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 <http://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 <[email protected] 
<mailto:[email protected]>> wrote:

    Hi,

    I am trying to set up my asterisk 1.8.5 with voip.ms <http://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 <[email protected] 
<mailto:[email protected]>> wrote:

        On Thu, 9 Jun 2011, John Novack wrote:

            I use voip.ms <http://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 [email protected] <mailto:[email protected]>      
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

Reply via email to