[asterisk-users] Question about voip.ms service.
naren
naren.salem at gmail.com
Tue Sep 13 14:22:00 CDT 2011
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,
>>>>
>>>> -------------------------------------------------------------------------
>>>> Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867PST
>>>> Newline Fax:
>>>> +1-760-731-3000
>>>>
>>>>
>>>> --
>>>> _____________________________________________________________________
>>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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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
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>> --
>>
>> Dog is my Co-pilot
>>
>>
>
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