[asterisk-users] FWD and IPCall

[Huw Richards]

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

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