[AsteriskBrasil] SIP Trunk Asterisk x Alcatel

Sexta Fevereiro 22 14:47:25 BRT 2013

Não há necessidade de autenticação, ambos os lados estão com IPs 
cadastrados e pertencem a uma rede considerada interna.
O parametro host sendo diferente dynamic não exigem username/password 
no asterisk.

O grande lance é no pacote com a informação abaixo:

INVITE sip:9202 em 172.16.200.92:5060 SIP/2.0

Minha dúvida é de onde surgiu isso... Alguma sugestão?

Em 2013-02-22 13:02, Guilherme Rezende escreveu:
> Estou achando estranho que não existe parâmetro algum de
> autenticação...  Talvez dentro da conf da Alcatel seja necessário
> habilitar o forward de sua rede, que no caso deve ser 172.16.1.0/24.
> Acredito ser algo de permissão.
>
> Em 22/02/2013 09:49, Jefferson B. Limeira escreveu:
>> Bom dia,
>>
>>     Estamos participando da integração de um asterisk 1.6.2.11 com 
>> uma
>> Alcatel via SIP. Quando ligo do asterisk para a Alcatel recebo um
>> forwarding da chamada de volta para o asterisk. Segue maiores
>> informações:
>>
>> sip.conf:
>>
>> [alcatel]
>> host=172.16.1.3
>> context=from-Alcatel
>> type=friend
>> nat=no
>> disallow=all
>> allow=alaw
>>
>> extensions.conf
>>
>> exten =>  _6X.,1,Dial(SIP/${EXTEN:1}@alcatel)
>>    same =>  n,HangUp
>>
>> no console do asterisk durante a chamada
>>
>>       -- Executing [69202 em saida:1] Dial("SIP/jefferson-00001a43",
>> "SIP/9202 em alcatel") in new stack
>>     == Using SIP RTP CoS mark 5
>>       -- Called 9202 em alcatel
>>
>>       -- Now forwarding SIP/jefferson-00001a43 to
>> 'Local/9202 em from-Alcatel' (thanks to SIP/alcatel-00001a44)
>>
>> [Feb 22 09:36:24] NOTICE[19305]: chan_local.c:538 local_call: No 
>> such
>> extension/context 9202 em from-Alcatel while calling Local channel
>> [Feb 22 09:36:24] NOTICE[19305]: app_dial.c:789 do_forward: Failed 
>> to
>> dial on local channel for call forward to '9202 em from-Alcatel'
>>     == Everyone is busy/congested at this time (1:0/0/1)
>>       -- Executing [69202 em saida:2] Hangup("SIP/jefferson-00001a43", 
>> "")
>> in new stack
>>     == Spawn extension (TI, 69202, 2) exited non-zero on
>> 'SIP/jefferson-00001a43'
>>
>>
>> Segue sip debug deste peer
>>
>> asterisk*CLI>  sip set debug peer alcatel
>> SIP Debugging Enabled for IP: 172.16.1.3:5060
>>     == Using SIP RTP CoS mark 5
>>       -- Executing [69202 em TI:1] Dial("SIP/jefferson-00001a46",
>> "SIP/9202 em alcatel") in new stack
>>     == Using SIP RTP CoS mark 5
>> Audio is at 172.16.200.92 port 5404
>> Adding codec 0x8 (alaw) to SDP
>> Adding non-codec 0x1 (telephone-event) to SDP
>> Reliably Transmitting (no NAT) to 172.16.1.3:5060:
>> INVITE sip:9202 em 172.16.1.3 SIP/2.0
>> Via: SIP/2.0/UDP 172.16.200.92:5060;branch=z9hG4bK7b79bc0e;rport
>> Max-Forwards: 70
>>   From: "jefferson"<sip:jefferson em 172.16.200.92>;tag=as1b46e101
>> To:<sip:9202 em 172.16.1.3>
>> Contact:<sip:jefferson em 172.16.200.92>
>> Call-ID: 7f135fc326d306e80dc2bf4d6af36a20 em 172.16.200.92
>> CSeq: 102 INVITE
>> User-Agent: Asterisk PBX 1.6.2.11
>> Date: Fri, 22 Feb 2013 12:38:28 GMT
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
>> INFO
>> Supported: replaces, timer
>> Content-Type: application/sdp
>> Content-Length: 238
>>
>> v=0
>> o=root 303517300 303517300 IN IP4 172.16.200.92
>> s=Asterisk PBX 1.6.2.11
>> c=IN IP4 172.16.200.92
>> t=0 0
>> m=audio 5404 RTP/AVP 8 101
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=ptime:20
>> a=sendrecv
>> ---
>> -- Called 9202 em alcatel
>>
>> <--- SIP read from UDP:172.16.1.3:5060 --->
>> SIP/2.0 100 Trying
>> To:<sip:9202 em 172.16.1.3>
>>   From: "jefferson"<sip:jefferson em 172.16.200.92>;tag=as1b46e101
>> Call-ID: 7f135fc326d306e80dc2bf4d6af36a20 em 172.16.200.92
>> CSeq: 102 INVITE
>> Via: SIP/2.0/UDP
>> 
>> 172.16.200.92:5060;received=172.16.200.92;branch=z9hG4bK7b79bc0e;rport=5060
>> Content-Length: 0
>> <------------->
>> --- (7 headers 0 lines) ---
>>
>> <--- SIP read from UDP:172.16.1.3:5060 --->
>> SIP/2.0 100 Trying
>> To:<sip:9202 em 172.16.1.3>
