[Asterisk-Users] Asterisk(OS X) & X-Lite

[Kenton Powell]

I'm new to Asterisk and have installed it on OS X. I am trying to use
an X-Lite softphone to connect through Asterisk to FreeWorld Dialup and
am not having any luck.

I have looked at several settings pages at voip-info.com but have had
no luck. I am on a school network which I assume has some security but
the particulars are unknown as they are not very helpful. It is
necessary to authenticate each machine through custom perl scripts
which I believe log MAC addresses; don't know if that poses a problem.
Additionally, I am using a wireless switch that passes my connection to
a school server which assigns IP addresses.

I ran both Assistants and have a FreeWorld Dialup number and have
routed that number to an extension in Asterisk. I cannot connect using
X-Lite using the following instructions. I keep getting login timeout
messages.

X-Lite configuration:
Menu | System Settings | SIP Proxy | default
Display Name: Full Name
User Name & Authorization User: <username>
Password: <secret>
Domain/Realm: x.x.x.x
SIP Proxy: x.x.x.x

x.x.x.x = IP address of Asterisk

Are the username and authorization user the same name or number
provided from FreeWorld Dialup or is one from the Asterisk setup? I
assume that the IP address of Asterisk is the IP of the Mac with
Asterisk installed.

Has anyone successfully configured Asterisk on OS X with FreeWorld
Dialup? Do I need to use an outbound proxy? If so, how do I configure
that?

Any help or suggestions are greatly appreciated. Thank in advance.

Kenton

Log:

© 2004 Xten Networks, Inc. All rights reserved.
X-Lite release 1103m build stamp 14266
License key: A91031380B8611D99AAB000393BE4F08

Established SIP protocol listen on: 172.30.247.226:5060

Discovered Symmetric NAT Firewall

SIP: 172.30.247.226:5060
RTP: 172.30.247.226:8000
NAT: 63.211.54.166

PROXY#0: 172.30.247.226:5060


SEND TIME: 36063869
SEND >> 172.30.247.226:5060
REGISTER sip:172.30.247.226 SIP/2.0
Via: SIP/2.0/UDP
63.211.54.166:5060;rport;branch=z9hG4bKCF9C0A670C2D11D9801D000393BE4F08
From: Kenton Powell <sip:***@172.30.247.226>;tag=233694996
To: Kenton Powell <sip:***@172.30.247.226>
Contact: "Kenton Powell" <sip:***@63.211.54.166:5060>
Call-ID: ***@172.30.247.226
CSeq: 8393 REGISTER
Expires: 1800
Max-Forwards: 70
User-Agent: X-Lite release 1103m
Content-Length: 0


RECEIVE TIME: 36063870
RECEIVE << 172.30.247.226:5060
REGISTER sip:172.30.247.226 SIP/2.0
Via: SIP/2.0/UDP
63.211.54.166:5060;rport;branch=z9hG4bKCF9C0A670C2D11D9801D000393BE4F08
From: Kenton Powell <sip:***@172.30.247.226>;tag=233694996
To: Kenton Powell <sip:***@172.30.247.226>
Contact: "Kenton Powell" <sip:***@63.211.54.166:5060>
Call-ID: ***@172.30.247.226
CSeq: 8393 REGISTER
Expires: 1800
Max-Forwards: 70
User-Agent: X-Lite release 1103m
Content-Length: 0


SEND TIME: 36065380
SEND >> 172.30.247.226:5060
REGISTER sip:172.30.247.226 SIP/2.0
Via: SIP/2.0/UDP
63.211.54.166:5060;rport;branch=z9hG4bKCF9C0A670C2D11D9801D000393BE4F08
From: Kenton Powell <sip:***@172.30.247.226>;tag=233694996
To: Kenton Powell <sip:***@172.30.247.226>
Contact: "Kenton Powell" <sip:***@63.211.54.166:5060>
Call-ID: ***@172.30.247.226
CSeq: 8393 REGISTER
Expires: 1800
Max-Forwards: 70
User-Agent: X-Lite release 1103m
Content-Length: 0


RECEIVE TIME: 36065382
RECEIVE << 172.30.247.226:5060
REGISTER sip:172.30.247.226 SIP/2.0
Via: SIP/2.0/UDP
63.211.54.166:5060;rport;branch=z9hG4bKCF9C0A670C2D11D9801D000393BE4F08
From: Kenton Powell <sip:***@172.30.247.226>;tag=233694996
To: Kenton Powell <sip:***@172.30.247.226>
Contact: "Kenton Powell" <sip:***@63.211.54.166:5060>
Call-ID: ***@172.30.247.226
CSeq: 8393 REGISTER
Expires: 1800
Max-Forwards: 70
User-Agent: X-Lite release 1103m
Content-Length: 0


