Re: [asterisk-users] Asterisk Behind Firewall

[Madushan Geethanga]

Hi,

I have used a sonicwall  Firewall, it has a sip transformation feature. It
is necessary to use a firewall to protect your server

Best Regards,
Madushan

On Mon, Jan 4, 2016 at 11:45 PM, IPN Comm  wrote:

> I was wondering if anyone can give me any pointers or insights of whether
> or not to have an asterisk server behind a firewall.
>
> I have always ran Asterisk on a public IP but was wondering if I should
> move it to a local IP behind a firewall.
>
> I am looking to set up a location with 300 SIP phones.
>
> Normally, I would put the Asterisk server on one public IP and let the SIP
> phones get DHCP from a router on a different IP and they would register to
> the Public Asterisk server from that IP address.
>
> Should I move the asterisk server behind the same router?
>
> If so, how should the server be set up and what is the best
> router/firewall hardware to accomplish this environment?
>
> Thanks,
> -H
>
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[asterisk-users] Forwarding call if extension busy

[Luca Bertoncello]

Hi and happy new year!

My question:

- two extensions:  and 
- an active call on 
- incoming calls to  should be forwarded to  (call advice!) and 

I know how can I forward an incoming call to more than an extension,  
but I have no idea how can I get the information, that  has  
already an active call...


I think, I need something like:

exten => _,1,Verbose(2,Incoming call for  - [${CALLERID(num)}])
exten => _,n,GotoIf(  ?busy)
exten => _,n,Dial(SIP/,19,RcxX)
exten => _,n,VoiceMail(,us)
exten => _,n,Hangup
exten => _(busy),n,Dial(SIP//,19,RcxX)
exten => _,n,VoiceMail(,us)
exten => _,n,Hangup

Well, the problem is the second line, of course...

Of course the extension  is NOT "really busy", since the phone can  
support more active channels, but I hope I explained my problem...


Any suggestion?

Thanks
Luca Bertoncello
(lucab...@lucabert.de)


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Re: [asterisk-users] Forwarding call if extension busy

[Luca Bertoncello]

Zitat von Kevin Larsen :

Hi Kevin


For instance, my customer service department has this thing against ever
having their phone ring a call while they are already on a call, so for
these special little snowflakes, I have  the following line:

same => n(voice),GotoIf($["${DEVICE_STATE(sip/${EXTEN})}" !=
"NOT_INUSE"]?voicebusy)


I think, this is EXACTLY what I'm looking for...

I'll try this evening!

Thanks
Luca Bertoncello
(lucab...@lucabert.de)


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Re: [asterisk-users] Forwarding call if extension busy

[jg]

Hi and happy new year!

My question:

- two extensions:  and 
- an active call on 
- incoming calls to  should be forwarded to  (call advice!) and 

I know how can I forward an incoming call to more than an extension, but I have no idea how 
can I get the information, that  has already an active call...


I think, I need something like:

exten => _,1,Verbose(2,Incoming call for  - [${CALLERID(num)}])
exten => _,n,GotoIf(  ?busy)
exten => _,n,Dial(SIP/,19,RcxX)
exten => _,n,VoiceMail(,us)
exten => _,n,Hangup
exten => _(busy),n,Dial(SIP//,19,RcxX)
exten => _,n,VoiceMail(,us)
exten => _,n,Hangup

Well, the problem is the second line, of course...

Of course the extension  is NOT "really busy", since the phone can support more active 
channels, but I hope I explained my problem...


Any suggestion?

Thanks
Luca Bertoncello
(lucab...@lucabert.de)


There may not be a general solution as the end points can accept more than a single call 
themselves as described by yourself, i.e. the phone may not be in a busy state unless the max. 
number of calls has been reached or a call has been actively rejected. In that case you might 
put another Dial just after the first Dial application. If there is still no answer, VoiceMail 
gets called.


