Making Asterisk work with your VOIP


#1

Hey guys,

This may sound dumb but how do you make your VOIP adapter and Asterisk communicate/work together?

I’m sure I could setup Asterisk but how do you make all the calls route through Asterisk?

Any detailed info, descriptions and tutorials would be appreciated.

Thanks!


#2

I would go to voip-info.org/wiki-Asterisk and do some reading.


#3

some more tutorials on this subject:

asteriskguru.com/tutorials/a … phone.html


#4

Thanks guys. I’ve been reading this already and installed Asterisk.

I understand how to setup things for outgoing calls once I hopefully get the info from my VOIP provider…

What I don’t understand is how to setup Asterisk to handle the incoming calls.

I think what you do is setup this area and context=[contextsetupbelowsomehwere]

[quote];[sip_proxy]
; For incoming calls only. Example: FWD (Free World Dialup)
;type=user
;context=from-fwd

[/quote]

Is my understanding correct ?

There are different contexts mention but none mention my provider’s name and I don’t know what the cisco ones are for. I take it I just need to setup/make a context and then say context=[mycontextsetup] ?

[quote];[cisco1]
;type=friend
;username=cisco1
;secret=blah
;qualify=200 ; Qualify peer is no more than 200ms away
;nat=yes ; This phone may be natted
; Send SIP and RTP to IP address that packet is
; received from instead of trusting SIP headers
;host=dynamic ; This device registers with us
;canreinvite=no ; Asterisk by default tries to redirect the
; RTP media stream (audio) to go directly from
; the caller to the callee. Some devices do not
; support this (especially if one of them is
; behind a NAT).
;defaultip=192.168.0.4[/quote]

Or should I use defaults?

[quote];secret=blah
;host=dynamic
;defaultip=192.168.0.4
;amaflags=default ; Choices are default, omit, billing, documentation
;accountcode=markster ; Users may be associated with an accountcode to ease billing
[/quote]

Thanks guys!


#5

you should make your own, and define it in sip.conf or iax.conf, and make the same one in extensions.conf

That way all (relative to asterisk, not to the user) incoming calls for this user, will go to this context.


#6

Thanks zoa

But I’m still confused on how to setup the context to route calls from my VOIP service to Asterisk.

If I make a context for the below how does that work? What is that defaultip for? The IP of Asterisk? Sorry but I’m still trying to wrap my ahead around this.

;secret=blah ;host=dynamic [b];defaultip=192.168.0.4[/b] ;amaflags=default ; Choices are default, omit, billing, documentation ;accountcode=markster ; Users may be associated with an accountcode to ease billing

I have a VOIP service and I can understand how to route calls out of Asterisk but I’m not sure how to setup the context so that it interacts with my VOIP service.

If you could explain more that would be great.

Thanks


#7

most of the time, the incoming calls will come in without providing a username and password and might arrive in the default context.

If you put a hostname= or defaultip in your config file for this remote server (your provide) it might match it and send it to the context you define for this “server”.


#8

Sorry zoa you went over my head there :frowning:

[i]most of the time, the incoming calls will come in without providing a username and password and might arrive in the default context.

If you put a hostname= or defaultip in your config file for this remote server (your provide) it might match it and send it to the context you define for this “server”.[/i]

Are you talking about “host=” from?

;secret=blah ;host=dynamic [b];defaultip=192.168.0.4[/b] ;amaflags=default ; Choices are default, omit, billing, documentation ;accountcode=markster ; Users may be associated with an accountcode to ease billing

Or this:

;[cisco1]
;type=friend
;username=cisco1
;secret=blah
;qualify=200 ; Qualify peer is no more than 200ms away
;nat=yes ; This phone may be natted
; Send SIP and RTP to IP address that packet is
; received from instead of trusting SIP headers
;host=dynamic ; This device registers with us
;canreinvite=no ; Asterisk by default tries to redirect the
; RTP media stream (audio) to go directly from
; the caller to the callee. Some devices do not
; support this (especially if one of them is
; behind a NAT).
;defaultip=192.168.0.4


#9

Hi guys

I can get this device to successfully make outgoing calls and I even see it is capable of receiving calls BUT IT NEVER RINGS.

I thought it wasn’t receiving anything incoming but for fun I picked up the phone even though it wasn’t ringing and a phone conversion works just fine…of course I really need my phone to actually ring so I know when to pick up :smile:

Any suggestions?

Thanks guys!


#10

Uhm I found the answer.
My phone is not rotary but it is a little older!

It seems some ATA’s or most cannot make old style phones ring even though it is touch tone.

forums.digium.com/viewtopic.php? … 1ca26ca82d


#11

Does anyone ever get this message and know how to fix it? It causes me to not be able to dial out and it happens intermittently.

thx