I'm using them for inbound and outbound on Asterisk and FreeSwitch Sent from my iPhone
On Sep 13, 2011, at 5:14 PM, "Danny Nicholas" <da...@debsinc.com> wrote: > That’s what this part of extensions.conf should do: > > ; inbound context example for your DID numbers, do not add the number 1 in > front > > > > [voipms-inbound] > > exten => 7863643011,1,Answer() ;your DID > > > > > > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of naren > Sent: Tuesday, September 13, 2011 4:09 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Question about voip.ms service. > > > > Yup, that part I got. What I am not clear about is how to set up the DID to > go to my URI. When I select "manage DIDs" and click on the one I want to > change, I see the following options for routing the DID > > > > x SIP/IAX - [main account] IAX2/100000 <- with my account number > > x SIP URI - SIP:mysi...@myuri.com:5060 > > x System - Hangup > > > > There are several other options but they are not selectable for me because I > have not set up to use them. > > > > I used to have the routing set to SIP URI where I was able to specify my URI > where the call was routed to. But with the SIP/IAX option I do not have that > ability. > > > > I am missing something fundamental here. My asterisk has the iax.conf and > extensions.conf entries ready to receive calls from voip.ms, but I don't know > how to tel voip.ms to send the calls to my asterisk with the IAX protocol. > > > > I understand this is probably a question for the voip.ms folks, but since a > couple of people mentioned earlier that they were rocking with IAX, I thought > it would be an easy question for them to point me in the right direction. > > > > Thanks. > > On Tue, Sep 13, 2011 at 3:32 PM, Dave Aibel <dai...@pervasivetelecom.com> > wrote: > > I was lurking in this conversation and I went to look more carefully > at the voip.ms site. I found sample files at > http://wiki.voip.ms/article/PBXs#Asterisk_.28SIP.29 > > Hope that helps. > > > > On Tue, Sep 13, 2011 at 3:59 PM, naren <naren.sa...@gmail.com> wrote: > > 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 <da...@debsinc.com> wrote: > >> > >> Did you read the “IAX/SIP registration” section (under Authentication) on > >> voip.ms? > >> > >> > >> > >> From: asterisk-users-boun...@lists.digium.com > >> [mailto:asterisk-users-boun...@lists.digium.com] 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 <naren.sa...@gmail.com> 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 > >> <jnov...@stromberg-carlson.org> 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 <naren.sa...@gmail.com> 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 <asterisk....@sedwards.com> > >> 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 sedwa...@sedwards.com 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 > > > > > > -- > > _____________________________________________________________________ > > -- 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 > > > > > > -- > > +++++++++++++++++++++++++++++++++++++++++ > Dave Aibel > > President & CEO > Pervasive Telecommunications, Inc. > > email: dai...@pervasivetelecom.com > > (603)367.3512 > (603)367.9942 > (401)862.4203 (c) > > dai...@pervasivetelcom.com > > > -- > _____________________________________________________________________ > -- 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
-- _____________________________________________________________________ -- 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