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

Danny Nicholas danny at debsinc.com
Tue Sep 13 16:14:23 CDT 2011


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-bounces at lists.digium.com [mailto:asterisk-users-bounces at 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:mysipid at 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 <daibel at 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.salem at 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 <danny at debsinc.com> wrote:
>>
>> 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,
>> -------------------------------------------------------------------------
>> Steve Edwards       sedwards at sedwards.com      Voice: +1-760-468-3867 <tel:%2B1-760-468-3867>  PST
>> Newline                                              Fax: +1-760-731-3000 <tel:%2B1-760-731-3000> 
>>
>> --
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>
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--

+++++++++++++++++++++++++++++++++++++++++
Dave Aibel

President & CEO
Pervasive Telecommunications, Inc.

email: daibel at pervasivetelecom.com

(603)367.3512 <tel:%28603%29367.3512> 
(603)367.9942 <tel:%28603%29367.9942> 
(401)862.4203 <tel:%28401%29862.4203>  (c)

daibel at pervasivetelcom.com


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