[asterisk-users] FWD and IPCall

Mon Jan 7 19:34:01 CST 2008

I'm an idiot... I dialled wrong on my phone... I changed it, and was
able to use the Echo application. Dialling for a call to my softphone
as we speak!

On 1/7/08, Huw Richards <huw.richards at oprig.com> wrote:
> I think you said that you already had an ipkall account? If so, logon to
> the account and on the resulting screen there are 2 fields that you need
> to change:
>
> The first is the "SIP Phone Number" - if you have already tried to
> forward the IPKALL number to your fwd account, this field will contain
> your fwd number. You can change this to be anything - my convention is
> to use the actual phone number that IPKALL assigned me.
>
> The second is "SIP Proxy" - again, if you have tried to forward the
> IPKALL number to your fwd account, this field will contain
> "fwd.pulver.net" (or something similar). Change this value to the
> hostname you setup at no-ip.org.
>
> -----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 20:03
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] FWD and IPCall
>
> 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--
> > > > >
> > > > > asterisk-users mailing list
> > > > > To UNSUBSCRIBE or update options visit:
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> > > > >
> > > > > _______________________________________________
> > > > > --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
> > >
> > > _______________________________________________
> > > --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
>
> _______________________________________________
> --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
>
> _______________________________________________
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>
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>

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