[asterisk-users] Question about voip.ms service.

Tue Sep 13 14:25:35 CDT 2011

Did you read the “IAX/SIP registration” section (under Authentication) on voip.ms? 

From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at 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.salem at 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 <jnovack at 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.salem at 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.org at 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,
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Newline                                              Fax: +1-760-731-3000 <tel:%2B1-760-731-3000>  

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