asterisk to opensips conversion. all help appresciated

Question

m curently working on converting an esxisting asterisk server to opensips, for better perfomance

for the most part it is working, but ive encountered an issue i cant really figure out.

asterisk is doing this :

if ("${fromourmobile}" != "") // Check if mobile Call Waiting is set to "n"
                {
                    set(phonenumber=${FROM});
                    set(GROUP()=${phonenumber});
                    noop(Group Count: ${GROUP_COUNT(${phonenumber})});
                    if (${GROUP_COUNT(${phonenumber})} > 1)
                    {
                        Busy();
                    }
                 }

and this

if (${MATH(${EPOCH} % 2)} = 0)
                {
                    set(dialhost=193.88.58.86);
                    Dial(SIP/${numbertodial}@${dialhost},60,wWtT);
                    &hangupcausecheck(${numbertodial}, ${dialhost});
                    switch (${DIALSTATUS}) 
                    { 
                        case BUSY: 
                            busy;
                            break;
                        default:
                            break;
                    }   
                    set(dialhost=195.215.252.15);
                    Dial(SIP/${numbertodial}@${dialhost},60,wWtT);
                    &hangupcausecheck(${numbertodial}, ${dialhost});
                    switch (${DIALSTATUS}) 
                    { 
                        case BUSY: 
                            busy;
                            break;
                        default:
                            break;
                    }   

i cant seem to find a similar way to do this in opensips, mostly the group_count() and the hangupcausecheck()

furthermore is there any equivalent to the $server variable from asterisk?

Answer 1

First think you need understand is

Asterisk is pbx-like software. Opensips is PROXY software.

There are no GROUPs, playback etc in Opensips. For programming opensips you have be expert in programming and FULLY understand how SIP protocol works.

Channel count can be emulated by using dialogs and caching servers. But it will not work if you config have any single error in BYE/CANCEL handling. There are no way check channel is active in most cases(becuase it not track channel's RTP data).

There is no application like Dial. Instead of that you have rewrite INVITE packet for proper destination/number.