cisco-CP7960G/8.0
Contact: <sip:302 at 192.168.1.101:5060;transport=udp>
Proxy-Authorization: Digest 
username="302",realm="asterisk",uri="sip:9011905326471222 at 204.10.xxx.xxx",response="28fb3b24b9ba8f8585096a03a80dfe48",nonce="334c26ca",algorithm=md5
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Supported: replaces,join,norefersub
Content-Length: 274
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 6585 0 IN IP4 192.168.1.101
s=SIP Call
t=0 0
m=audio 16396 RTP/AVP 0 8 18 101
c=IN IP4 192.168.1.101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/0
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

--- (17 headers 13 lines)---
Using INVITE request as basis request - 
00115cd9-d0370007-64a30013-349be63b at 192.168.1.101
Sending to 192.168.1.101 : 5060 (NAT)
Found user '302'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.101:16396
Found description format PCMU
Found description format PCMA
Found description format G729
Found description format telephone-event
Capabilities: us - 0x1f07ff 
(g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|png|h261|h263|h263p), 
peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10c 
(ulaw|alaw|g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 
(telephone-event), combined - 0x1 (telephone-event)
Looking for 9011905326471222 in default (domain 204.10.xxx.xxx)
Reliably Transmitting (NAT) to 68.5.xxx.xxx:1025:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 
192.168.1.101:5060;branch=z9hG4bK4716b54a;received=68.5.xxx.xxx
From: "302" <sip:302 at 204.10.xxx.xxx>;tag=00115cd9d037086a148db8bd-5c06b810
To: <sip:9011905326471222 at 204.10.xxx.xxx>;tag=as10ca5f54
Call-ID: 00115cd9-d0370007-64a30013-349be63b at 192.168.1.101
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9011905326471222 at 204.10.xxx.xxx>
Content-Length: 0


---
<-- SIP read from 68.5.xxx.xxx:1025:
ACK sip:9011905326471222 at 204.10.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK4716b54a
From: "302" <sip:302 at 204.10.xxx.xxx>;tag=00115cd9d037086a148db8bd-5c06b810
To: <sip:9011905326471222 at 204.10.xxx.xxx>;tag=as10ca5f54
Call-ID: 00115cd9-d0370007-64a30013-349be63b at 192.168.1.101
CSeq: 102 ACK
Content-Length: 0


--- (7 headers 0 lines)---
Destroying call '00115cd9-d0370007-64a30013-349be63b at 192.168.1.101'