yup it should work, infact thats how I was routing out, but now am changing, I am not passing the calls out from asterisk to my sip provider, but sending out from SER, theproblem that I see here is, that if you sip provider will not authenticate IP addresses, but requires a user/pass , I cant see howto do that. I didnt see the point of sending the call to asterisk, unless it was because I wanted asterisk to handle the billing, since it has lots of already developed callng card apps, but since I was using mediaproxy, and all my media stream were gogin to go through SER, I felt I may as well just use asterisk for the voicemail stuff, which it seems very good at doing Iqbal Felipe Martins wrote:

Hi everyone,

I'm trying to make a call, based on, when a certain user auths my SER by mysql, if he dials "1", SER forwards the INVITE to Asterisk, The user dials "1" to open external calls and then the wanted number, doing this Asterisk forwards again to Go2call VoIP Server with a certain username and password, in order to auth my user and close the Tunnel, making the Call possible. What I want to know is that if its possible, or am I crazy !!

I thinking of doing this by the following way:

1. Configure a Certain [context] block with the lines below at extensions.conf:

a)

Context "Go2Call_Calls": Extensions Description 1 Mark

b)

exten => <external_phone_number>,1,Answer exten => <external_phone_number>,2,Playback(tt-weasels) exten => <external_phone_number>,3,VoiceMail(45) exten => <external_phone_number>,4,Hangup

2. Configure a peer at sip.conf in order to my asterisk can talk to my SER Sip Proxy:

[<external_phone_number>] type=peer username=<external_phone_number> insecure=yes canreinvite=no context=Go2Call_Calls mailbox=<external_phone_number> host=dynamic

Will it work ??? Does it work this way, or am I getting crazy ? Is there anyone who have already made that ? I'm gonna be doing this till I get our Telecom Sip Numbers to company from our Telecom Provider.

Anyone ??