What is suggested then on routing calls through my wan port?  Is there any
cheap kind of equipment I can buy.  I am trying to setup a network with
phones -> sipx -> router to wan -> sip trunking provider service as I have
software that I want to launch a product with.

I have been also trying to test calls from outside the router to the
internal phones as I wanted to make sure media could work through the
router.  I was hoping I could put sipx in the DMZ or something.  What do I
need to connect sipx to the wan(cheaply) and still be able to hang my
computers off the router.

does asterisk work better?  I heard alot of people have done that behind a
router, or what cheap router can I use.

thx,
dean

2008/8/8 Tony Graziano <[EMAIL PROTECTED]>

> SIP is not exactly simple. You need to understand why it uses DNS, the
> options it uses for DHCP and so forth.
>
> It is "rolled" into a nice package to deliver all the basic needs into a
> single CD because a lot of admins will run it on a separate IP
> network/vlan or in a lab environment to learn the technology and
> protocol before they decide to run a department or business on it.
>
> Get a hub, a laptop with x-lite or something on it, and set it up all by
> itself.
>
> TWO DHCP server's is a bad thing. A linksys router can't push out the
> options that SIP needs. If the SIPX server has the linksys as its
> default router, the router can be a router, and sipx could do DNS, DHCP
> & NTP. But you ought to learn how to administer them in Linux first
> before trying to make it a production system.
>
> Trying to route calls or users through the WAN port of a Linksys router
> for SIP is a bad idea.
>
> Get your feet wet like Mike suggests. If you are a Linux newbie, try
> getting webmin (http://www.webmin.com) on it to have insight to DHCP,
> DNS & NTP.
>
> Tony
> >>> "Dean Hiller" <[EMAIL PROTECTED]> 08/07/08 20:26 PM >>>
> I am not sure I know how to do all that with dns, dhcp as that seemed
> like
> the harder route.  isn't that bad to  have two dhcp machines on your
> network
> as my linksys is doing dhcp?  Iam pretty sure I can turn dhcp off on the
> linksys but if I do that, and sipx is assigning ip addresses, will my
> network still work properly.  Doesn't the linksys have to have an subnet
> ip
> of 192.168.1.1, how will it get this address if sipx is the dhcp.
>
> It would ROCK if the sipx setup tutorial actually told you to
> 1. step 1, turn off router's dhcp addressing if you have linksys etc.
> and
> what the configuration is(as in any router, I think we can find the
> settings).
>
> I guess I will try turning off dhcp and see if it just works and try it
> out.
> thx,
> dean
>
> On Fri, Aug 8, 2008 at 5:43 AM, Picher, Michael
> <[EMAIL PROTECTED]>wrote:
>
> >  Back up the train…
> >
> >
> >
> > Why don't you lab this thing on some private addressing on a regular
> > network…  you have my head spinning here with yours, and your friends
> and
> > you both have things in dmz's.
> >
> >
> >
> > Walk before you run.  Stay away from nats, dns mess and firewalls
> until you
> > know you have a working solution.  Follow the wiki, use dns/dhcp off
> of the
> > pbx and make a system work first…  then you can throw twists at it
> once you
> > understand it a little better…
> >
> >
> >
> > Mike
> >
> >
> >
> > *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On
> Behalf Of
> > *Dean Hiller
> > *Sent:* Wednesday, August 06, 2008 11:26 PM
> > *To:* Picher, Michael
> > *Cc:* sipx-users@list.sipfoundry.org
> > *Subject:* Re: [sipx-users] ACD just rings and rings (friend's issue)
> >
> >
> >
> > My friend is having a different issue.  His sipx is behind a
> router/NAT as
> > well, but he actually has DNS sipx.xxxxx.com pointing to that router
> and
> > then has sipx in the DMZ(I am doing something similar).
> >
> > For some reason, when sipx receives an invite for [EMAIL PROTECTED],
> it
> > forwards it to [EMAIL PROTECTED]  why is this?  What is going on here?
> > sipx should accept the invite and send invite to
> [EMAIL PROTECTED],
