[Asterisk-Users] test call request

[Walker Haddock]

On Mon, Nov 24, 2003 at 02:10:48PM -0200, listas iPfone wrote:
> Hi all!
> 
> We set up a sipserver using asterisk X ix66 and need some test calls from around world to verify if it is working ok.
> 
> If you can :-) please call us:
> 
> sip:ipfone at sipserver.com.br >  direct to snom200
> 
> or
> 
> sip:asterisk at sipserver.com.br > to asterisk >> snom200
> 
> Thank?s for all
> 
> Miklos

Miklos,

OK, I just dialed, looks like you answered.  However my * attempts a native bridge between my grandstream phone and your sipserver.  Do you have a suggestions on how I can set up a stanza in sip.conf so I can call you and keep * from trying a native bridge?

-->console log:

    -- Executing Dial("SIP/2400-3989", "sip/ipfone at sipserver.com.br|60") in new stack
    -- Called ipfone at sipserver.com.br
    -- SIP/sipserver.com.br-c906 is ringing
    -- SIP/sipserver.com.br-c906 is ringing
    -- SIP/sipserver.com.br-c906 is ringing
    -- SIP/sipserver.com.br-c906 is ringing
    -- SIP/sipserver.com.br-c906 answered SIP/2400-3989
    -- Attempting native bridge of SIP/2400-3989 and SIP/sipserver.com.br-c906

-->extensions.conf:

exten => 900000,1,Dial(sip/ipfone at sipserver.com.br|60)
exten => 900000,2,Hangup


Thanks, Walker
-- 
********   DataCrest, Inc. -- Technically Superior   ******************
Walker Haddock                       http://www.datacrest.com
DataCrest, Inc.                    e-mail:  wh at datacrest.com
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