Re: [AsteriskBrasil] Asterisk com API

2017-03-17 Por tôpico Sylvio Jollenbeck
Posso ajudar...

Em 17 de mar de 2017 23:54, "Guilherme Alano" 
escreveu:

> Ola pessoal,
>
>
> Alguém poderia me dar um norte, para uma integração com uma API Json.
>
>
> No caso tenho que integrar um CRM com o asterisk através da API.
>
>
> Alguém trabalha com este tipo de customização/serviço?
>
>
> Obrigado.
>
> Att.,
>
> Guilherme Alano
>
> ___
> KHOMP: completa linha de placas externas FXO, FXS, GSM e E1
> Media Gateways de 1 a 64 E1s para SIP com R2, ISDN e SS7
> Intercomunicador e acesso remoto via rede IP e telefones IP
> Conheça todo o portfólio em www.Khomp.com
> ___
> Para remover seu email desta lista, basta enviar um email em branco para
> asteriskbrasil-unsubscr...@listas.asteriskbrasil.org
>
___
KHOMP: completa linha de placas externas FXO, FXS, GSM e E1
Media Gateways de 1 a 64 E1s para SIP com R2, ISDN e SS7
Intercomunicador e acesso remoto via rede IP e telefones IP
Conheça todo o portfólio em www.Khomp.com
___
Para remover seu email desta lista, basta enviar um email em branco para 
asteriskbrasil-unsubscr...@listas.asteriskbrasil.org

[AsteriskBrasil] Asterisk com API

2017-03-17 Por tôpico Guilherme Alano
Ola pessoal,


Alguém poderia me dar um norte, para uma integração com uma API Json.


No caso tenho que integrar um CRM com o asterisk através da API.


Alguém trabalha com este tipo de customização/serviço?


Obrigado.

Att.,

Guilherme Alano
___
KHOMP: completa linha de placas externas FXO, FXS, GSM e E1
Media Gateways de 1 a 64 E1s para SIP com R2, ISDN e SS7
Intercomunicador e acesso remoto via rede IP e telefones IP
Conheça todo o portfólio em www.Khomp.com
___
Para remover seu email desta lista, basta enviar um email em branco para 
asteriskbrasil-unsubscr...@listas.asteriskbrasil.org

Letter of Introduction

2017-03-17 Por tôpico Shruti Engineering
*Letter of Introduction*

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*, **Brake Liner*
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*, AC
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 Thanking you in anticipation

Yours Faithfully


 
*Thanks & Regards*
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*Shruti Engineering  *
 1&7 Hira Estate,GIDC Vtava,Ahmedabad  Gujrat  ( India)
 Contact No- (+91) 8264261548,9723348428
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*E-mail :-* shrutieng...@gmail.com ,  shrutieng2...@gmail.com ,
[image: http://shrutiengineering.com/] [image:
http://shrutiengineering.com/] 


Re: [AsteriskBrasil] Received response: "Forbidden" em um Grandstream HT-503

2017-03-17 Por tôpico Vitor Mazuco
Olá a todos, obrigado pela ajuda.

Eu vou seguir os seus conselhos, mas alguém ja teve esse problema?

É estranho, pq eu consigo fazer ligação, mas não consigo receber
ligação vindo do Asterisk.

Abraços.

