[AsteriskBrasil] Problemas com Ocupado.
Bom dia Srs.
Estou com o seguinte problema.
Quando efetuo uma ligação via tronco IAX, se o ramal da outra ponta estiver
ocupado, não me dá o sinal de Ocupado.
Olhem o que acontece.
[Aug 14 10:38:38] NOTICE[3057] chan_dahdi.c: MFC/R2 call has been accepted
on chan 34
[Aug 14 10:38:38] VERBOSE[3593] logger.c: -- Executing
[106...@from-internal:1] Dial(DAHDI/34-1, IAX2//6103|90) in new
stack
[Aug 14 10:38:38] NOTICE[3593] app_dial.c: Hey! chan DAHDI/34-1's
context='from-internal', and exten='106103'
[Aug 14 10:38:38] DEBUG[3593] chan_iax2.c: prepending 2 to prefs
[Aug 14 10:38:38] VERBOSE[3593] logger.c: -- Called /6103
[Aug 14 10:38:38] VERBOSE[3062] logger.c: -- Call accepted by 10.12.2.46
(format gsm)
[Aug 14 10:38:38] VERBOSE[3062] logger.c: -- Format for call is gsm
[Aug 14 10:39:03] VERBOSE[3593] logger.c: -- Hungup 'IAX2/-16384'
[Aug 14 10:39:03] VERBOSE[3593] logger.c: -- No one is available to
answer at this time (1:0/0/0)
[Aug 14 10:39:03] VERBOSE[3593] logger.c: -- Executing
[106...@from-internal:2] Hangup(DAHDI/34-1, ) in new stack
[Aug 14 10:39:03] VERBOSE[3593] logger.c: == Spawn extension
(from-internal, 106103, 2) exited non-zero on 'DAHDI/34-1'
[Aug 14 10:39:03] VERBOSE[3593] logger.c: -- Executing [...@from-internal:1]
Macro(DAHDI/34-1, hangupcall) in new stack
[Aug 14 10:39:03] VERBOSE[3593] logger.c: -- Executing
[...@macro-hangupcall:1] ResetCDR(DAHDI/34-1, w) in new stack
[Aug 14 10:39:03] DEBUG[3593] app_macro.c: Executed application: ResetCDR
[Aug 14 10:39:03] VERBOSE[3593] logger.c: -- Executing
[...@macro-hangupcall:2] NoCDR(DAHDI/34-1, ) in new stack
[Aug 14 10:39:03] DEBUG[3593] app_macro.c: Executed application: NoCDR
[Aug 14 10:39:03] VERBOSE[3593] logger.c: -- Executing
[...@macro-hangupcall:3] GotoIf(DAHDI/34-1, 1?skiprg) in new stack
[Aug 14 10:39:03] VERBOSE[3593] logger.c: -- Goto (macro-hangupcall,s,6)
[Aug 14 10:39:03] DEBUG[3593] app_macro.c: Executed application: GotoIf
[Aug 14 10:39:03] VERBOSE[3593] logger.c: -- Executing
[...@macro-hangupcall:6] GotoIf(DAHDI/34-1, 1?skipblkvm) in new stack
[Aug 14 10:39:03] VERBOSE[3593] logger.c: -- Goto (macro-hangupcall,s,9)
[Aug 14 10:39:03] DEBUG[3593] app_macro.c: Executed application: GotoIf
[Aug 14 10:39:03] VERBOSE[3593] logger.c: -- Executing
[...@macro-hangupcall:9] GotoIf(DAHDI/34-1, 1?theend) in new stack
[Aug 14 10:39:03] VERBOSE[3593] logger.c: -- Goto
(macro-hangupcall,s,11)
[Aug 14 10:39:03] DEBUG[3593] app_macro.c: Executed application: GotoIf
[Aug 14 10:39:03] VERBOSE[3593] logger.c: -- Executing
[...@macro-hangupcall:11] Hangup(DAHDI/34-1, ) in new stack
[Aug 14 10:39:03] VERBOSE[3593] logger.c: == Spawn extension
(macro-hangupcall, s, 11) exited non-zero on 'DAHDI/34-1' in macro
'hangupcall'
[Aug 14 10:39:03] VERBOSE[3593] logger.c: == Spawn extension
(from-internal, h, 1) exited non-zero on 'DAHDI/34-1'
[Aug 14 10:39:03] DEBUG[3593] chan_dahdi.c: disconnecting MFC/R2 call on
chan 34
[Aug 14 10:39:03] DEBUG[3593] chan_dahdi.c: Chan 34 - CAS Tx [CLEAR BACK]
0x0C
[Aug 14 10:39:03] DEBUG[3593] chan_dahdi.c: Chan 34 - CAS Raw Tx 0x0D
[Aug 14 10:39:03] VERBOSE[3593] logger.c: -- Hungup 'DAHDI/34-1'
Apenas da Hangup.