>>   From: "jefferson"<sip:jefferson em 172.16.200.92>;tag=as1b46e101
>> Call-ID: 7f135fc326d306e80dc2bf4d6af36a20 em 172.16.200.92
>> CSeq: 102 INVITE
>> Via: SIP/2.0/UDP
>> 
>> 172.16.200.92:5060;received=172.16.200.92;branch=z9hG4bK7b79bc0e;rport=5060
>> Content-Length: 0
>> <------------->
>> --- (7 headers 0 lines) ---
>>
>> <--- SIP read from UDP:172.16.1.3:5060 --->
>> INVITE sip:9202 em 172.16.200.92:5060 SIP/2.0
>> Record-Route:<sip:172.16.1.3;lr;transport=UDP>
>> Via: SIP/2.0/UDP
>> 
>> 172.16.1.3:5060;branch=z9hG4bK467b772a9f9ef266e7d514eede2bf348cc48719d6f59fafe0690167af90cbb1d
>> Via: SIP/2.0/UDP
>> 
>> 172.16.200.92:5060;received=172.16.200.92;branch=z9hG4bK7b79bc0e;rport=5060
>> Max-Forwards: 69
>>   From: "jefferson"<sip:jefferson em 172.16.200.92>;tag=as1b46e101
>> To:<sip:9202 em 172.16.1.3>
>> Contact:<sip:jefferson em 172.16.1.3:5060>
>> Call-ID: 7f135fc326d306e80dc2bf4d6af36a20 em 172.16.200.92
>> CSeq: 102 INVITE
>> User-Agent: Asterisk PBX 1.6.2.11
>> Date: Fri, 22 Feb 2013 12:38:28
>> Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO
>> Supported: replaces,timer
>> Content-Type: application/sdp
>> Content-Length: 236
>> Session-Expires: 1800
>>
>> v=0
>> o=root 303517300 303517300 IN IP4 172.16.1.3
>> s=Asterisk PBX 1.6.2.11
>> c=IN IP4 172.16.1.3
>> t=0 0
>> m=audio 5404 RTP/AVP 8 101
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=ptime:20
>> a=sendrecv
>>
>> <------------->
>> --- (17 headers 11 lines) ---
>>
>> <--- Transmitting (no NAT) to 172.16.1.3:5060 --->
>> SIP/2.0 100 Trying
>> Via: SIP/2.0/UDP
>> 
>> 172.16.1.3:5060;branch=z9hG4bK467b772a9f9ef266e7d514eede2bf348cc48719d6f59fafe0690167af90cbb1d;received=172.16.1.3
>> Via: SIP/2.0/UDP
>> 
>> 172.16.200.92:5060;received=172.16.200.92;branch=z9hG4bK7b79bc0e;rport=5060
>> Record-Route:<sip:172.16.1.3;lr;transport=UDP>
>>   From: "jefferson"<sip:jefferson em 172.16.200.92>;tag=as1b46e101
>> To:<sip:9202 em 172.16.1.3>
>> Call-ID: 7f135fc326d306e80dc2bf4d6af36a20 em 172.16.200.92
>> CSeq: 102 INVITE
>> Server: Asterisk PBX 1.6.2.11
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
>> INFO
>> Supported: replaces, timer
>> Contact:<sip:jefferson em 172.16.200.92>
>> Content-Length: 0
>>
>> <------------>
>>       -- Now forwarding SIP/jefferson-00001a46 to
>> 'Local/9202 em from-Alcatel' (thanks to SIP/alcatel-00001a47)
>> [Feb 22 09:38:28] NOTICE[19309]: chan_local.c:538 local_call: No 
>> such
>> extension/context 9202 em from-Alcatel while calling Local channel
>> [Feb 22 09:38:28] NOTICE[19309]: app_dial.c:789 do_forward: Failed 
>> to
>> dial on local channel for call forward to '9202 em from-Alcatel'
>> Scheduling destruction of SIP dialog
>> '7f135fc326d306e80dc2bf4d6af36a20 em 172.16.200.92' in 32000 ms 
>> (Method:
>> INVITE)
>> Reliably Transmitting (no NAT) to 172.16.1.3:5060:
>> CANCEL sip:9202 em 172.16.1.3 SIP/2.0
>> Via: SIP/2.0/UDP 172.16.200.92:5060;branch=z9hG4bK7b79bc0e;rport
>> Max-Forwards: 70
>>   From: "jefferson"<sip:jefferson em 172.16.200.92>;tag=as1b46e101
>> To:<sip:9202 em 172.16.1.3>
>> Call-ID: 7f135fc326d306e80dc2bf4d6af36a20 em 172.16.200.92
>> CSeq: 102 CANCEL
>> User-Agent: Asterisk PBX 1.6.2.11
>> Content-Length: 0
>>
>> ---
>> Scheduling destruction of SIP dialog
>> '7f135fc326d306e80dc2bf4d6af36a20 em 172.16.200.92' in 32000 ms 
>> (Method:
>> INVITE)
>>     == Everyone is busy/congested at this time (1:0/0/1)
>>       -- Executing [69202 em TI:2] Hangup("SIP/jefferson-00001a46", "") 
>> in
>> new stack
>>     == Spawn extension (TI, 69202, 2) exited non-zero on
>> 'SIP/jefferson-00001a46'
>>
>> <--- SIP read from UDP:172.16.1.3:5060 --->
>> SIP/2.0 200 OK
>> To:<sip:9202 em 172.16.1.3>
>>   From: "jefferson"<sip:jefferson em 172.16.200.92>;tag=as1b46e101
>> Call-ID: 7f135fc326d306e80dc2bf4d6af36a20 em 172.16.200.92
>> CSeq: 102 CANCEL
>> Via: SIP/2.0/UDP
>> 
>> 172.16.200.92:5060;received=172.16.200.92;branch=z9hG4bK7b79bc0e;rport=5060
>> Content-Length: 0
>> <------------->
>>
>

-- 
[]'s Jefferson B. Limeira
jbl em internexxus.com.br
(41) 9928-8628