[asterisk-users] FWD and IPCall

Shane D chatter8712 at gmail.com
Mon Jan 7 19:02:33 CST 2008 

Okay. What do you mean in step 4/5 (I don't remember which) where you
write something about "Use your IPKall number as the sip number" I am
signing up for IPKall... Right?

On 1/7/08, Shane D <chatter8712 at gmail.com> wrote:
> 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
> > > >
> > > > _______________________________________________
> > > > --Bandwidth and Colocation Provided by http://www.api-digital.com--
> > > >
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> > > >
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> > > >
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> > > > 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
> > >
> > > _______________________________________________
> > > --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:
> >    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
>


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

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