O Meu contexto esta da seguinte Forma.
exten = _106XXX,1,Dial(IAX2//${EXTEN:2},90)
exten = _106XXX,n,Goto(${DIALSTATUS})
exten = _106XXX,n(BUSY),Busy(30)
exten = _106XXX,n(CHANUNAVAIL),Goto(BUSY)
exten = _106XXX,n(CONGESTION),1,Goto(BUSY)
exten = _106XXX,n,Hangup()
Abraços
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[AsteriskBrasil] Problemas com ocupado
Bom dia Lista, estou com úm ramal sip só dando ocupado, todas as chamadas que eu realizo elas dão ocupado...abaixo vai o debug, se alguem poder ajudar fico grato [2454] type=friend username=2454 accountcode=2454 regexten=2454 callerid=2401 amaflags=billing secret=xxx nat=yes dtmfmode=RFC2833 qualify=yes canreinvite=yes disallow=all allow=ulaw allow=alaw allow=gsm allow=g729 host=dynamic context=a2billing regseconds=0 cancallforward=yes --- Destroying call '[EMAIL PROTECTED]' Retransmitting #4 (NAT) to 201.22.164.167:5060: OPTIONS sip:201.22.164.167 SIP/2.0 Via: SIP/2.0/UDP 201.48.251.15:5060;branch=z9hG4bK2232d31a;rport From: Unknown sip:[EMAIL PROTECTED];tag=as1b92be04 To: sip:201.22.164.167 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Impacto Voip Pbx Max-Forwards: 70 Date: Thu, 21 Aug 2008 12:37:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Destroying call '[EMAIL PROTECTED]' asterisk1*CLI -- SIP read from 201.22.164.167:59317: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.104:44598;branch=z9hG4bK1440404089;rport Route: sip:201.48.251.15:5060;lr From: Claudio sip:[EMAIL PROTECTED];tag=1407274989 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 90 INVITE Contact: sip:[EMAIL PROTECTED]:44598 Max-Forwards: 70 Supported: replaces, path, timer User-Agent: Grandstream GXW-4004 V1.1A 1.0.0.67 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE Content-Type: application/sdp Accept: application/sdp, application/dtmf-relay Content-Length: 297 v=0 o=2454 8002 8000 IN IP4 192.168.0.104 s=SIP Call c=IN IP4 192.168.0.104 t=0 0 m=audio 18038 RTP/AVP 18 4 0 8 101 a=sendrecv a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:4 G723/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16,32-36,54 --- (15 headers 14 lines)--- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 192.168.0.104 : 44598 (NAT) Reliably Transmitting (NAT) to 201.22.164.167:59317: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.104:44598;branch=z9hG4bK1440404089;received=201.22.164.167;rport=59317 From: Claudio sip:[EMAIL PROTECTED];tag=1407274989 To: sip:[EMAIL PROTECTED];tag=as75d3af08 Call-ID: [EMAIL PROTECTED] CSeq: 90 INVITE User-Agent: Impacto Voip Pbx Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=72cfe697 Content-Length: 0 --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Found user '2454' asterisk1*CLI -- SIP read from 201.22.164.167:59317: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.104:44598;branch=z9hG4bK1440404089;rport Route: sip:201.48.251.15:5060;lr From: Claudio sip:[EMAIL PROTECTED];tag=1407274989 To: sip:[EMAIL PROTECTED];tag=as75d3af08 Call-ID: [EMAIL PROTECTED] CSeq: 90 ACK Content-Length: 0 --- (8 headers 0 lines)--- asterisk1*CLI -- SIP read from 201.22.164.167:59317: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.104:44598;branch=z9hG4bK1021032546;rport Route: sip:201.48.251.15:5060;lr From: Claudio sip:[EMAIL PROTECTED];tag=1407274989 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 91 INVITE Contact: sip:[EMAIL PROTECTED]:44598 Proxy-Authorization: Digest username=2454, realm=asterisk, nonce=72cfe697, uri=sip:[EMAIL PROTECTED], response=d223043cc27813ce35691920977491c0, algorithm=MD5 Max-Forwards: 70 Supported: replaces, path, timer User-Agent: Grandstream GXW-4004 V1.1A 1.0.0.67 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE Content-Type: application/sdp Accept: application/sdp, application/dtmf-relay Content-Length: 297 v=0 o=2454 8002 8000 IN