SEND TIME: 36068384
SEND >> 172.30.247.226:5060
REGISTER sip:172.30.247.226 SIP/2.0
Via: SIP/2.0/UDP
63.211.54.166:5060;rport;branch=z9hG4bKCF9C0A670C2D11D9801D000393BE4F08
From: Kenton Powell <sip:***@172.30.247.226>;tag=233694996
To: Kenton Powell <sip:***@172.30.247.226>
Contact: "Kenton Powell" <sip:***@63.211.54.166:5060>
Call-ID: ***@172.30.247.226
CSeq: 8393 REGISTER
Expires: 1800
Max-Forwards: 70
User-Agent: X-Lite release 1103m
Content-Length: 0


RECEIVE TIME: 36068386
RECEIVE << 172.30.247.226:5060
REGISTER sip:172.30.247.226 SIP/2.0
Via: SIP/2.0/UDP
63.211.54.166:5060;rport;branch=z9hG4bKCF9C0A670C2D11D9801D000393BE4F08
From: Kenton Powell <sip:***@172.30.247.226>;tag=233694996
To: Kenton Powell <sip:***@172.30.247.226>
Contact: "Kenton Powell" <sip:***@63.211.54.166:5060>
Call-ID: ***@172.30.247.226
CSeq: 8393 REGISTER
Expires: 1800
Max-Forwards: 70
User-Agent: X-Lite release 1103m
Content-Length: 0


RECEIVE TIME: 36068883
RECEIVE << 172.30.247.226:5060


SEND TIME: 36074392
SEND >> 172.30.247.226:5060
REGISTER sip:172.30.247.226 SIP/2.0
Via: SIP/2.0/UDP
63.211.54.166:5060;rport;branch=z9hG4bKCF9C0A670C2D11D9801D000393BE4F08
From: Kenton Powell <sip:***@172.30.247.226>;tag=233694996
To: Kenton Powell <sip:***@172.30.247.226>
Contact: "Kenton Powell" <sip:***@63.211.54.166:5060>
Call-ID: ***@172.30.247.226
CSeq: 8393 REGISTER
Expires: 1800
Max-Forwards: 70
User-Agent: X-Lite release 1103m
Content-Length: 0


RECEIVE TIME: 36074424
RECEIVE << 172.30.247.226:5060
REGISTER sip:172.30.247.226 SIP/2.0
Via: SIP/2.0/UDP
63.211.54.166:5060;rport;branch=z9hG4bKCF9C0A670C2D11D9801D000393BE4F08
From: Kenton Powell <sip:***@172.30.247.226>;tag=233694996
To: Kenton Powell <sip:***@172.30.247.226>
Contact: "Kenton Powell" <sip:***@63.211.54.166:5060>
Call-ID: ***@172.30.247.226
CSeq: 8393 REGISTER
Expires: 1800
Max-Forwards: 70
User-Agent: X-Lite release 1103m
Content-Length: 0


RECEIVE TIME: 36078923
RECEIVE << 172.30.247.226:5060

[Joe Woss]

I am new to the Asterisk-users Mailing list, but i assume that you
post your answers to the group?

Kenton,

Post by Kenton Powell
Menu | System Settings | SIP Proxy | default
Display Name: Full Name
User Name & Authorization User: <username>
Password: <secret>
Domain/Realm: x.x.x.x
SIP Proxy: x.x.x.x
x.x.x.x = IP address of Asterisk
Are the username and authorization user the same name or number
provided from FreeWorld Dialup or is one from the Asterisk setup? I
assume that the IP address of Asterisk is the IP of the Mac with
Asterisk installed.

This is from the sip.conf file in /etc/asterisk/
you must specify an entry for the user you are trying to connect as.
Look at the voip-info.org Wiki for help on sip.conf

You can email me personally for more help after you've read the wiki.

-Joe

[Benjamin on Asterisk Mailing Lists]

Post by Joe Woss

Post by Kenton Powell
Menu | System Settings | SIP Proxy | default
Display Name: Full Name
User Name & Authorization User: <username>
Password: <secret>
Domain/Realm: x.x.x.x
SIP Proxy: x.x.x.x

This is from the sip.conf file in /etc/asterisk/

No, this is not sip.conf, it is from X-Lite's settings.

Post by Joe Woss
you must specify an entry for the user you are trying to connect as.
Look at the voip-info.org Wiki for help on sip.conf

On MacOSX you can do this without editing configuration files. And in
fact, he said he was using the Assistants (GUI based configuration
tools for Asterisk) to configure Asterisk.

The FWD Assistant configures Asterisk to use IAX. The Extension
Assistant lets the user choose whether to use SIP or IAX or both.
These Assistants have been used by many Mac users with success. It is
very unlikely that his problem is with the Asterisk configuration.

