It;s the 'sâtandard destination which is where every âunmatchedâ call will go, do you have anonymous or guest calls allowed?
When Asterisk receives a SIP call with no user (phone number) field, it starts the call at extension s (I think that is for start). The most common reason for that is that the call is from a provider and you have registered with them without specifying a user part for your contact address.
Normally, you should set the contact user to the DID, in such cases, although I think things will work if you only have one DID and you treat the DID has being called âsâ.
Asterisk didnât start as a SIP PABX, and, for analogue calls, it is possible to have the dialplan handle everything, including production of dial tone, so, if you set up Asterisk to start the call immediately the line is seized, for an analogue call, it will start in the s extension from where it can generated dialtone and wait for the dialled number.
do you recognize these 5NNNN numbers?
yes, those extension I used , I thought at first its cuz when i called to the queue they ring all agent but just one answer will show in dest but now I noticed its a responde time and doesnt show who Answered just S
This topic was automatically closed 31 days after the last reply. New replies are no longer allowed.