> > shouldn't it?
> >
> > thanks,
> > dean
> >
> > On Thu, Aug 7, 2008 at 9:45 AM, Dean Hiller <[EMAIL PROTECTED]>
> wrote:
> >
> > Here are my screenshots(sorry for the delay...got hit with lots of
> > issues).  I am going to run a bunch of tests again today and see if I
> can
> > get farther.
> > thx,
> > dean
> >
> >
> >
> > On Mon, Aug 4, 2008 at 9:17 PM, Picher, Michael
> <[EMAIL PROTECTED]>
> > wrote:
> >
> > No, you only need to activate if you make changes to the ACD service.
> >
> >
> >
> > One thing that is typically recommended if you have multi-line phones
> is to
> > create another extension that is used for the ACD only and make it a
> second
> > line on the phone.  This way when a user calls out on their phone they
> do
> > not cause busy issues with the ACD calls…
> >
> >
> >
> > Not suggesting that this will solve your problem, it's just a typical
> > recommendation.
> >
> >
> >
> > What are your queue settings?  (shoot me a screen shot if you'd like).
> >
> >
> >
> > Mike
> >
> >
> >
> > *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On
> Behalf Of
> > *Dean Hiller
> > *Sent:* Monday, August 04, 2008 9:10 AM
> > *To:* Picher, Michael
> > *Cc:* sipx-users@list.sipfoundry.org
> > *Subject:* Re: [sipx-users] ACD just rings and rings
> >
> >
> >
> > I reactivated a few times.  do I also have to reactivate when agents
> sign
> > in and out?  I guess I will try again tomorrw the reactivating.  I
> keep
> > getting no agents, but I have signed them in, so I am not sure what is
> > wrong.  any logs somewhere to look at?
> > thx,
> > dean
> >
> > On Mon, Aug 4, 2008 at 6:42 PM, Picher, Michael
> <[EMAIL PROTECTED]>
> > wrote:
> >
> > Any time you make an ACD change you have to re-activate the ACD.
> >
> >
> >
> > *From:* [EMAIL PROTECTED] [mailto:
> > [EMAIL PROTECTED] *On Behalf Of *Dean Hiller
> > *Sent:* Monday, August 04, 2008 1:43 AM
> > *To:* sipx-users@list.sipfoundry.org
> > *Subject:* [sipx-users] ACD just rings and rings
> >
> >
> >
> > I setup and ACD and added 2000 and 3000 to the lines(setting rocket as
> > queue which I added already).  I added 8 agents into this rocket
> queue, but
> > when I call 2000 or 3000, it just rings and rings and rings.
> >
> > Also, my extension range is 1000 to 2999.  My agents are all 2001 to
> > 2008.    I can call from 1000 extension to 2001 just fine, but when I
> call
> > 3000 or 2000, I expect 2001 to ring since it is in the queue. * I see
> no
> > calls coming in on the ACD page though like 2000 and 3000 are not
> lines of
> > the ACD.  *what am I doing wrong with the ACD here?
> > thanks,
> >
> > --
> > Dean Hiller
> > CEO/Founder of Extreme Software Offshoring
> > http://xsoftware.biz
> > http://www.linkedin.com/in/deanhiller
> > Beijing Cell: 136-991-41547
> > US Phone: 303-376-5776
> >
> >
> >
> >
> > --
> > Dean Hiller
> > CEO/Founder of Extreme Software Offshoring
> > http://xsoftware.biz
> > http://www.linkedin.com/in/deanhiller
> > Beijing Cell: 136-991-41547
> > US Phone: 303-376-5776
> >
> >
> >
> >
> > --
> > Dean Hiller
> > CEO/Founder of Extreme Software Offshoring
> > http://xsoftware.biz
> > http://www.linkedin.com/in/deanhiller
> > Beijing Cell: 136-991-41547
> > US Phone: 303-376-5776
> >
> >
> >
> >
> > --
> > Dean Hiller
> > CEO/Founder of Extreme Software Offshoring
> > http://xsoftware.biz
> > http://www.linkedin.com/in/deanhiller
> > Beijing Cell: 136-991-41547
> > US Phone: 303-376-5776
> >
>
>
>
> --
> Dean Hiller
> CEO/Founder of Extreme Software Offshoring
> http://xsoftware.biz
> http://www.linkedin.com/in/deanhiller
> Beijing Cell: 136-991-41547
> US Phone: 303-376-5776
>
>


-- 
Dean Hiller
CEO/Founder of Extreme Software Offshoring
http://xsoftware.biz
http://www.linkedin.com/in/deanhiller
Beijing Cell: 136-991-41547
US Phone: 303-376-5776
_______________________________________________
sipx-users mailing list
sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoundry.org/archive/sipx-users
Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users

Reply via email to