Cancelado

voip: sip application

Hello,

I am trying to do a click to speak service in compliance with RFC 3515 using the SIP protocol.

In this scenario, I have to connect a telephone number A to a predefined telephone number B towards a web interface.

To do this, I use Asterisk server (version 1.2.7.1).

I configured my dialplan like this (I'll show just what's interesting in it):

[default]

exten => _X.,1,Dial(SIP/${EXTEN}@mysipprovider,60,G(default^1^1)

;I use sipdiscount as SIP provider defined in [url removed, login to view]

exten => 1,1,Hangup()

exten => 1,2,Transfer(SIP/${NUMB}@mysipprovider)

;where NUMB is the predefined number to dial

Note that the G option in the first line of the context means that:

when the call is answered, the caller goes to the context default, extension 1 with priority 1;

the callee goes to the context default, extension 1 with priority 2.

I tested this with a softphone named twinkle.

Dialing the telephone number A from twinkle, Asterisk connects the two numbers, that is, number A and twinkle.

When the call is setup, twinkle hangups and, in the same time, I see a REFER message send to number A from my sipdiscount account with the Header field Refer-To: Number B.

Unfortunately, my sip provider, that is, [url removed, login to view] , does not allow this method and so it returns to me a message 405 "Method Not Allowed".

I have tried to use another sip carrier, like [url removed, login to view], but it also does not accept REFER method.

do u know a voip provider that permit me to use "refer"?

or

do u know how solve this problem?

if so, give me your quote.

Thanks

Habilidades: Serviços de audio, Engenharia, Python, Administrador do Sistema, Wireless

Ver mais: web application quote, voip web server, fun sip application, what application, VOIP, voip server, softphone, sip softphone, sip server, sip provider, sip dial, rfc, context , compliance call, audio voip, audio python, telephone application, connect time server, message asterisk, python web application, tried, voip web sip, interesting application, asterisk extension number, asterisk sip softphone

Acerca do Empregador:
( 0 comentários ) bari, Italy

ID do Projeto: #130058

4 freelancers estão ofertando em média $300 para este trabalho

kias

Hi , Your experiment looks very intresting. If you are willing we can both work together to sort out. I m not interested in money. I know few providers , but still needs to be tested . Please advise . Regards, Mais

$300 USD in 10 dias
(3 Comentários)
1.4
saravananvoip

Hi,sir, What is the probel for my debugging your issues i am trying your problem till now i will reach you soon wiht solution. Thanks, saravanan.s

$300 USD in 5 dias
(0 Comentários)
0.0
Expdeveloper

Please check PMB.

$300 USD in 15 dias
(0 Comentários)
0.0
st41ker

Check my PMB.

$300 USD in 0 dias
(0 Comentários)
0.0