Dear All,
I have the following setup;
SER/External Asterisk <--> Firwall <--> Internal Asterisk
<-VPN-> Users
At the moment;
Anybody can register with our SER proxy and call each other using VoIP.
Anybody can call one of our internal users via our SER/Asterisk gateway.
The INVITE is sent to our external Asterisk Server, this act as a UA and
uses IAX2 to send the call to our internal Asterisk server.
Our internal users use a VPN to connect to our corporate HQ. They
register with our Internal Asterisk server and can make internal and
PSTN calls.
What I would like to do is to redirect external SIP calls to our
external Asterisk server. e.g if I call sip:shad@voipdomain.org I would
like the call to be routed from our Internal Asterisk server to our
External Asterisk server via IAX2 and for the external asterisk server
to act as a UA and make the call.
I have tried the following syntax on our internal server;
exten => _sip.,1,Dial(IAX2/bxxxxxx:yyyyyy@192.X.y.x/${EXTEN})
However this does not seem to work?
How do I change my dial plan so that SIP calls are routed from my
internal Asterisk box to my external Asterisk box over IAX2?
Warm Regards and Thanks
Shad Mortazavi
---------------
Nexus Management Inc
Shad Mortazavi wrote:> What I would like to do is to redirect external SIP calls to our > external Asterisk server. e.g if I call sip:shad@voipdomain.org I would > like the call to be routed from our Internal Asterisk server to our > External Asterisk server via IAX2 and for the external asterisk server > to act as a UA and make the call. > > I have tried the following syntax on our internal server; > > exten => _sip.,1,Dial(IAX2/bxxxxxx:yyyyyy@192.X.y.x/${EXTEN}) > > However this does not seem to work?Have you tried this? exten => shad,1,Dial(IAX2/bxxxxxx:yyyyyy@192.X.y.x/${EXTEN})
I believe that they covered this exact procedures at www.voip-info.org. Look for the topic on connecting two Asterisk servers. They outline three different ways that you can do so.>From: "Eric \"ManxPower\" Wieling" <eric@fnords.org> >Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion ><asterisk-users@lists.digium.com> >To: Asterisk Users Mailing List - Non-Commercial Discussion ><asterisk-users@lists.digium.com> >Subject: Re: [Asterisk-Users] Routing SIP calls via URI >Date: Wed, 29 Mar 2006 13:18:07 -0600 > >Shad Mortazavi wrote: > >>What I would like to do is to redirect external SIP calls to our >>external Asterisk server. e.g if I call sip:shad@voipdomain.org I would >>like the call to be routed from our Internal Asterisk server to our >>External Asterisk server via IAX2 and for the external asterisk server >>to act as a UA and make the call. >> >>I have tried the following syntax on our internal server; >> >>exten => _sip.,1,Dial(IAX2/bxxxxxx:yyyyyy@192.X.y.x/${EXTEN}) >> >>However this does not seem to work? > >Have you tried this? > >exten => shad,1,Dial(IAX2/bxxxxxx:yyyyyy@192.X.y.x/${EXTEN}) >_______________________________________________ >--Bandwidth and Colocation provided by Easynews.com -- > >Asterisk-Users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
Dear Group;
I can confirm that I have read through the three examples in
www.voip-info.org.
These examples are excellent and address a couple of the questions. I
have IAX2 working between several asterisk servers on our VPN and
between the DMZ and our LAN.
Also
exten => shad,1,Dial(IAX2/bxxxxxx:yyyyyy@192.X.y.x/${EXTEN})
This answers part of the question;
However what I want to do is to send any outbound sip calls via our
external SIP server.
i.e;
VPN LAN IAX2 DMZ Internet
Internal UA <-------> Internal (*) <------> External
(*)<------>
ExternalUA
We have an extensive internal dial plan, X dial the UK, Y dial USA, 1XXX
for Voicemail, 2xxx for Meetme, etc.
Do I need to setup a prefix to dial the internet? And then route all
calls to the External(*) based on this prefix?
Thanks
Shad Mortazavi
------------------------------
Nexus Group Technical Manager
n|m Nexus Management Inc
Dear Group;
I am closer to where I want to be. I could still do with some help.
For my Internal(*)I setup the following;
extensions.conf
---------------
[SIPOUT]
exten => _6.,1,Dial(SIP/${EXTEN:1}@192.y.x.1x0)
If I dial sip:6shad@blablabla.com I see the call go to the External(*)
In my external server I have;
Sip.conf
---------
[sip_proxy-out]
type=peer ; we only want to call out, not be called
secret=****
username=nexus*** ; Authentication user for outbound
proxies
fromuser=nexus*** ; Many SIP providers require this!
fromdomain=****.***.com
host=********
usereqphone=yes
and in the extensions.conf I have;
exten =>_6.,1,Dial(SIP/${EXTEN:1}@sip_proxy-out)
This all works!
The problem is it only works if I dial a user that exists on the SER
Server. eg sip:6shad@****.***.com .
It breaks if I call 555555555@voiptalk.org.
When I look at the INVITE packets the URI is being transformed when it
goes from the Internal(*) to the external (*) over IAX2. Rather than
being 555555555@voiptalk.org. it is translated to russia@voiptalk.org !
This explains why calls to users on the SER server work.
I would appreciate an explanation of this phenomena and how to preserver
my URI going form the internal(*) to the external(*).
Warm Regards and Thanks
Shad Mortazavi
---------------
Nexus Group Technical Manager
n|m Nexus Management Inc
-----Original Message-----
From: Shad Mortazavi
Sent: Thursday, March 30, 2006 10:30 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Routing SIP calls via URI
Dear Group;
I can confirm that I have read through the three examples in
www.voip-info.org.
These examples are excellent and address a couple of the questions. I
have IAX2 working between several asterisk servers on our VPN and
between the DMZ and our LAN.
Also
exten => shad,1,Dial(IAX2/bxxxxxx:yyyyyy@192.X.y.x/${EXTEN})
This answers part of the question;
However what I want to do is to send any outbound sip calls via our
external SIP server.
i.e;
VPN LAN IAX2 DMZ Internet
Internal UA <-------> Internal (*) <------> External
(*)<------>
ExternalUA
We have an extensive internal dial plan, X dial the UK, Y dial USA, 1XXX
for Voicemail, 2xxx for Meetme, etc.
Do I need to setup a prefix to dial the internet? And then route all
calls to the External(*) based on this prefix?
Thanks
Shad Mortazavi
------------------------------
Nexus Group Technical Manager
n|m Nexus Management Inc
Dear Group,
I was able to fix this problem;
The solution was to use a prefix to dial out.
The next challenge was to send the SIP Domain over IAX2!. I found that
if I included @SIPDOMAIN it would break the IAX2 communications.
exten => _6.,1,Dial(IAX2/bxxxxxx:yyyyyy@192.X.y.x/${EXTEN}@SIPDOMAIN),
breakes because @SIPDOMAIN is treated as the target context. You also
can not include @Context after the @SIPDOMAIN.
I created a new variable DS which was a concatenation of EXTEN and
SIPDOMAIN separated by % and not @ and I was now able to pass this over
IAX2;
DS = EXTEN%SIPDOMAIN.
exten => _6.,1,Dial(IAX2/bxxxxxx:yyyyyy@192.X.y.x/${DS}).
At the other end I used the CUT command and substring facilities in
Asterisk to split DS by the % eliminator; I re-formed a new variable
which was
DS = EXTEN@SIPDOMAIN.
I can now pass calls from my internal Asterisk server to my external
Asterisk server using IAX2 and then call any external VoIP number.
Warm Regards
Shad Mortazavi
------------------------------
Nexus Group Technical Manager
n|m Nexus Management Inc
-----Original Message-----
From: Shad Mortazavi
Sent: Thursday, March 30, 2006 10:30 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Routing SIP calls via URI
Dear Group;
I can confirm that I have read through the three examples in
www.voip-info.org.
