[Asterisk-Users] Question receiving calls via SIP

[Steven Kokinos]

Hello-

I am in the process of adding a new provider to my asterisk box (both 
for outbound termination as well as inbound DID). They are going to be 
delivering and receiving traffic via SIP only.

Now, in IAX via Voicepulse or others I know that I can simply have one 
registration statement along with an inbound context, then in 
extension.conf map the outbound context.

from iax.conf:

register => in-xxxxxxxx:xxxxxxxx at gw5.voicepulse.com

from extensions.conf
[voicepulse-in]
exten => 212xxxxxxx,1,Dial(${PHONES1}&${PHONES2},30)
exten => 212xxxxxxx,2,Voicemail2(u${PHONES1VM})
exten => 212xxxxxxx,3,Hangup

I know this way I only have to register once, but can receive calls on 
several inbound DID numbers without any problem, provided they are all 
mapped similar to what I have above within extensions.conf.

My question is whether or not the same thing will work for a sip 
provider, as it will be pretty cumbersome from a networking standpoint 
to have a registration statement for each DID (as opposed to simply 
having a new extension statement).  In the syntax of the sip 
registration statement it appears to always end with the /extension 
that is supposed to be associated with that account.

Can the /extension be left off of the statement in sip.conf, and picked 
up in the same way DID's are used in iax.conf? If not, why are there 
separate methods for registering each protocol, would seem cleaner to 
have a consistent way of dealing with this.

Regards,

-Steve