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