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 >> <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 > -- +++++++++++++++++++++++++++++++++++++++++ Dave Aibel President & CEO Pervasive Telecommunications, Inc. email: dai...@pervasivetelecom.com (603)367.3512 <tel:%28603%29367.3512> (603)367.9942 <tel:%28603%29367.9942> (401)862.4203 <tel:%28401%29862.4203> (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