[AsteriskBrasil] Interligar 2 Elastix
Otavio Asterisk
otavioasterisk em gmail.com
Segunda Fevereiro 8 11:21:25 BRST 2010
Galera, bom dia.
Há um tempo atras eu postei uma dúvida quanto a interligar dois PBXIP via
SIP.
O Mestre ASterisk me respondeu com um link para dois tutorias do seu site.
Um deles, esse aqui:
http://mestreasterisk.com.br/configuracao/interligar-servidores-asterisk-via-sip/
eu o segui, mas quanto tento fazer ligação do PBXIP1 para o 2, no log do 1
aparecem umas mensagens estranhas, do tipo: 489 Bad Event, SIP/2.0 403
Forbidden.
No 2 não aparece nada....
Eu joguei o log num arquivo .txt... e anexei o mesmo aqui no email...
Se alguém puder me ajudar, agradeço desde já..
Valeu
--
Otávio
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Um anexo em HTML foi limpo...
URL: http://listas.asteriskbrasil.org/pipermail/asteriskbrasil/attachments/20100208/052122a0/attachment.htm
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Call-ID: OThkZGUyNGJmZTRhMDM0NGNjMzZlYjFhMzg2NGZmOGU.
CSeq: 1 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
elastix*CLI>
<--- SIP read from 192.168.1.10:5060 --->
INVITE sip:2000 em 192.168.1.11;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK-d8754z-5f7d3f5944ff5ef8-1---d8754z-
Max-Forwards: 70
Contact: <sip:1000 em 192.168.1.10:5060;transport=UDP>
To: <sip:2000 em 192.168.1.11;transport=UDP>
From: "1000"<sip:1000 em 192.168.1.11;transport=UDP>;tag=f33fc456
Call-ID: OThkZGUyNGJmZTRhMDM0NGNjMzZlYjFhMzg2NGZmOGU.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Proxy-Authorization: Digest username="1000",realm="asterisk",nonce="651a0cf6",uri="sip:2000 em 192.168.1.11;transport=UDP",response="9f6e298f3ab4d7c344dace2d8fc45f47",algorithm=MD5
User-Agent: Zoiper rev.5324
Content-Length: 327
v=0
o=Zoiper_user 0 0 IN IP4 192.168.1.10
s=Zoiper_session
c=IN IP4 192.168.1.10
t=0 0
m=audio 8000 RTP/AVP 3 0 8 110 98 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:110 speex/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (13 headers 15 lines) ---
Sending to 192.168.1.10 : 5060 (NAT)
Using INVITE request as basis request - OThkZGUyNGJmZTRhMDM0NGNjMzZlYjFhMzg2NGZmOGU.
Found user '1000'
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 110
Found RTP audio format 98
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.10:8000
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format speex for ID 110
Found audio description format iLBC for ID 98
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.1.10:8000
Looking for 2000 in contexto1000 (domain 192.168.1.11)
list_route: hop: <sip:1000 em 192.168.1.10:5060;transport=UDP>
<--- Transmitting (NAT) to 192.168.1.10:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK-d8754z-5f7d3f5944ff5ef8-1---d8754z-;received=192.168.1.10
From: "1000"<sip:1000 em 192.168.1.11;transport=UDP>;tag=f33fc456
To: <sip:2000 em 192.168.1.11;transport=UDP>
Call-ID: OThkZGUyNGJmZTRhMDM0NGNjMzZlYjFhMzg2NGZmOGU.