IP4 192.168.0.104 s=SIP Call c=IN IP4 192.168.0.104 t=0 0 m=audio 18038 RTP/AVP 18 4 0 8 101 a=sendrecv a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:4 G723/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16,32-36,54 --- (16 headers 14 lines)--- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 192.168.0.104 : 44598 (NAT) Found user '2454' Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.0.104:18038 Found description format G729 Found description format G723 Found description format PCMU Found description format PCMA Found description format telephone-event Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 06230911858 in a2billing (domain 201.48.251.15) Reliably Transmitting (NAT) to 201.22.164.167:59317: SIP/2.0 404 Not Found Via:
Re: [AsteriskBrasil] Problemas com ocupado
Se usar nat=yes, o canreinvite=no deve ser especificado, não se pode usar os dois juntos. CanReinvite é usado para que o trafego de voz não passe pelo servidor, ele será feito via RTP entre os ramais, Peer to Peer ou IP para IP, como preferir, o NAT não permite que seja estabelecida uma conexão assim. - Original Message - From: Eduardo_Impacto To: asteriskbrasil@listas.asteriskbrasil.org Sent: Thursday, August 21, 2008 9:43 AM Subject: [AsteriskBrasil] Problemas com ocupado Bom dia Lista, estou com úm ramal sip só dando ocupado, todas as chamadas que eu realizo elas dão ocupado...abaixo vai o debug, se alguem poder ajudar fico grato [2454] type=friend username=2454 accountcode=2454 regexten=2454 callerid=2401 amaflags=billing secret=xxx nat=yes dtmfmode=RFC2833 qualify=yes canreinvite=yes disallow=all allow=ulaw allow=alaw allow=gsm allow=g729 host=dynamic context=a2billing regseconds=0 cancallforward=yes --- Destroying call '[EMAIL PROTECTED]' Retransmitting #4 (NAT) to 201.22.164.167:5060: OPTIONS sip:201.22.164.167 SIP/2.0 Via: SIP/2.0/UDP 201.48.251.15:5060;branch=z9hG4bK2232d31a;rport From: Unknown sip:[EMAIL PROTECTED];tag=as1b92be04 To: sip:201.22.164.167 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Impacto Voip Pbx Max-Forwards: 70 Date: Thu, 21 Aug 2008 12:37:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Destroying call '[EMAIL PROTECTED]' asterisk1*CLI -- SIP read from 201.22.164.167:59317: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.104:44598;branch=z9hG4bK1440404089;rport Route: sip:201.48.251.15:5060;lr From: Claudio sip:[EMAIL PROTECTED];tag=1407274989 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 90 INVITE Contact: sip:[EMAIL PROTECTED]:44598 Max-Forwards: 70 Supported: replaces, path, timer User-Agent: Grandstream GXW-4004 V1.1A 1.0.0.67 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE Content-Type: application/sdp Accept: application/sdp, application/dtmf-relay Content-Length: 297 v=0 o=2454 8002 8000 IN IP4 192.168.0.104 s=SIP Call c=IN IP4 192.168.0.104 t=0 0 m=audio 18038 RTP/AVP 18 4 0 8 101 a=sendrecv a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:4 G723/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16,32-36,54 --- (15 headers 14 lines)--- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 192.168.0.104 : 44598 (NAT) Reliably Transmitting (NAT) to 201.22.164.167:59317: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.104:44598;branch=z9hG4bK1440404089;received=201.22.164.167;rport=59317 From: Claudio sip:[EMAIL PROTECTED];tag=1407274989 To: sip:[EMAIL PROTECTED];tag=as75d3af08 Call-ID: [EMAIL PROTECTED] CSeq: 90 INVITE User-Agent: Impacto Voip Pbx Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=72cfe697 Content-Length: 0 --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Found user '2454' asterisk1*CLI -- SIP read from 201.22.164.167:59317: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.104:44598;branch=z9hG4bK1440404089;rport Route: sip:201.48.251.15:5060;lr From: Claudio