Re: [asterisk-users] Voice mail and caller ID
On Fri, Jun 12, 2015 at 3:10 PM, D'Arcy J.M. Cain da...@vex.net wrote:
On Fri, 12 Jun 2015 11:49:05 -0700
John Kiniston johnkinis...@gmail.com wrote:
Try this for CHAN_SIP:
same = n,Set(Peer=${SIPCHANINFO(peername)}) ; Get the peer
same = n,Set(MailBox=${SIPPEER(${Peer},mailbox)}); Get the
mailbox same = n,VoicemailMain(${MailBox}@LocalSets,s) ; If we
have a mailbox defined log into it
Perfect. Thanks. However, I didn't bother setting a variable. I just
use it directly.
same = n,VoicemailMain(${SIPCHANINFO(peername)}@LocalSets,s)
However...
http://www.voip-info.org/wiki/view/Asterisk+func+sipchaninfo says that
SIPCHANINFO is deprecated and that we should use CHANNEL instead. I
tried that and it said pbx.c: Function CHANNEL not registered. Does
that mean that this solution will eventually fail when SIPCHANINFO is
removed in some future release? I am running 11.17.1.
No. It means that you have not loaded func_channel.so.
Richard
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Re: [asterisk-users] Voice mail and caller ID
Try this for CHAN_SIP:
same = n,Set(Peer=${SIPCHANINFO(peername)}) ; Get the peer
same = n,Set(MailBox=${SIPPEER(${Peer},mailbox)}); Get the mailbox
same = n,VoicemailMain(${MailBox}@LocalSets,s) ; If we have a
mailbox defined log into it
If you are using PJSIP it's more complex
same = n,Set(EndPoint=${CHANNEL(endpoint)}) ; Get the peer
same = n,Set(MailBox=${PJSIP_ENDPOINT(${EndPoint},mailboxes)})
same = n,ExecIf($[${ISNULL(${MailBox})} =
1]?Set(MailBox=${AST_SORCERY(res_pjsip,aor,${EndPoint},mailboxes)}))
same = n,VoicemailMain(${MailBox}@LocalSets,s) ; If we have a
mailbox defined log into it
On Fri, Jun 12, 2015 at 11:23 AM, D'Arcy J.M. Cain da...@vex.net wrote:
I have this in my sip.conf:
exten = *98,1,Verbose(0,CALLERID number is ${CALLERID(num)})
same = n,VoicemailMain(${CALLERID(num)}@LocalSets,s)
same = n,Hangup
However, my extensions are set up so that they always show the external
number, not the extension:
[foobar2](client-phone)
secret=x
callerid=Candace 551212
mailbox=foobar2@LocalSets
So the caller ID is 551212 but the voice mail is foobar2. Is there
any way to get the actual extension that called? Can I create a
variable in the extension that I can read instead of ${CALLERID(num)}?
I tried setting a random string (xaccount) and reading it with
${ENV(xaccount)} but it's not an environment variable and didn't work.
Cheers.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net
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A human being should be able to change a diaper, plan an invasion, butcher
a hog, conn a ship, design a building, write a sonnet, balance accounts,
build a wall, set a bone, comfort the dying, take orders, give orders,
cooperate, act alone, solve equations, analyze a new problem, pitch manure,
program a computer, cook a tasty meal, fight efficiently, die gallantly.
Specialization is for insects.
---Heinlein
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Re: [asterisk-users] Voice mail and caller ID
On Fri, 12 Jun 2015 11:49:05 -0700
John Kiniston johnkinis...@gmail.com wrote:
Try this for CHAN_SIP:
same = n,Set(Peer=${SIPCHANINFO(peername)}) ; Get the peer
same = n,Set(MailBox=${SIPPEER(${Peer},mailbox)}); Get the
mailbox same = n,VoicemailMain(${MailBox}@LocalSets,s) ; If we
have a mailbox defined log into it
Perfect. Thanks. However, I didn't bother setting a variable. I just
use it directly.
same = n,VoicemailMain(${SIPCHANINFO(peername)}@LocalSets,s)
However...
http://www.voip-info.org/wiki/view/Asterisk+func+sipchaninfo says that
SIPCHANINFO is deprecated and that we should use CHANNEL instead. I
tried that and it said pbx.c: Function CHANNEL not registered. Does
that mean that this solution will eventually fail when SIPCHANINFO is
removed in some future release? I am running 11.17.1.
Cheers.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net
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[asterisk-users] RES: Banco de dados interno no Asterisk e variáveis em SIP HEADERS
Prezado Fernando,
Muito obrigado por sua complementação na resposta!
Surgiram algumas dúvidas agora:
A única forma de retornar os dados num header field, como o Rafael dos Santos
Saraiva sugeriu envolve criar outro channel?
Ou seja, o que eu preciso é que a mesma execução do dia plan obtenha um valor
recebido do Sip Client, execute uma query num banco de dados e em seguida
inclua a resposta como novo hearder field na mensagem a ser enviada de resposta
ao mesmo SIP Client.