CSeq: 2 INVITE
ser-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:2000 em 192.168.1.11>
Content-Length: 0
<------------>
-- Executing [2000 em contexto1000:1] Dial("SIP/1000-0a0f9b78", "SIP/1001/2000") in new stack
Audio is at 192.168.1.11 port 10140
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.1.12:5060:
INVITE sip:2000 em 192.168.1.12 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK63684dce;rport
From: "device" <sip:1000 em 192.168.1.11>;tag=as3f0010ce
To: <sip:2000 em 192.168.1.12>
Contact: <sip:1000 em 192.168.1.11>
Call-ID: 5237928d4a0ebce93235ca1379ff3e91 em 192.168.1.11
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 03 Feb 2010 07:47:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 238
elastix*CLI>
v=0
o=root 7992 7992 IN IP4 192.168.1.11
s=session
c=IN IP4 192.168.1.11
t=0 0
m=audio 10140 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
<--- SIP read from 192.168.1.12:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK63684dce;received=192.168.1.11;rport=5060
From: "device" <sip:1000 em 192.168.1.11>;tag=as3f0010ce
To: <sip:2000 em 192.168.1.12>;tag=as17dc6a33
Call-ID: 5237928d4a0ebce93235ca1379ff3e91 em 192.168.1.11
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4cc328e6"
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Transmitting (NAT) to 192.168.1.12:5060:
ACK sip:2000 em 192.168.1.12 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK63684dce;rport
From: "device" <sip:1000 em 192.168.1.11>;tag=as3f0010ce
To: <sip:2000 em 192.168.1.12>;tag=as17dc6a33
Contact: <sip:1000 em 192.168.1.11>
Call-ID: 5237928d4a0ebce93235ca1379ff3e91 em 192.168.1.11
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Audio is at 192.168.1.11 port 10140
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.1.12:5060:
INVITE sip:2000 em 192.168.1.12 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK73b01775;rport
From: "device" <sip:1000 em 192.168.1.11>;tag=as3f0010ce
To: <sip:2000 em 192.168.1.12>
Contact: <sip:1000 em 192.168.1.11>
Call-ID: 5237928d4a0ebce93235ca1379ff3e91 em 192.168.1.11
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Proxy-Authorization: Digest username="1001", realm="asterisk", algorithm=MD5, uri="sip:2000 em 192.168.1.12", nonce="4cc328e6", response="6b312b28dc6ff85285f167902b988b19"
Date: Wed, 03 Feb 2010 07:47:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 238
v=0
o=root 7992 7993 IN IP4 192.168.1.11
s=session
c=IN IP4 192.168.1.11
t=0 0
m=audio 10140 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called 1001/2000
<--- SIP read from 192.168.1.12:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK73b01775;received=192.168.1.11;rport=5060
From: "device" <sip:1000 em 192.168.1.11>;tag=as3f0010ce
To: <sip:2000 em 192.168.1.12>;tag=as17dc6a33
Call-ID: 5237928d4a0ebce93235ca1379ff3e91 em 192.168.1.11
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Transmitting (NAT) to 192.168.1.12:5060:
ACK sip:2000 em 192.168.1.12 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK73b01775;rport
From: "device" <sip:1000 em 192.168.1.11>;tag=as3f0010ce
To: <sip:2000 em 192.168.1.12>;tag=as17dc6a33
Contact: <sip:1000 em 192.168.1.11>
Call-ID: 5237928d4a0ebce93235ca1379ff3e91 em 192.168.1.11
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
-- SIP/1001-0a1533c0 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [2000 em contexto1000:2] Hangup("SIP/1000-0a0f9b78", "") in new stack
== Spawn extension (contexto1000, 2000, 2) exited non-zero on 'SIP/1000-0a0f9b78'
Scheduling destruction of SIP dialog 'OThkZGUyNGJmZTRhMDM0NGNjMzZlYjFhMzg2NGZmOGU.' in 32000 ms (Method: INVITE)
<--- Reliably Transmitting (NAT) to 192.168.1.10:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK-d8754z-5f7d3f5944ff5ef8-1---d8754z-;received=192.168.1.10
From: "1000"<sip:1000 em 192.168.1.11;transport=UDP>;tag=f33fc456
To: <sip:2000 em 192.168.1.11;transport=UDP>;tag=as39b0c200
Call-ID: OThkZGUyNGJmZTRhMDM0NGNjMzZlYjFhMzg2NGZmOGU.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
elastix*CLI>
<--- SIP read from 192.168.1.10:5060 --->
ACK sip:2000 em 192.168.1.11;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK-d8754z-5f7d3f5944ff5ef8-1---d8754z-
Max-Forwards: 70
To: <sip:2000 em 192.168.1.11;transport=UDP>;tag=as39b0c200
From: "1000"<sip:1000 em 192.168.1.11;transport=UDP>;tag=f33fc456
Call-ID: OThkZGUyNGJmZTRhMDM0NGNjMzZlYjFhMzg2NGZmOGU.
CSeq: 2 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '5237928d4a0ebce93235ca1379ff3e91 em 192.168.1.11' Method: INVITE
elastix*CLI>
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