[OpenSIPS-Users] Asterisk call features

Bogdan-Andrei Iancu bogdan at voice-system.ro
Wed May 12 09:44:48 CEST 2010


Hi Toqeer,

Simply detect in opensips route script the dial codes and send such 
calls to Asterisk:

    if ($rU=~"^[1-9][0-9]$") {
          rewritehostport("Asterisk_Ip:Asterisk_port");
          t_relay();
          exit()
    }

Put the above block somewhere before the lookup(location) part.

Regards,
Bogdan


toqeer ali wrote:
> Currently i have a setup opensips/asterisk/a2billing cluster. Is this 
> possible in opensips  to pass standard asterisk feature codes to 
> asterisk from openips so the users of  asterisk can use of message 
> center, call forwarding, call transfer, 3 way calling, return call, 
> and other standard calling features available at the dial prompt.
>
> Our network is all static IP based with dedicated servers 
> (1) Opensips    ALL ENPOINTS REGISTER WITH OPENSIPS 
> (2) Asterisk Media Server 
>  
>
> For i can  register and make calls but have no features at the dial 
> prompt.  i want to use features of asterisk.
>
>     * 69 -  Last Number Called
>     * 70 -  Activate Call Waiting
>     * 71 -  Deactivate Call Waiting
>     * 72 -  Call Forwarding
>     * 73 -  Deactivate Call Forwarding
>     * 78 - Enable Do not disturb
>     * 79 - Disable Do not disturb
>     * 90 -  Call Forward Busy
>     * 91 - Disable call forward busy
>     * 97 -  Message Center
>     * 98 -  Enter Message Center
>
> Any help will highly appreciated.
>
> -- 
> Toqeer Ali Syed
>
> Red Hat Certified Engineer
> mob:     +92 321 9059916
> ------------------------------------------------------------------------
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
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>   


-- 
Bogdan-Andrei Iancu
www.voice-system.ro




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