[Benjamin on Asterisk Mailing Lists]

Post by Kenton Powell
I'm new to Asterisk and have installed it on OS X. I am trying to use
an X-Lite softphone to connect through Asterisk to FreeWorld Dialup and
am not having any luck.
I have looked at several settings pages at voip-info.com but have had
no luck. I am on a school network which I assume has some security but
the particulars are unknown as they are not very helpful. It is
necessary to authenticate each machine through custom perl scripts
which I believe log MAC addresses; don't know if that poses a problem.
Additionally, I am using a wireless switch that passes my connection to
a school server which assigns IP addresses.
I ran both Assistants and have a FreeWorld Dialup number and have
routed that number to an extension in Asterisk. I cannot connect using
X-Lite using the following instructions. I keep getting login timeout
messages.
Menu | System Settings | SIP Proxy | default
Display Name: Full Name
User Name & Authorization User: <username>
Password: <secret>
Domain/Realm: x.x.x.x
SIP Proxy: x.x.x.x
x.x.x.x = IP address of Asterisk
Are the username and authorization user the same name or number
provided from FreeWorld Dialup or is one from the Asterisk setup? I
assume that the IP address of Asterisk is the IP of the Mac with
Asterisk installed.
Has anyone successfully configured Asterisk on OS X with FreeWorld
Dialup? Do I need to use an outbound proxy? If so, how do I configure
that?

If you have used the FWD Assistant, it will have configured your
Asterisk server to use FWD through IAX, not SIP. Since you seem to be
behind NAT and a firewall, using IAX is a good idea anyway. IAX is NAT
friendly, SIP is not.

As far as your X-Lite connecting to Asterisk goes, I presume X-Lite
and Asterisk are running on different Macs, is that correct?

If so, you need to find out if that wireless bridge you're talking
about uses NAT between your X-Lite Mac and your Asterisk Mac. SIP
doesn't work well with NAT and that is probably the reason why you
can't get through to your Asterisk Mac. You could download iaxComm (an
IAX softphone - link on the MacOSX page of the Wiki at
www.voip-info.org) and again use IAX instead of SIP.

In any event, if you want to use Asterisk as a "VoIP Hub" instead of
connecting to FWD directly from your softphone, then the chain would
be like this ...

[softphone]---internal extension--->[asterisk]---FWD number--->[FWD]

Wherever there is NAT between two points in the chain, you are saving
yourself a lot of trouble if you use IAX instead of SIP for that
particular connection, but you don't necessarily have to use the same
protocol all the way. Asterisk can translate between SIP and IAX for
you.

rgds
benjk

[Kenton Powell]

Post by Benjamin on Asterisk Mailing Lists
If you have used the FWD Assistant, it will have configured your
Asterisk server to use FWD through IAX, not SIP. Since you seem to be
behind NAT and a firewall, using IAX is a good idea anyway. IAX is NAT
friendly, SIP is not.

I have decided to use the IAX protocol enabled by the FWD Assistant
instead of SIP because of the compatibility issues.

Post by Benjamin on Asterisk Mailing Lists
As far as your X-Lite connecting to Asterisk goes, I presume X-Lite
and Asterisk are running on different Macs, is that correct?

That actually is incorrect. I am trying to setup the Asterisk server
and softphone on the same machine with the hope of extending extensions
to other computers or phones after I test the configuration.

Post by Benjamin on Asterisk Mailing Lists
If so, you need to find out if that wireless bridge you're talking
about uses NAT between your X-Lite Mac and your Asterisk Mac. SIP
doesn't work well with NAT and that is probably the reason why you
can't get through to your Asterisk Mac. You could download iaxComm (an
IAX softphone - link on the MacOSX page of the Wiki at
www.voip-info.org) and again use IAX instead of SIP.

The switch I have is not using NAT since it is not functioning as a
DHCP server. In the future that may become an issue if the school's
switches are using DHCP but I will cross that bridge when I get there.
As a proactive measure I have decided to use the IAX protocol iaxComm
softphone to hopefully avoid NAT problems.

Post by Benjamin on Asterisk Mailing Lists
In any event, if you want to use Asterisk as a "VoIP Hub" instead of
connecting to FWD directly from your softphone, then the chain would
be like this ...
[softphone]---internal extension--->[asterisk]---FWD number--->[FWD]

That is the plan.

Post by Benjamin on Asterisk Mailing Lists
Wherever there is NAT between two points in the chain, you are saving
yourself a lot of trouble if you use IAX instead of SIP for that
particular connection, but you don't necessarily have to use the same
protocol all the way. Asterisk can translate between SIP and IAX for
you.

That is interesting and good to know.

In any event, although I have made progress in planning the network and
have installed iaxComm but I still can't get it to connect to the
server. I have established accounts with the New Extension Assistant
based on the following

Account Name (This is based on personal preference correct?)
Host is the IP address of the asterisk server. (Can the client and
server run on the same machine?)
Username is your username on the remote asterisk server. (Based on the
users created in the New Extension Assistant, correct?)
iaxComm uses MD5 authentication. If the remote server doesn't like your
password, make sure it's not misconfigured to use plaintext
authentication. (How can I check this, would this be an issue?

Thanks again for your quick and helpful response.

Kenton

[Benjamin on Asterisk Mailing Lists]

Post by Kenton Powell
I am trying to setup the Asterisk server
and softphone on the same machine with the hope of extending extensions
to other computers or phones after I test the configuration.

OK. Now, it is not impossible to run both the client and server on the
same machine, but it is a little tricky and requires special settings.
This is not covered by the Extension Assistant.

You may want to check the Wiki page I have set up explaining how to do this:

http://www.voip-info.org/wiki-Localhost+Gateway

rgds
benjk