Posibility of sip forking?


#1

I have search the net and this forum for days, without luck.
What i am trying to do is register two ore more sip phones with same extension.
So far when phone number two registers with asterisk replaces contact information.
SER is able to do so with no extra configuration.
The goal is: have two or more phones ring when extension is dialed and allow establish
connection with first answered phone.
As of now i was able to emulate ( sort of) with this:
exten => s,1,Dial(SIP/s,10)
exten =>s,2,Dial(SIP/s2,10)
exten =>s,3,Dial(SIP/s3,10)
exten =>s,4,Voicemail(us)

But this is not the same.
Any sugestions?


#2

You should be able to make them all ring together with:

exten => s,1,Dial(SIP/s&SIP/s2&SIP/s3,10)
exten => s,2,Voicemail(us)


#3

Thank you.
This configuration is working for receiving calls but this is not what i was hopping for.
I would prefer to be able to register more the one phone/sip client with asterisk.
This will give me corect caller ID information and ability to check voice mail ( with setup: (s${CALLERIDNUM} ))
It would also work with unlimited number of phones for i.e. sales or customer service with single exension. Is the registration like this possible?
I would like to also be able to setup same sip extension to be host=1.2.3.4 and host=dynamic.
in case of fxs on 2600 with static and ata18x or 79xx with dynamic IP address.


#4

Yeah, i understand the difference. I don’t know if it’s possible to do what you want or not - probably not. What happens when you try it?

The allowguest option in sip.conf may help you do something similar to what you want to do - combined with firewall rules to allow you to restrict access, if necessary. There doesn’t seem to be any documentation about how guest accounts work on voip-info though.


#5

Asterisk only allows one endpoint per ‘extension’ for SIP.


#6

"Asterisk only allows one endpoint per ‘extension’ for SIP."
And how many for h323 or iax?
Is there a way to change this?


#7

[quote=“trzeci”]"Asterisk only allows one endpoint per ‘extension’ for SIP."
And how many for h323 or iax?
Is there a way to change this?[/quote]

You would have to do your own development within Asterisk, there is no way to change this from a configuration perspective.


#8

You could plonk the sales extensions into a queue.
Then route inbound calls into the queue. Any extension that is logged in into the queue would get the call.

What helped me get to grips with all sorts of strategies, was installing Asterisk@home, then set things up using the web interface, then look in the various config files to check what’s under the hood.
Ring-groups, queues… there are a couple of ways to achieve what you’re after.

Bye,
Leo