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<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
