I see what you mean. Maybe if you call their support they can tell you what you need to know. If not, voicepulse is a pretty good provider.
From: [email protected] [mailto:[email protected]] On Behalf Of naren Sent: Tuesday, September 13, 2011 2:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Question about voip.ms service. 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 <tel:%2B1-760-468-3867> PST Newline Fax: +1-760-731-3000 <tel:%2B1-760-731-3000> -- _____________________________________________________________________ -- 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
-- _____________________________________________________________________ -- 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