sip:[EMAIL PROTECTED];tag=1407274989 To: sip:[EMAIL PROTECTED];tag=as75d3af08 Call-ID: [EMAIL PROTECTED] CSeq: 90 ACK Content-Length: 0 --- (8 headers 0 lines)--- asterisk1*CLI -- SIP read from 201.22.164.167:59317: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.104:44598;branch=z9hG4bK1021032546;rport Route: sip:201.48.251.15:5060;lr From: Claudio sip:[EMAIL PROTECTED];tag=1407274989 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 91 INVITE Contact: sip:[EMAIL PROTECTED]:44598 Proxy-Authorization: Digest username=2454, realm=asterisk, nonce=72cfe697, uri=sip:[EMAIL PROTECTED], response=d223043cc27813ce35691920977491c0, algorithm=MD5 Max-Forwards: 70 Supported: replaces, path, timer User-Agent: Grandstream GXW-4004 V1.1A 1.0.0.67 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE Content-Type: application/sdp Accept: application/sdp, application/dtmf-relay Content-Length: 297 v=0 o=2454 8002 8000 IN IP4 192.168.0.104 s=SIP Call c=IN IP4 192.168.0.104 t=0 0 m=audio 18038 RTP/AVP 18 4 0 8 101 a=sendrecv a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:4 G723/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16,32-36,54 --- (16 headers 14 lines)--- Using INVITE request as basis request - [EMAIL PROTECTED] Sending
Re: [AsteriskBrasil] Problemas com ocupado
Alterei a configuração ,mas continua com o mesmo problema - Original Message - From: Sebastiao Rocha To: asteriskbrasil@listas.asteriskbrasil.org Sent: Thursday, August 21, 2008 10:14 AM Subject: Re: [AsteriskBrasil] Problemas com ocupado Se usar nat=yes, o canreinvite=no deve ser especificado, não se pode usar os dois juntos. CanReinvite é usado para que o trafego de voz não passe pelo servidor, ele será feito via RTP entre os ramais, Peer to Peer ou IP para IP, como preferir, o NAT não permite que seja estabelecida uma conexão assim. - Original Message - From: Eduardo_Impacto To: asteriskbrasil@listas.asteriskbrasil.org Sent: Thursday, August 21, 2008 9:43 AM Subject: [AsteriskBrasil] Problemas com ocupado Bom dia Lista, estou com úm ramal sip só dando ocupado, todas as chamadas que eu realizo elas dão ocupado...abaixo vai o debug, se alguem poder ajudar fico grato [2454] type=friend username=2454 accountcode=2454 regexten=2454 callerid=2401 amaflags=billing secret=xxx nat=yes dtmfmode=RFC2833 qualify=yes canreinvite=yes disallow=all allow=ulaw allow=alaw allow=gsm allow=g729 host=dynamic context=a2billing regseconds=0 cancallforward=yes --- Destroying call '[EMAIL PROTECTED]' Retransmitting #4 (NAT) to 201.22.164.167:5060: OPTIONS sip:201.22.164.167 SIP/2.0 Via: SIP/2.0/UDP 201.48.251.15:5060;branch=z9hG4bK2232d31a;rport From: Unknown sip:[EMAIL PROTECTED];tag=as1b92be04 To: sip:201.22.164.167 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Impacto Voip Pbx Max-Forwards: 70 Date: Thu, 21 Aug 2008 12:37:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Destroying call '[EMAIL PROTECTED]' asterisk1*CLI -- SIP read from 201.22.164.167:59317: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.104:44598;branch=z9hG4bK1440404089;rport Route: sip:201.48.251.15:5060;lr From: Claudio sip:[EMAIL PROTECTED];tag=1407274989 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 90 INVITE Contact: sip:[EMAIL PROTECTED]:44598 Max-Forwards: 70 Supported: replaces, path, timer User-Agent: Grandstream GXW-4004 V1.1A 1.0.0.67 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE Content-Type: application/sdp Accept: application/sdp, application/dtmf-relay Content-Length: 297 v=0 o=2454 8002 8000 IN IP4 192.168.0.104 s=SIP Call c=IN IP4 192.168.0.104 t=0 0 m=audio 18038 RTP/AVP 18 4 0 8 101 a=sendrecv a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:4 G723/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16,32-36,54 --- (15 headers 14 lines)--- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 