[Asterisk-Users] test call request

[WipeOut]

Walker Haddock wrote:

>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
>  
>
Adding "canreinvite=no" to your sip.conf for that phone should do it..

Later..