[Asterisk-Users] new here : logic of ser and asterisk all confused---longish

[Asterisk .]

Hello,

comments inline...

--- Iqbal <iqbal at gigo.co.uk> wrote:
> if (uri =~ "sip:2[0-9]*@sip.ipclouds.co.uk"){
> 
>         log(1, "Forwarding to Asterisk\n");
>         rewritehostport("193.218.160.25:5090");
> 
>         break;
>     }
> 
> which I think means any number starting with a 2 send to asterisk
> server..now when I dial this , in the SER logs it shows the message
> Forwarding to Asterisk, and then waits, but in asterisk sip debug there
> is nothing, not a sausage

I am afraid you are not sending the calls to Asterisk, but just rewriting the host and port. After
rewriting, forward/relay the calls to Asterisk.

> SERADDRESS=sip.ipclouds.co.uk:5060
> 
> [OUTGOING]
> ; Line below added for ser --- iqbal
> exten => 1000,1,Dial(SIP/${EXTEN}@${SERADDRESS},20,r)
> 
> seeing all this it would seems that asterisk and ser go into a loop,
> cause extensions simply sends it back to SER, which is what seems to
> happen, and the ser.cfg sends it back to extensions.

Dont Asterisk complain about a '482 Loop Detected' error? The Dial statement will create a new
INVITE and will be relayed to SER, which will send it back to Asterisk, thus resulting in a loop.
Asterisk will drop this call. For dialing extensions use either Asterisk or SER. IMO, use ser for
all extension dialing, and have appropriate forwarding and failure routing in the ser.cfg to send
calls to Asterisk for the PBX features and voicemail.

Regards, Girish



		
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