VoIP Mailing List Archives

[chatter8712 at gmail.com]

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

[huw.richards at oprig.com]

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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[chatter8712 at gmail.com]

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:

[b.benchev at gmail.com]

[chatter8712 at gmail.com]

I will try that when I return home in about two hours. I'll let you know.

On 1/7/08, Benchev <b.benchev at gmail.com> wrote:

[chatter8712 at gmail.com]

I get the following output:

Jan 7 10:56:22 NOTICE[4666]: chan_iax2.c:7536 socket_read:
Registration of '886036' rejected: 'Registration Refused' from:
'192.246.69.186'

It shouldn't be trying to do something on my network (192) should it?

On 1/7/08, Shane D <chatter8712 at gmail.com> wrote:

[chatter8712 at gmail.com]

Ignore the part about 192... That happens to be the IP of iax2.fwdnet.net.

If anyone has set up FWD successfully for incoming only, let me know.

On 1/7/08, Shane D <chatter8712 at gmail.com> wrote:

[asteriskfan at wcta.net]

192.246.x.x =! 192.168.x.x if that is what you're thinking.

Anyway, "registration Refused" sounds like you're getting up to FWD and
attempting to authenticate, but failing at that point.

double-check your iax.conf settings against the FWD "Extra Features"
settings... of course make sure IAX is selected too with FWD.

-Troy
Shane D wrote:

[huw.richards at oprig.com]

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:

[Info at IPKall.com]

Try using the IP address, and not the dynamic URL, does anything change?

IPKall
IPKall Forum
http://voxilla.com/PNphpBB2-viewforum-f-38.html


-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Huw Richards
Sent: Monday, January 07, 2008 1:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FWD and IPCall

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:

[chatter8712 at gmail.com]

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:

[chatter8712 at gmail.com]

Still rejected.

On 1/7/08, www.IPKall.com <Info at ipkall.com> wrote:

[huw.richards at oprig.com]

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:

[chatter8712 at gmail.com]

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:

[huw.richards at oprig.com]

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: