[OpenSIPS-Users] help with asterisk - opensips transition

[Mike Claudi Pedersen]

im 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?

i hope you guys can help me, and thx for alle the help ive gotten so far
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.opensips.org/pipermail/users/attachments/20140526/60d4270c/attachment.htm>