[Asterisk-Users] sip not working suddenly

[Jonathan k. Creasy]

Anyone know what's causing this:


<-- SIP read from x.x.x.x:56800:
ACK sip:566 at x.x.x.x:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.200.16;branch=z9hG4bKba5c00f818387D67
From: "Xxx xxx" <sip:user1 at x.x.x.x>;tag=69375B3E-ACF6C78B
To: <sip:566 at x.x.x.x;user=phone>;tag=as57402fc2
CSeq: 1 ACK
Call-ID: 758a4aea-c82e1e2c-cc3440f1 at 192.168.200.16
Contact: <sip:user1 at 192.168.200.16>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_301-UA/1.5.2.0054
Max-Forwards: 70
Content-Length: 0


--- (11 headers 0 lines)---
Destroying call '758a4aea-c82e1e2c-cc3440f1 at 192.168.200.16'
lou01*CLI>
<-- SIP read from x.x.x.x:56800:
INVITE sip:566 at x.x.x.x:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.200.16;branch=z9hG4bKc56387b5FF26D6A0
From: "Xxx xxx" <sip:user1 at x.x.x.x>;tag=69375B3E-ACF6C78B
To: <sip:566 at x.x.x.x;user=phone>
CSeq: 2 INVITE
Call-ID: 758a4aea-c82e1e2c-cc3440f1 at 192.168.200.16
Contact: <sip:user1 at 192.168.200.16>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_301-UA/1.5.2.0054
Supported: 100rel,replace
Allow-Events: talk,hold,conference
Proxy-Authorization: Digest username="user1", realm="asterisk",
nonce="07b9f9a3", uri="sip:566 at x.x.x.x:5060;user=phone",
response="a8f005540682f07a88e023d50135cce0", algorithm=MD5
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 253

v=0
o=- 1130439113 1130439113 IN IP4 192.168.200.16
s=Polycom IP Phone
c=IN IP4 192.168.200.16
t=0 0
a=sendrecv
m=audio 2228 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000

--- (15 headers 11 lines)---
Using INVITE request as basis request -
758a4aea-c82e1e2c-cc3440f1 at 192.168.200.16
Reliably Transmitting (NAT) to x.x.x.x:56800:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
192.168.200.16;branch=z9hG4bKc56387b5FF26D6A0;received=x.x.x.x;rport=568
00
From: "Xxx xxx" <sip:user1 at x.x.x.x>;tag=69375B3E-ACF6C78B
To: <sip:566 at x.x.x.x;user=phone>;tag=as71adaedb
Call-ID: 758a4aea-c82e1e2c-cc3440f1 at 192.168.200.16
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:566 at 69.64.15.18>
Proxy-Authenticate: Digest realm="asterisk", nonce="56bff437"
Content-Length: 0


---
Scheduling destruction of call
'758a4aea-c82e1e2c-cc3440f1 at 192.168.200.16' in 15000 ms

<-- SIP read from x.x.x.x:56800:
INVITE sip:566 at x.x.x.x:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.200.16;branch=z9hG4bKc56387b5FF26D6A0
From: "xxx xxx" <sip:user1 at x.x.x.x>;tag=69375B3E-ACF6C78B
To: <sip:566 at x.x.x.x;user=phone>
CSeq: 2 INVITE
Call-ID: 758a4aea-c82e1e2c-cc3440f1 at 192.168.200.16
Contact: <sip:user1 at 192.168.200.16>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_301-UA/1.5.2.0054
Supported: 100rel,replace
Allow-Events: talk,hold,conference
Proxy-Authorization: Digest username="user1", realm="asterisk",
nonce="07b9f9a3", uri="sip:566 at x.x.x.x:5060;user=phone",
response="a8f005540682f07a88e023d50135cce0", algorithm=MD5
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 253

v=0
o=- 1130439113 1130439113 IN IP4 192.168.200.16
s=Polycom IP Phone
c=IN IP4 192.168.200.16
t=0 0
a=sendrecv
m=audio 2228 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000

--- (15 headers 11 lines)---
Ignoring this INVITE request
Transmitting (NAT) to x.x.x.x:56800:
SIP/2.0 488 Not Acceptable Here (codec error)
Via: SIP/2.0/UDP
192.168.200.16;branch=z9hG4bKc56387b5FF26D6A0;received=x.x.x.x;rport=568
00
From: "xxx xxx" <sip:xxx at x.x.x.x>;tag=69375B3E-ACF6C78B
To: <sip:566 at x.x.x.x;user=phone>;tag=as71adaedb
Call-ID: 758a4aea-c82e1e2c-cc3440f1 at 192.168.200.16
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:566 at x.x.x.x>
Content-Length: 0


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