[asterisk-users] Softphones IAX vs. SIP, remote connectivity.

Nick Ellson grimm at nickellson.com
Thu Sep 7 11:40:14 MST 2006



Hello Michael,

I just had both Mom and my brother up as extensions on my Asterisk pbx 
using IAX2, the Cubix phone for now, but I downloaded and tried several. I 
loke multiple lines, but a clean GUI is better for my family..

Oh yeah, it worked flawlessly :)

I open one port to my server udp/4569 and that was it. I shut the rest 
off.

For remote family, IAX2 will be what I use right now.

Anybody see a Video capable version for Windows? The MAC has one, darn it.



Nick


-- 
Nick Ellson
CCDA, CCNP, CCSP, CCAI,
MCSE 2000, Security+, Network+
Network Hobbyist, VFR Private Pilot.


On Thu, 7 Sep 2006, Ferguson, Michael wrote:

> Hi "Guys"
>
> I too am trying to do exactly the same thing in being a provider for family members. My Asterisk server is on a public ip, my home is behind a Watchguard Firebox, my job is also behind a Firebox. I am using a combination of Cisco 7960, Linksys 941 and XTEN Softphone. Sometimes it works and sometimes it does not.
>
> You idea on using a IAX2 softphone appears to be what will solve my problem.
>
> Thanks very much.... Post more ideas. 'preciate it.
>
>
>
>
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Nick Ellson
> Sent: Thursday, September 07, 2006 9:07 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Softphones IAX vs. SIP, remote connectivity.
>
>
> Bruce,
>
> I *just* tested the XtremePhone, IAX2 softphone. Other than trying to figure out how to get it to send proper CallerID to the other phones, it worked right off, in both directions. Excellent!
>
> Perhaps working the IAX2 angle will be less of a hassle, I will go looking for one that does video now.
>
> Maybe it's time to buy an IAX2-ATA adaptor and see how well that works over the net.
>
> Nick
>
> As for the SIP logs, I start Asterisk with -vvvvc already, I did a sip debug and tried my call from the house to my remote SIP phone. YIKES!!
> Gunna take a bit to understand all that, but I think I did see an INVITE, and a CANCEL twice in a row and I did not hit the hang-up switch. So that might explain why no connection is made, and the called gets my voice-mail (according to my wife)
>
>
>
> --
> Nick Ellson
> CCDA, CCNP, CCSP, CCAI,
> MCSE 2000, Security+, Network+
> Network Hobbyist, VFR Private Pilot.
>
>
> On Thu, 7 Sep 2006, Bruce Reeves wrote:
>
>> Nick,
>>
>> I have done what you are talking about as far as being a provider for family
>> members. I used an IAX softphone mainly to eliminate the need for so many
>> holes in the firewall. And secondly because the idefisk IAX softphone
>> allowed me to extract the zip version, configure the phone, and zip the
>> folder up and email it to my family members. So for my mom it was simply
>> unzip the folder and
>>
>> On 9/7/06, Nick Ellson <grimm at nickellson.com> wrote:
>>>
>>>
>>>  Bob,
>>>
>>>  I will up the logs today, have my phone at work with me. (though the Wife
>>>  and Kids are not up yet ;)
>>>
>>>  Anything specific I should target?
>>>
>>>
>>>  Nick
>>>
>>>
>>>  --
>>>  Nick Ellson
>>>  CCDA, CCNP, CCSP, CCAI,
>>>  MCSE 2000, Security+, Network+
>>>  Network Hobbyist, VFR Private Pilot.
>>>
>>>
>>>  On Thu, 7 Sep 2006, Bob Chiodini wrote:
>>>
>>>>  Nick,
>>>>
>>>>  Anything helpful in the asterisk or system logs.
>>>>
>>>>  Try bumping up the debug and verbose levels see what shows up on the
>>>>  console.
>>>>
>>>>  Weird that it would work inbound and not outbound.
>>>>
>>>>  Bob...
>>>>
>>>>
>>>>  On Thu, 2006-09-07 at 04:48 -0700, Nick Ellson wrote:
>>>>>
>>>>>  Hey all,
>>>>>
>>>>>  A previous annoyance with not being able to call out to my brother on
>>>  FWD
>>>>>  from my Asterisk system had me thinking that since I have my own PBX,
>>>  and
>>>>>  that system has it's own 1-to-1 static NAT to the internet, I should
>>>>>  be
>>>
>>>>>  able to act as the provider for him or any of my family, and have them
>>>  as
>>>>>  local extensions of my PBX, right?
>>>>>
>>>>>  So I took my laptop to work (using the X-Lite SIP softphone) and watch
>>>  my
>>>>>  ACL logs on my router for any denies to my Asterisk box. As expected
>>>>>  udp/5060, then once that was open, a series of randomish udp/10000+
>>>>>  requests. My phone registered, and I tried to call one of the phones
>>>>>  behind a PAP2. Worked first shot, and just as clear and responsive as
>>>  it
>>>>>  was when I was home. But, the phones at home could not call me, they
>>>  when
>>>>>  to voice mail.
>>>>>
>>>>>  I had heard that SIP doesn't survive NAT all that well, and that IAX
>>>>>  native phones do a better job. My question is, given my description of
>>>  how
>>>>>  I am set up and what I am trying to accomplish, should I be looking at
>>>  SIP
>>>>>  or is IAX a more robust choice? (I was hoping to get video working as
>>>>>  well, h.263 I believe it is).
>>>>>
>>>>>  Nick
>>>>>
>>>>>
>>>> _______________________________________________
>>>>  --Bandwidth and Colocation provided by Easynews.com --
>>>>
>>>>  asterisk-users mailing list
>>>>  To UNSUBSCRIBE or update options visit:
>>>>    http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>
>>> _______________________________________________
>>>  --Bandwidth and Colocation provided by Easynews.com --
>>>
>>>  asterisk-users mailing list
>>>  To UNSUBSCRIBE or update options visit:
>>>     http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>>
>>
>> --
>> Bruce
>> Nortex Networks
>>
>>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>


More information about the asterisk-users mailing list