Call routing for sudo provider system


#1

Hello all,

I currently have asterisk 1.6 installed on a Ubuntu server that I am using as a sudo internal SIP proxy for a lab environment.

I am new to Asterisk and have stripped out or overlooked a lot of function in order to get a simple 7 digit dialing proxy for lab use.

I currently have calls between endpoints working correctly but I am having to register each line individually in order to route the call. This causes multiple registration attempts from an ISR to register and make 10 lines active.

I have attempted to route multiple destinations to a single user but in doing this the TO: address is changed and I cannot route the call correctly on the far end.

I have searched a lot on this issue, but because it’s a little different I am having a hard time finding direct answers. Does anyone have a call routing suggestion I could try to accomplish this.

Ultimate goal is to forward numbers 5551000-5551100 to a single registration keeping the TO: field as the original called number, allowing me to route multiple numbers to a single registration.

Currently by extension.com and sip.conf are setup as follows:

exten => _555XXXX,1,Dial(SIP/${EXTEN})

[5553200]
type=friend
defaultuser=5553200
secret=benchb
host=dynamic
context=default


#2

Anybody??


#3

I don’t understand “sudo” in this context.

Asterisk is a back to back user agent, not a proxy.


#4

sudo as in not real, fake.

I’m trying to get it to act as a proxy, or at least feel like a proxy. The call routing and features work great, but I am wanting to get the dial-plan fixed so I can route multiple extensions to a single registry.

I’ve taken a quick look at OpenSER and actually running a full on proxy but the learning curve seems pretty steep for what I am needing it for.

Any suggestions would be appreciated.


#5

You meant “pseudo”.


#6

yes I did mean pseudo. too much linux, not enough grammar.

I’m attempting to give opensips a try


#7

I’m not sure, but I think you’re talking about Dial(SIP/peer/${EXTEN}).


#8

Bingo.

That is exactly what I am looking for. Thank you very much as i have spent the last two days on failing opensips installs.


#9

Glad I could help. Regarding dialplan applications you could use “core show applications” and "core show application " from the Asterisk CLI to find quite useful information.