[asterisk-users] Question about voip.ms service.

Tue Sep 13 16:09:03 CDT 2011

Yup, that part I got. What I am not clear about is how to set up the DID to
go to my URI. When I select "manage DIDs" and click on the one I want to
change, I see the following options for routing the DID

x SIP/IAX - [main account] IAX2/100000 <- with my account number
x SIP URI - SIP:mysipid at myuri.com:5060
x System - Hangup

There are several other options but they are not selectable for me because I
have not set up to use them.

I used to have the routing set to SIP URI where I was able to specify my URI
where the call was routed to. But with the SIP/IAX option I do not have that
ability.

I am missing something fundamental here. My asterisk has the iax.conf and
extensions.conf entries ready to receive calls from voip.ms, but I don't
know how to tel voip.ms to send the calls to my asterisk with the IAX
protocol.

I understand this is probably a question for the voip.ms folks, but since a
couple of people mentioned earlier that they were rocking with IAX, I
thought it would be an easy question for them to point me in the right
direction.

Thanks.

On Tue, Sep 13, 2011 at 3:32 PM, Dave Aibel <daibel at pervasivetelecom.com>wrote:

> I was lurking in this conversation and I went to look more carefully
> at the voip.ms site. I found sample files at
> http://wiki.voip.ms/article/PBXs#Asterisk_.28SIP.29
>
> Hope that helps.
>
>
> On Tue, Sep 13, 2011 at 3:59 PM, naren <naren.salem at gmail.com> wrote:
> > I see the section you are talking about. It is on the home page if I am
> not
> > logged in. I see the Authentication section and the text "IAX/SIP
> > registration", but it doesn't seem to be a link. I am not sure how I can
> > find the page that has the details about the IAX/SIP registration. I see
> in
> > the wiki there is a page that has the configuration info for iax.conf and
> > extensions.conf.
> > Thanks for your help.
> > naren
> >
> > On Tue, Sep 13, 2011 at 2:25 PM, Danny Nicholas <danny at debsinc.com>
> wrote:
> >>
> >> Did you read the “IAX/SIP registration” section (under Authentication)
> on
> >> voip.ms?
> >>
> >>
> >>
> >> From: asterisk-users-bounces at lists.digium.com
> >> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of naren
> >> Sent: Tuesday, September 13, 2011 2:22 PM
> >> To: John Novack
> >> Cc: Asterisk Users Mailing List - Non-Commercial Discussion
> >> Subject: Re: [asterisk-users] Question about voip.ms service.
> >>
> >>
> >>
> >> Ok... this is probably a dumb question but I can't figure out how to set
> >> voip.ms to use IAX for my DID... with SIP I was able to specify the URI
> so I
> >> pointed it to my asterisk installation, but with IAX I don't have that
> >> option. Is that supposed to work some other way?
> >>
> >>
> >>
> >> Thanks a bunch!
> >>
> >> On Mon, Sep 12, 2011 at 11:18 PM, naren <naren.salem at gmail.com> wrote:
> >>
> >> I am novice with Asterisk, I had to piece together a lot of bits of info
> >> from lots of internet searches to get my very basic setup working. I
> >> probably shouldn't say that because it seems like Nat is not a very
> basic
> >> setup with Asterisk.
> >>
> >>
> >>
> >> The reason for wanting to stay with SIP is because I have my setup
> working
> >> with that protocol with an incoming and an outgoing line. I just wanted
> to
> >> add a second outgoing with voip.ms.
> >>
> >>
> >>
> >> But, I have come so far, so well why not... I will give IAX a shot, and
> >> see what traps I need to wade through :)
> >>
> >>
> >>
> >> Thanks
> >>
> >>
> >>
> >> On Mon, Sep 12, 2011 at 11:09 AM, John Novack
> >> <jnovack at stromberg-carlson.org> wrote:
> >>
> >> Never have had a problem with their IAX service.
> >>
> >> And ( for now ) a little hedge against the hackers.
> >>
> >> Since Asterisk is involved, why not use IAX anyway?
> >>
> >>
> >> John Novack
> >>
> >>
> >> naren wrote:
> >>
> >>
> >>
> >> I also found this... seems like voip.ms outbound is broken for now!
> >>
> >>
> >>
> >> http://pbxinaflash.com/forum/showthread.php?t=10735
> >>
> >>
> >>
> >>
> >>
> >> On Sun, Sep 11, 2011 at 10:34 PM, naren <naren.salem at gmail.com> wrote:
> >>
> >> Hi,
> >>
> >>
> >>
> >> I am trying to set up my asterisk 1.8.5 with voip.ms. I had no problem
> >> with the incoming, but my outgoing is not working. If at all possible, I
> >> would like to stick with SIP. Since the original poster (Glen) had
> mentioned
> >> that he had gotten outgoing working, I was wondering if you would be
> kind
> >> enough to post some thoughts on that. Were you able to get it working
> with
> >> just the default example sip.conf / extensions.conf settings that they
> have
> >> on their website?
> >>
> >>
> >>
> >> I have pretty much the same settings. When I dial out, the destination
> >> rings, but I can't hear a ringback tone from on the source side ( I am
> using
> >> a PAP2T router with a phone). I have set up outgoing with actionvoip
> before
> >> and that is working fine, so I am thinking my router settings for my
> ports
> >> are correct - but I am no expert.
> >>
> >>
> >>
> >> I would really appreciate it if you could post the relevant section of
> >> your sip.conf for me.
> >>
> >>
> >>
> >> Thanks!
> >>
> >> Naren
> >>
> >>
> >>
> >> On Thu, Jun 9, 2011 at 3:22 PM, Steve Edwards <
> asterisk.org at sedwards.com>
> >> wrote:
> >>
> >> On Thu, 9 Jun 2011, John Novack wrote:
> >>
> >> I use voip.ms and have no issues using IAX and Asterisk 1.4.xx
> >>
> >>
> >>
> >> 'slam-dunk.'
> >>
> >>
> >>
> >> Though they suggest SIP, I chose IAX and have 4569 UDP open in my
> firewall
> >>
> >> a
> >>
> >> Their on line config samples just work!
> >>
> >>
> >>
> >> is
> >>
> >>
> >>
> >> Suggest you check your firewall and your configs, and above all post
> some
> >> more information
> >>
> >>
> >>
> >> IAX
> >>
> >>
> >>
> >> If you really want to upset some, top post as I have just done!
> >>
> >>
> >>
> >> Agreed.
> >>
> >>
> >>
> >> The real issue is communication, top bottom or in the middle
> >>
> >>
> >>
> >> Sometimes, it's just about being considerate to 'the next guy.'
> >>
> >> --
> >> Thanks in advance,
> >>
> -------------------------------------------------------------------------
> >> Steve Edwards       sedwards at sedwards.com      Voice: +1-760-468-3867PST
> >> Newline                                              Fax:
> +1-760-731-3000
> >>
> >> --
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> >> Dog is my Co-pilot
> >>
> >>
> >>
> >>
> >>
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>
>
>
> --
> +++++++++++++++++++++++++++++++++++++++++
> Dave Aibel
>
> President & CEO
> Pervasive Telecommunications, Inc.
>
> email: daibel at pervasivetelecom.com
>
> (603)367.3512
> (603)367.9942
> (401)862.4203 (c)
>
> daibel at pervasivetelcom.com
>
> --
> _____________________________________________________________________
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