[AsteriskBrasil] SIP Trunk Asterisk x Alcatel

Sexta Fevereiro 22 13:02:22 BRT 2013

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
> <------------->
>