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: +1-760-468-3867PST >>> 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