Em 16/03/2017, Alex Freitas escreveu:
> segue tutorial
>
>
>
> 2017-03-16 14:13 GMT-03:00 Luciano Cavalcante Souza :
>
>> verifica os tutorias da loja mundi
>>
>>
>> *Sds,*
>>
>> *Luciano Cavalcante Souza*
>> *Tecnólogo em Gestão da Tecnologia da Informação*
>> *Mobile: + 55 79 98814.5895 <(79)%2098814-5895>(vivo)*
>> *e-mail: lucin...@gmail.com *
>> *Perfil no Linkdin *
>>
>> *Sobre o Google Apps: Google Apps *
>>
>> Concentre-se nos pontos FORTES, reconheça as FRAQUEZAS, agarre as
>> OPORTUNIDADES e proteja-se contra as AMEAÇAS.
>>
>> 2017-03-16 12:00 GMT-03:00 Vitor Mazuco :
>>
>>> Ola a todos!
>>>
>>> Estou com um problema de ligação, não estou conseguindo receber
>>> ligações de meu ASTERISK para o meu FXO Grandstream.
>>>
>>> Ele dá erro de "Forbidden" from" conforme as msg abaixo.
>>>
>>> Já tentei de tudo, mas nao acho o problema.
>>>
>>> Lembrando que eu uso um LOAD BALANCE nesse Grandstream para fazer o
>>> balanceador de rede.
>>>
>>> Seria esse um problema de NAT/Firewall?
>>>
>>> Obrigado quem puder me ajudar.
>>>
>>> Log na CLI:
>>>
>>> Using SIP RTP CoS mark 5
>>> -- Executing [27100@ramais:1] MixMonitor("SIP/2000-bd8b",
>>> "/media/HDExterno/gravacoes/feitas/APTO/27100/1489675579.120546.wav")
>>> in new stack
>>> -- Executing [27100@ramais:2] Dial("SIP/2000-bd8b",
>>> "SIP/136/100,60,tT") in new stack
>>>   == Begin MixMonitor Recording SIP/2000-bd8b
>>>   == Using SIP RTP CoS mark 5
>>> -- Called SIP/136/100
>>> [2017-03-16 11:46:19] WARNING[1554][C-98b9]: chan_sip.c:23843
>>> handle_response_invite: Received response: "Forbidden" from
>>> ';tag=as57804b2e'
>>>   == Everyone is busy/congested at this time (1:0/0/1)
>>> -- Auto fallthrough, channel 'SIP/2000-bd8b' status is
>>> 'CHANUNAVAIL'
>>>   == MixMonitor close filestream (mixed)
>>>   == End MixMonitor Recording SIP/2000-bd8b
>>>
>>>
>>> SIP Debuug:
>>>
>>> Called SIP/136/100
>>>
>>> <--- SIP read from UDP:192.168.25.169:3329 --->
>>> SIP/2.0 100 Trying
>>> Via: SIP/2.0/UDP 192.168.25.24:5089;branch=z9hG4bK4433efa4;rport=5089
>>> From: ;tag=as62bede9e
>>> To: 
>>> Call-ID: 692c5d293d0fae0853872d6a3206af86@192.168.25.24:5089
>>> CSeq: 102 INVITE
>>> Supported: replaces, path, timer, eventlist
>>> User-Agent: Grandstream HT-503 V2.0A 1.0.14.1 chip V2.2
>>> Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO,
>>> REFER,
>>> UPDATE
>>> Content-Length: 0
>>>
>>> <->
>>> --- (10 headers 0 lines) ---
>>>
>>> <--- SIP read from UDP:192.168.25.169:3329 --->
>>> SIP/2.0 403 Forbidden
>>> Via: SIP/2.0/UDP 192.168.25.24:5089;branch=z9hG4bK4433efa4;rport=5089
>>> From: ;tag=as62bede9e
>>> To: ;tag=1820807938
>>> Call-ID: 692c5d293d0fae0853872d6a3206af86@192.168.25.24:5089
>>> CSeq: 102 INVITE
>>> Supported: replaces, path, timer, eventlist
>>> User-Agent: Grandstream HT-503 V2.0A 1.0.14.1 chip V2.2
>>> Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO,
>>> REFER,
>>> UPDATE
>>> Content-Length: 0
>>>
>>> <->
>>> --- (10 headers 0 lines) ---
>>> Transmitting (NAT) to 192.168.25.169:3329:
>>> ACK sip:100@192.168.25.169 SIP/2.0
>>> Via: SIP/2.0/UDP 192.168.25.24:5089;branch=z9hG4bK4433efa4;rport
>>> Max-Forwards: 70
>>> From: ;tag=as62bede9e
>>> To: ;tag=1820807938
>>> Contact: 
>>> Call-ID: 692c5d293d0fae0853872d6a3206af86@192.168.25.24:5089
>>> CSeq: 102 ACK
>>> User-Agent: Asterisk PBX 13.10.0
>>> Content-Length: 0
>>>
>>>
>>> ---
>>> [2017-03-16 11:34:53] WARNING[1554][C-98af]: chan_sip.c:23843
>>> handle_response_invite: Received response: "Forbidden" from
>>> ';tag=as62bede9e'
>>> Scheduling destruction of SIP dialog
>>> '692c5d293d0fae0853872d6a3206af86@192.168.25.24:5089' in 6400 ms
>>> (Method: INVITE)
>>>   == Everyone is busy/congested at this time (1:0/0/1)
>>> -- Auto fallthrough, channel 'SIP/2000-bd7a' status is
>>> 'CHANUNAVAIL'
>>>   == MixMonitor close filestream (mixed)
>>>   == End MixMonitor Recording SIP/2000-bd7a
>>>
>>> See my sip.conf
>>>
>>> ;;
>>> [136]
>>> type=friend
>>> defaultuser=136
>>> secret=X
>>> qualify=yes
>>> ;nat=no
>>> nat=force_rport,comedia
>>> context=ramais
>>> ;insecure=invite,port
>>> disallow=all
>>> allow=ulaw,alaw,gsm
>>> host=dynamic
>>> canreinvite=no
>>> regext=136
>>> callgroup=1
>>> pickupgroup=1
>>> ___
>>> KHOMP: