[asterisk-users] FWD and IPCall

Huw Richards huw.richards at oprig.com
Mon Jan 7 19:01:02 CST 2008 

http://www.no-ip.com/services/managed_dns/free_dynamic_dns.html 

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Shane D
Sent: Monday, January 07, 2008 19:50
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FWD and IPCall

no-ip.org appears to want to charge me money... Is there a free
alternative?

On 1/7/08, Huw Richards <huw.richards at oprig.com> wrote:
>
> If you want to forward your ipkall number directly to your asterisk
> server:
>
> 1. If your asterisk server is on a private LAN and is connected to the
> internet via a router, enable the router to port forward UDP/5060 &
> UDP/10000-20000 to your asterisk server (assuming you have not changed
> rtp config parameters in rtp.conf).
>
> 2. Check that the firewall (if any) on your asterisk server allows
> connections on UDP/5060 & UDP/10000-20000
>
> 3a. Static public IP address - use the fully qualified domain name
> assigned to the IP address (or setup an account on www.no-ip.org with
a
> name of your choice)
>
> 3b. Dynamic public IP address - setup an account on www.no-ip.org with
a
> name of your choice - install the dynamic ip address update client to
> monitor any change of your ip address (downloads & instructions on
> no-ip.org website)
>
> 4. Goto www.ipkall.com and login to your account. Use your ipkall
number
> as the SIP Phone Number and then the name you selected in 3a or 3b as
> the SIP Proxy.
>
> 5. Wait 60 minutes for changes to take affect (!)
>
> 6. Edit asterisk sip configuration to allow calls from ipkall:
>
> vi /etc/asterisk/sip.conf and find the section beginning [general]
>
> Add/replace the following:
>
> externhost=the name you setup in 3a. or 3b.
> localnet=your private LAN e.g. 192.168.2.0/255.255.255.0
>
> Add a new section at the bottom of the file:
>
> [ipkall.com]
> host=voiper.ipkall.com
> context=from-ipkall
> dtmfmode=rfc2833
> insecure=invite
> type=friend
> canreinvite=no
> disallow=all
> allow=ulaw ; you can add other codecs if you want once the setup works
>
> Save the file. The section you added tells asterisk to accept calls
from
> voiper.ipkall.com and to place them in the "from-ipkall" context. This
> context can be whatever you want. You may need to change the insecure=
> line if you are using asterisk 1.2
>
> 7. Edit asterisk dialplan configuration to handle calls from ipkall:
>
> vi /etc/asterisk/extensions.conf and add at the bottom:
>
> [from-ipkall]
> exten => <IPKALL-NUMBER>,1,NoOp(from-ipkall)
> exten => <IPKALL-NUMBER>,2,NoOp(${CALLERIDNAME}/${CALLERIDNUM})
> exten => <IPKALL-NUMBER>,3,Dial(Local/200 at internal)
>
> Save the file. The section you added tells asterisk what to do with
> calls that are received in the "from-ipkall" context. Replace the
> <IPKALL-NUMBER> with whatever you entered in the SIP Phone number
field
> on the ipkall website (I recommended your ipkall number).
>
> In the "from-ipkall" section:
> 1: display "from-ipkall" on the console
> 2: display the caller id & name
> 3. phone the local extension 200 in context "local" - replace this
line
> with your personal requirements.
>
> Connect to the asterisk console (asterisk -R on my server) and "sip
> reload" followed by "dialplan reload" (asterisk 1.4) or "extensions
> reload" (asterisk 1.2). "sip reload" will re-read the sip.conf file &
> "dialplan reload"/"extensions reload" will re-read the extensions.conf
> file.
>
> Phone your ipkall number and see if anything is displayed on the
console
> and/or your phone rings.
>
> If nothing on the console when you phone, try "sip set debug peer
> ipkall.com" (asterisk 1.4 - not sure of the command for asterisk 1.2)
> and phone again.
>
> Post back your results.
>
>
>
>
>
>
>
>
>
>
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Shane D
> Sent: Monday, January 07, 2008 17:32
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] FWD and IPCall
>
> Okay... That was kind of confusing. Would you contact me off-list to
> help me specifically?
>
> I've double-checked everything for the IAX, and it's a no-go. Maybe
> I'll try this SIP thing. But then again, if I can just hook IPKall to
> the server directly, I don't need FWD...
>
> On 1/7/08, Huw Richards <huw.richards at oprig.com> wrote:
> > My config is as follows
> >
> > Excerpt of sip.conf:
> >
> > [general]
> > externhost=fully.qualified.domain.name
> > localnet=192.168.2.0/255.255.255.0
> > srvlookup=no
> > defaultexpiry=3600
> > dtmfmode=rfc2833
> >
