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

[naren]

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,
>>> -------------------------------------------------------------------------
>>> Steve Edwards       sedwards at sedwards.com      Voice: +1-760-468-3867PST
>>> Newline                                              Fax:
>>> +1-760-731-3000
>>>
>>>
>>> --
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>>
>>
>
>
> --
> _____________________________________________________________________
> -- 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
>
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>
> --
>
> Dog is my Co-pilot
>
>
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