[asterisk-users] Sending SIP 183 Session Progressing

[Olle E Johansson]

16 aug 2006 kl. 07.26 skrev Dinesh Nair:

>
>
> On 08/15/06 23:30 Michael J. Tubby B.Sc (Hons) G8TIC said the  
> following:
>> I suspect your problem is with the softphone implementation...
>
> definitely, the SIP spec iianm says that UACs should play a ringing  
> tone when the 180 is received.
>
>> Occasionally calls which go from 100 -> 180 without going via the  
>> 183 result in the Cisco ringing and combined rining genrated by  
>> the telephone exchange which is weird but ok.
>
> the supplementary question then is, since i can't change the  
> softphone would i break anything if i forced the sending of the 183  
> packet anyways from within chan_sip ?
Don't do it within chan_sip, do it within the dialplan by using  
playback with the no answer option before you dial out...

You can check the user agent with a dialplan function.

/O