> > register => <fwd-id>:<fwd-pwd>@fwd.pulver.com/<fwd-id>
> >
> > [sipfwd]
> > type=peer
> > secret=<fwd-pwd>
> > username=<fwd-id>
> > fromdomain=fwd.pulver.com
> > host=fwd.pulver.com
> > disallow=all
> > allow=ulaw
> > canreinvite=yes
> > insecure=invite
> > qualify=yes
> > context=from-fwd
> >
> > Excerpt of extensions.conf:
> >
> > [from-fwd]
> > exten => <fwd-id>,1,NoOp(from-fwd)
> > exten => <fwd-id>,n,Dial(whatever)
> >
> > I have a dynamic public IP address, so I use http://www.no-ip.org to
> map
> > my IP address to name. My router port forwards UDP/5060 &
> > UDP/10000-20000 to the internal asterisk server.
> >
> > However, I do not have ipkall forwarding to my fwd account. I have
it
> > forwarding directly to my asterisk server using the no-ip.org
address
> I
> > registered.
> >
> > e.g. forward to sip:xxx at fully.qualified.domain.name on ipkall
website
> > and then in sip.conf:
> >
> > [ipkall.com]
> > host=voiper.ipkall.com
> > context=from-ipkall
> > dtmfmode=rfc2833
> > insecure=invite
> > type=friend
> > canreinvite=no
> > disallow=all
> > allow=ulaw
> >
> > And in extensions.conf:
> >
> > [from-ipkall]
> > exten => xxx,1,NoOp(from-ipkall)
> > exten => xxx,n,Dial(whatever)
> >
> >
> >
> >
> > -----Original Message-----
> > From: asterisk-users-bounces at lists.digium.com
> > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Shane
D
> > Sent: Monday, January 07, 2008 12:09
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [asterisk-users] FWD and IPCall
> >
> > It's Iax2. Is there a way of using amore reliable sip
> > connectoin/something slightly different?
> >
> > If so, how would I go about that.
> >
> > On 1/7/08, Huw Richards <huw.richards at oprig.com> wrote:
> > > You haven't said if your connection to fwd is SIP or IAX2 but I
have
> > > found IAX2 connections to fwd to be unreliable. Other people may
> have
> > > different results.
> > >
> > > -----Original Message-----
> > > From: asterisk-users-bounces at lists.digium.com
> > > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
Shane
> D
> > > Sent: Monday, January 07, 2008 10:17
> > > To: asterisk-users at lists.digium.com
> > > Subject: [asterisk-users] FWD and IPCall
> > >
> > > Hello All,
> > >
> > > I have a problem. I have tried everything that is in the book "The
> > > Future of Telephony" as well as on the FWD (freeworlddialup)
> website,
> > > and there is still a problem. My asterisk box is not able to
> associate
> > > with the FWD server. I get:
> > > Registration Rejected by [insert IP], and I can't use my IPCall
> number
> > > to reach my Asterisk box.
> > > Any suggestions?
> > > --
> > > -Shane
> > > Blog: http://blind-geek.com/blog/
> > > CoOwner: http://sjtechzone.com
> > > AIM: inhaddict
> > > Skype: chatter8712
> > >
> > > _______________________________________________
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http://www.api-digital.com--
> > >
> > > asterisk-users mailing list
> > > To UNSUBSCRIBE or update options visit:
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> > >
> > > _______________________________________________
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http://www.api-digital.com--
> > >
> > > asterisk-users mailing list
> > > To UNSUBSCRIBE or update options visit:
> > >    http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> >
> >
> > --
> > -Shane
> > Blog: http://blind-geek.com/blog/
> > CoOwner: http://sjtechzone.com
> > AIM: inhaddict
> > Skype: chatter8712
> >
> > _______________________________________________
> > --Bandwidth and Colocation Provided by http://www.api-digital.com--
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >    http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> > _______________________________________________
> > --Bandwidth and Colocation Provided by http://www.api-digital.com--
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >    http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
> --
> -Shane
> Blog: http://blind-geek.com/blog/
> CoOwner: http://sjtechzone.com
> AIM: inhaddict
> Skype: chatter8712
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
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>
> _______________________________________________
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>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
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>


-- 
-Shane
Blog: http://blind-geek.com/blog/
CoOwner: http://sjtechzone.com
AIM: inhaddict
Skype: chatter8712

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