192.168.0.104 : 44598 (NAT) Reliably Transmitting (NAT) to 201.22.164.167:59317: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.104:44598;branch=z9hG4bK1440404089;received=201.22.164.167;rport=59317 From: Claudio sip:[EMAIL PROTECTED];tag=1407274989 To: sip:[EMAIL PROTECTED];tag=as75d3af08 Call-ID: [EMAIL PROTECTED] CSeq: 90 INVITE User-Agent: Impacto Voip Pbx Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=72cfe697 Content-Length: 0 --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Found user '2454' asterisk1*CLI -- SIP read from 201.22.164.167:59317: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.104:44598;branch=z9hG4bK1440404089;rport Route: sip:201.48.251.15:5060;lr From: Claudio sip:[EMAIL PROTECTED];tag=1407274989 To: sip:[EMAIL PROTECTED];tag=as75d3af08 Call-ID: [EMAIL PROTECTED] CSeq: 90 ACK Content-Length: 0 --- (8 headers 0 lines)--- asterisk1*CLI -- SIP read from 201.22.164.167:59317: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.104:44598;branch=z9hG4bK1021032546;rport Route: sip:201.48.251.15:5060;lr From: Claudio sip:[EMAIL PROTECTED];tag=1407274989 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 91 INVITE Contact: sip:[EMAIL PROTECTED]:44598 Proxy-Authorization: Digest username=2454, realm=asterisk, nonce=72cfe697, uri=sip:[EMAIL PROTECTED], response=d223043cc27813ce35691920977491c0, algorithm=MD5 Max-Forwards: 70 Supported: replaces, path, timer User-Agent: Grandstream GXW-4004 V1.1A 1.0.0.67 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE Content-Type
Re: [AsteriskBrasil] Problemas com ocupado
Eduardo, Eduardo_Impacto escreveu: Bom dia Lista, estou com úm ramal sip só dando ocupado, todas as chamadas que eu realizo elas dão ocupado...abaixo vai o debug, se alguem poder ajudar fico grato ... Looking for 06230911858 in a2billing (domain 201.48.251.15) Reliably Transmitting (NAT) to 201.22.164.167:59317: SIP/2.0 404 Not Found 06230911858 não foi encontrado no contexto a2billing. Revise seu dialplan. Leonardo ___ Compre uma camiseta da AsteriskBrasil.org! http://www.voipmania.com.br Acesse o canal IRC de discussão sobre Asterisk em Português Brasileiro na rede Freenode.net: #asterisk-br ___ Lista de discussões AsteriskBrasil.org AsteriskBrasil@listas.asteriskbrasil.org http://listas.asteriskbrasil.org/mailman/listinfo/asteriskbrasil
Re: [AsteriskBrasil] Problemas com ocupado
Obrigado pela informação encontrei o erro na agi - Original Message - From: Leonardo Gomes Figueira [EMAIL PROTECTED] To: asteriskbrasil@listas.asteriskbrasil.org Sent: Thursday, August 21, 2008 4:12 PM Subject: Re: [AsteriskBrasil] Problemas com ocupado Eduardo, Eduardo_Impacto escreveu: Bom dia Lista, estou com úm ramal sip só dando ocupado, todas as chamadas que eu realizo elas dão ocupado...abaixo vai o debug, se alguem poder ajudar fico grato ... Looking for 06230911858 in a2billing (domain 201.48.251.15) Reliably Transmitting (NAT) to 201.22.164.167:59317: SIP/2.0 404 Not Found 06230911858 não foi encontrado no contexto a2billing. Revise seu dialplan. Leonardo ___ Compre uma camiseta da AsteriskBrasil.org! http://www.voipmania.com.br Acesse o canal IRC de discussão sobre Asterisk em Português Brasileiro na rede Freenode.net: #asterisk-br ___ Lista de discussões AsteriskBrasil.org AsteriskBrasil@listas.asteriskbrasil.org http://listas.asteriskbrasil.org/mailman/listinfo/asteriskbrasil __ Informação do NOD32 IMON 3375 (20080821) __ Esta mensagem foi verificada pelo NOD32 sistema antivírus http://www.eset.com.br ___ Compre uma camiseta da AsteriskBrasil.org! http://www.voipmania.com.br Acesse o canal IRC de discussão sobre Asterisk em Português Brasileiro na rede Freenode.net: #asterisk-br ___ Lista de discussões AsteriskBrasil.org AsteriskBrasil@listas.asteriskbrasil.org http://listas.asteriskbrasil.org/mailman/listinfo/asteriskbrasil
