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 <[email protected]> wrote: > Did you read the “IAX/SIP registration” section (under Authentication) on > voip.ms? **** > > ** ** > > *From:* [email protected] [mailto: > [email protected]] *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 <[email protected]> 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 < > [email protected]> 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 <[email protected]> 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 <[email protected]> > 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 [email protected] Voice: +1-760-468-3867 PST > Newline Fax: +1-760-731-3000 > **** > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users**** > > ** ** > > > > **** > > --**** > > _____________________________________________________________________**** > > -- Bandwidth and Colocation Provided by http://www.api-digital.com --**** > > New to Asterisk? Join us for a live introductory webinar every Thurs:**** > > http://www.asterisk.org/hello**** > > ** ** > > asterisk-users mailing list**** > > To UNSUBSCRIBE or update options visit:**** > > http://lists.digium.com/mailman/listinfo/asterisk-users**** > > ** ** > > -- **** > > ** ** > > Dog is my Co-pilot**** > > ** ** > > ** ** > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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