Music on Hold through remote server

hi
I want to play music on hold from one asterisk while SIP messagineg should be done through another asterisk.

my call flow should be like this.

   SIPUA  (A)............................Asterisk..................... .... SIPUA (B)......................Asterisk 
     <-----------------------------RTP Session ---------------------->
     ---------INVITE(Hold)------------------->
     <----------------200 OK-----------------
      ..........................................................---------------------------Invite (sdp of B)-------->                           
     .............................................................<---------------------------200 OK----------------
     ----------------ACK--------------------->
     ........................................................... ---------------------------ACK--------------------->
     ............................................................................................... <------MOH --------------
     ---------INVITE(UnHold)------------------->
     <----------------200 OK-----------------
     ............................................................ -----------------------BYE------------------------>                           
     ............................................................<------------------------200 OK----------------
     ----------------ACK--------------------->
                                                       
   <-----------------------------RTP Session ---------------------->       

How can we do this …
help me to sortout this problem

if i understand you correctly, you want to have one * box managing your system, but another box providing MOH, and you want to connect users to the 2nd box with reinvites?

I don’t think this is possible.

However you can get MOH from any arbitrary source via *, check out musiconhold.conf. Theoretically any console app that can spit out audio in the right format will work…

But if it can possible then it reduce asterisk burdon…

IT Will Reduce asterisk burdon that’s why it is very important

I don’t believe asterisk has this capability built in. You might be able to get something like this by putting OpenSER in front of * (a common setup when dealing with high volumes)…