These examples are excellent and address a couple of the questions. I
have IAX2 working between several asterisk servers on our VPN and
between the DMZ and our LAN.
Also
exten => shad,1,Dial(IAX2/bxxxxxx:yyyyyy@192.X.y.x/${EXTEN})
This answers part of the question;
However what I want to do is to send any outbound sip calls via our
external SIP server.
i.e;
VPN LAN IAX2 DMZ Internet
Internal UA <-------> Internal (*) <------> External
(*)<------>
ExternalUA
We have an extensive internal dial plan, X dial the UK, Y dial USA, 1XXX
for Voicemail, 2xxx for Meetme, etc.
Do I need to setup a prefix to dial the internet? And then route all
calls to the External(*) based on this prefix?
Thanks
Shad Mortazavi
------------------------------
Nexus Group Technical Manager
n|m Nexus Management Inc
But is there a way of doing this without a prefix?
because people should dial without prefixes: "name@domain.pt" , not
like:
"6name@domain.pt"
How can we make this without a prefix? something like:
if( !uri=~"@mydomain.pt" ){
forward the all to the Internet
}
:)
Thanks
Joao Pereira
Shad Mortazavi wrote:
>Dear Group,
>
>I was able to fix this problem;
>
>The solution was to use a prefix to dial out.
>
>The next challenge was to send the SIP Domain over IAX2!. I found that
>if I included @SIPDOMAIN it would break the IAX2 communications.
>
>exten => _6.,1,Dial(IAX2/bxxxxxx:yyyyyy@192.X.y.x/${EXTEN}@SIPDOMAIN),
>breakes because @SIPDOMAIN is treated as the target context. You also
>can not include @Context after the @SIPDOMAIN.
>
>I created a new variable DS which was a concatenation of EXTEN and
>SIPDOMAIN separated by % and not @ and I was now able to pass this over
>IAX2;
>
>DS = EXTEN%SIPDOMAIN.
>
>exten => _6.,1,Dial(IAX2/bxxxxxx:yyyyyy@192.X.y.x/${DS}).
>
>At the other end I used the CUT command and substring facilities in
>Asterisk to split DS by the % eliminator; I re-formed a new variable
>which was
>
>DS = EXTEN@SIPDOMAIN.
>
>I can now pass calls from my internal Asterisk server to my external
>Asterisk server using IAX2 and then call any external VoIP number.
>
>Warm Regards
>
>Shad Mortazavi
>------------------------------
>Nexus Group Technical Manager
>n|m Nexus Management Inc
>
>-----Original Message-----
>From: Shad Mortazavi
>Sent: Thursday, March 30, 2006 10:30 AM
>To: asterisk-users@lists.digium.com
>Subject: Re: [Asterisk-Users] Routing SIP calls via URI
>
>Dear Group;
>
>I can confirm that I have read through the three examples in
>www.voip-info.org.
>
>These examples are excellent and address a couple of the questions. I
>have IAX2 working between several asterisk servers on our VPN and
>between the DMZ and our LAN.
>
>Also
>
>exten => shad,1,Dial(IAX2/bxxxxxx:yyyyyy@192.X.y.x/${EXTEN})
>
>This answers part of the question;
>
>However what I want to do is to send any outbound sip calls via our
>external SIP server.
>
>i.e;
> VPN LAN IAX2 DMZ Internet
>Internal UA <-------> Internal (*) <------> External
(*)<------>
>ExternalUA
>
>We have an extensive internal dial plan, X dial the UK, Y dial USA, 1XXX
>for Voicemail, 2xxx for Meetme, etc.
>
>Do I need to setup a prefix to dial the internet? And then route all
>calls to the External(*) based on this prefix?
>
>Thanks
>
>Shad Mortazavi
>------------------------------
>Nexus Group Technical Manager
>n|m Nexus Management Inc
>
>
>_______________________________________________
>--Bandwidth and Colocation provided by Easynews.com --
>
>Asterisk-Users mailing list
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