[AsteriskBrasil] 7. Re: Integrando 1.6 com Avaya Definity via SIP

[Rafael Augusto]

Caro Guilherme, desde de já agradeço pela ajuda. Segue o debug, realmente está dando o 404, mas o contexto e o plano de discagem estão corretos. Está estranho. Obrigado.

<--- SIP read from TCP://10.200.132.97:29878 --->
INVITE sip:41190 em avaya.com SIP/2.0
From: "Rafael Augusto - Telecom" <sip:38495 em invalid.unknown.domain>;tag=8074e48ce16bdf112f24b3e81900
To: "41190" <sip:41190 em avaya.com>
Call-ID: 8074e48ce16bdf113f24b3e81900
CSeq: 1 INVITE
Max-Forwards: 71
Route: <sip:10.200.50.133;lr;phase=terminating;transport=tcp>
Record-Route: <sip:10.200.132.97;lr;transport=tcp>
Via: SIP/2.0/TCP 10.200.132.97;branch=z9hG4bK8074e48ce16bdf114f24b3e81900
User-Agent: Avaya CM/R015x.02.0.947.3
Supported: timer, replaces, join, histinfo, 100rel
Allow: INVITE, CANCEL, BYE, ACK, PRACK, SUBSCRIBE, NOTIFY, REFER, OPTIONS, INFO, PUBLISH
Contact: "Rafael Augusto - Telecom" <sip:38495 em 10.200.132.97;transport=tcp>
Session-Expires: 1200;refresher=uac
Min-SE: 1200
P-Asserted-Identity: "Rafael Augusto - Telecom" <sip:38495 em invalid.unknown.domain>
Accept-Language: en
Content-Type: application/sdp
History-Info: <sip:41190 em avaya.com>;index=1
History-Info: "41190" <sip:41190 em avaya.com>;index=1.1
Alert-Info: <cid:internal em avaya.com>;avaya-cm-alert-type=internal
Content-Length: 188
v=0
o=- 1 1 IN IP4 10.200.132.97
s=-
c=IN IP4 10.200.96.145
b=AS:64
t=0 0
m=audio 2176 RTP/AVP 18 127
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:127 telephone-event/8000
<------------->
--- (22 headers 10 lines) ---
  == Using SIP RTP CoS mark 5
  == Using UDPTL CoS mark 5
Sending to 10.200.132.97 : 5060 (no NAT)
Using INVITE request as basis request - 8074e48ce16bdf113f24b3e81900
No user '38495' in SIP users list
No matching peer for '38495' from '10.200.132.97:29878'
Found RTP audio format 18
Found RTP audio format 127
Peer audio RTP is at port 10.200.96.145:2176
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 127
Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.200.96.145:2176
Looking for 41190 in default (domain avaya.com)
<--- Reliably Transmitting (no NAT) to 10.200.132.97:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/TCP 10.200.132.97;branch=z9hG4bK8074e48ce16bdf114f24b3e81900;received=10.200.132.97
From: "Rafael Augusto - Telecom" <sip:38495 em invalid.unknown.domain>;tag=8074e48ce16bdf112f24b3e81900
To: "41190" <sip:41190 em avaya.com>;tag=as50d9a660
Call-ID: 8074e48ce16bdf113f24b3e81900
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0

<------------>
[May 14 09:56:15] NOTICE[30262]: chan_sip.c:17295 handle_request_invite: Call from '' to extension '41190' rejected because extension not found.
Scheduling destruction of SIP dialog '8074e48ce16bdf113f24b3e81900' in 32000 ms (Method: INVITE)
udppbxip01*CLI>
<--- SIP read from TCP://10.200.132.97:29878 --->
ACK sip:41190 em avaya.com SIP/2.0
From: "Rafael Augusto - Telecom" <sip:38495 em invalid.unknown.domain>;tag=8074e48ce16bdf112f24b3e81900
To: "41190" <sip:41190 em avaya.com>;tag=as50d9a660
Call-ID: 8074e48ce16bdf113f24b3e81900
Via: SIP/2.0/TCP 10.200.132.97;branch=z9hG4bK8074e48ce16bdf114f24b3e81900;received=10.200.132.97
CSeq: 1 ACK
Max-Forwards: 70
Route: <sip:10.200.50.133;lr;phase=terminating;transport=tcp>
User-Agent: Avaya CM/R015x.02.0.947.3
Content-Length: 0
 


      
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