You need to configure your phone to accept only a single call. Another approach would be to 
check from within Asterisk whether a particular endpoint has already active calls and Dial() as 
required, i.e. one would delete the phones with active calls from a given list. Since there is 
no real "busy" condition, this seems to be a cleaner approach.


At first you should be able to describe exactly which behavior you want.

jg

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Re: [asterisk-users] Forwarding call if extension busy

[Kevin Larsen]

asterisk-users-boun...@lists.digium.com wrote on 01/04/2016 08:55:40 AM:

> My question:
> 
> - two extensions:  and 
> - an active call on 
> - incoming calls to  should be forwarded to  (call advice!) and 

> 
> I know how can I forward an incoming call to more than an extension, 
> but I have no idea how can I get the information, that  has 
> already an active call...
> 

I am not sure if I completely understand what you are trying to do, but it 
sounds like you want to query the DEVICE_STATE function.

For instance, my customer service department has this thing against ever 
having their phone ring a call while they are already on a call, so for 
these special little snowflakes, I have  the following line:

same => n(voice),GotoIf($["${DEVICE_STATE(sip/${EXTEN})}" != 
"NOT_INUSE"]?voicebusy)

Basically, this little line looks at the extension and if it shows 
anything other than free (NOT_INUSE), it jumps to the voicebusy line in 
the dialplan. The voicebusy line just hits voicemail directly.

You can use this same idea to branch your logic and handle a variety of 
situations. In my case, I only want to actually perform the dial if the 
phone is currently not in use, so my logic was fairly simple.

See here for reference:
https://wiki.asterisk.org/wiki/display/AST/Device+State

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Re: [asterisk-users] Forwarding call if extension busy

[Luca Bertoncello]

Kevin Larsen  schrieb:

> I am not sure if I completely understand what you are trying to do, but it 
> sounds like you want to query the DEVICE_STATE function.

IT WORKS

Thank you very much!

Luca Bertoncello
(lucab...@lucabert.de)

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Re: [asterisk-users] Forwarding call if extension busy

[Kevin Larsen]

> Kevin Larsen  schrieb:
> 
> > I am not sure if I completely understand what you are trying to do, 
but it 
> > sounds like you want to query the DEVICE_STATE function.
> 
> IT WORKS
> 
> Thank you very much!
> 

Glad I was able to help. You are most welcome.

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[asterisk-users] Asterisk Behind Firewall

[IPN Comm]

I was wondering if anyone can give me any pointers or insights of whether
or not to have an asterisk server behind a firewall.

I have always ran Asterisk on a public IP but was wondering if I should
move it to a local IP behind a firewall.

I am looking to set up a location with 300 SIP phones.

Normally, I would put the Asterisk server on one public IP and let the SIP
phones get DHCP from a router on a different IP and they would register to
the Public Asterisk server from that IP address.

Should I move the asterisk server behind the same router?

If so, how should the server be set up and what is the best router/firewall
hardware to accomplish this environment?

Thanks,
-H
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Re: [asterisk-users] Asterisk Behind Firewall

[Ron Wheeler]

Both work.
If you have enough IP addresses to dedicate one to your Asterisk server, 
that removes one node in the path from the world.
You will need a firewall on the Asterisk server to protect it from 
outside meddling.
If you can put the Asterisk server on the same network as the SIP 
devices (using a second NIC) that should help performance.


Is the SIP network on the same network as your internet/data LAN?

Ron

On 04/01/2016 1:15 PM, IPN Comm wrote:
I was wondering if anyone can give me any pointers or insights of 
whether or not to have an asterisk server behind a firewall.


I have always ran Asterisk on a public IP but was wondering if I 
should move it to a local IP behind a firewall.


I am looking to set up a location with 300 SIP phones.

Normally, I would put the Asterisk server on one public IP and let the 
SIP phones get DHCP from a router on a different IP and they would 
register to the Public Asterisk server from that IP address.


Should I move the asterisk server behind the same router?

If so, how should the server be set up and what is the best 
router/firewall hardware to accomplish this environment?


Thanks,
-H





--
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President
Artifact Software Inc
email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102

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