Tudo isso pode ser executado no mesmo channel? Ou seja, sem precisar fazer um
Dial() para o Sip Client?
Por exemplo:
Suponha o seguinte, o SIP client envia um SIP INVITE para o Asterisk, contendo
um novo header field na mensagem. O dia plan executa, faz o que tem que fazer,
obtem um valor de um banco de dados e em seguida inclui esse valor como novo
header field na mensagem de resposta SIP ACK 100. Ou talvez na mensagem de
resposta SIP 180 (Ringing). Isso tudo seria feito num mesmo channel? O que
estou imaginando é usar as mensagem padrões SIP, que o Asterisk já sabe
manipular, e pegar 'carona' nelas para o transporte de pequenos dados.
Algo desse tipo é possível de ser feito?
No nosso projeto usaremos SIP com TCP, não com UDP, devido a outros requisitos.
Isso facilitará o uso da ideia com Json, certo?
Atenciosamente,
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
--
Só complementando a resposta do amigo Rodrigo.
O Comando SIPAddHeader vai adicionar um cabeçalho SIP, porém no channel
atual, e o Dial, criará outro channel, o qual não irá ter o cabeçalho
que você adicionou:
Se quiser que o cabeçalho SIP customizado esteja disponivel e seja
enviado para a Ponta B que o Dial está chamando, você terá que executar
uma Macro utilizando o canal novo que será criado pelo comando Dial.
Algo do Tipo:
[header]
exten = cid,1,SIPAddHeader(X-My-Header=MYCUSTOMHEADER)
same=n,Return(1)
[meudial]
exten = _X.,Dial(SIP/X.X.X.X/${EXTEN},,b(header^cid^1))
Porém, UDP tem suas limitações, e tentar incomporar JSON a SIP Message,
imagino que não consiga ter uma ambiente de fácil manutenção.
Uma ideia seria utilizar Kamailio ou OpenSIPs o que te da mais
ferramentas para gerenciar o SIP Message.
Ou você pode utilizar seu próprio esquema utilizando um sistema de
mensagens TCP como o ZeroMQ ou o GearmanD.
Atenciosamente / Best regards / Saludos,
P Antes de imprimir pense em sua responsabilidade e compromisso com o
Meio Ambiente!
-- Mensagem original --
De: Rafael dos Santos Saraiva rafaelsnsa em gmail.com
Para: asteriskbrasil em listas.asteriskbrasil.org
asteriskbrasil em listas.asteriskbrasil.org
Enviado(s): 12/06/2015 14:53:42
Assunto: Re: [AsteriskBrasil] RES: Banco de dados interno no Asterisk e
variáveis em SIP HEADERS
Rodrigo
Segue um exemplo de manipulação do SIP HEADER:
Servidor 1:
exten = _X.,1,Answer()
same = n,SIPAddHeader(Custom-variable: valor da minha variavel)
same = n,Dial(SIP/10.68.2.43/${EXTEN},30,tT)
same = n,HangUp
Servidor 2:
exten = _X.,1,Answer()
exten = _X.,n,NoOp(${SIP_HEADER(Custom-variable)})
exten = _X.,n,goto(ura,s,1)
exten = _X.,n,HangUp
Você enviar quaisquer valores que possam ser definidos numa variável.
Neste sites você encontra maiores informações:
http://www.voip-info.org/wiki/view/Asterisk+cmd+SipAddHeader
https://wiki.asterisk.org/wiki/display/AST/Home
O Jabber trabalha com o protocolo XMPP, de mensagens instantâneas.
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[asterisk-users] Can dial plan handle new proprietary SIP HEADER fields? How?
Dear asterisk-users, I have listened that a diaplan on Asterisk can extract information from proprietary SIP messages header fields. That is, if Asterisk receives a SIP message with a modified HEADER (containing proprietary fields) , is it possible to program the dial plan to make Asterisk extract the values of such fields, being possible to handle such values in diaplan, isn't it? If it is true, is it also possible to use dial plan to make Asterisk include proprietary SIP HEADER fields in a specific SIP message? Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voice mail and caller ID
I have this in my sip.conf:
exten = *98,1,Verbose(0,CALLERID number is ${CALLERID(num)})
same = n,VoicemailMain(${CALLERID(num)}@LocalSets,s)
same = n,Hangup
However, my extensions are set up so that they always show the external
number, not the extension:
[foobar2](client-phone)
secret=x
callerid=Candace 551212
mailbox=foobar2@LocalSets
So the caller ID is 551212 but the voice mail is foobar2. Is there
any way to get the actual extension that called? Can I create a
variable in the extension that I can read instead of ${CALLERID(num)}?
I tried setting a random string (xaccount) and reading it with
${ENV(xaccount)} but it's not an environment variable and didn't work.
Cheers.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net
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