[asterisk-users] Unable to receive Incoming calls to my DID. Please tell me the solution

[Lacy Moore - Aspendora]

I struggled with one provider for a long time until finally realizing my
username on their site was not my username that I was supposed to be using
in sip configuration.  Make sure you are using the right username and
password.  However, it would seem that you would not be able to make an
outgoing call using the wrong username/password combination.

One thing I have not seen in your posts is your firewall information.  Your
firewall may be setup to allow outgoing connections, but not incoming.  I
would not depend on info from a provider.  You may very well be registering
with them, but your firewall may be blocking the incoming call.  If you
think you have no firewall, check again.  IPTABLES might have loaded itself
and it may be blocking.  Try:

service iptables stop

and then try the incoming call again.  I've been burned twice due to this.
Something has changed in the way I configure my linux boxes, and for some
reason iptables is starting.


On 8/14/06, Rich Adamson <radamson at routers.com> wrote:
>
> Crazy Boy wrote:
> > Hi,
> >
> > Thank you for your response. As you said, I executed the command "sip
> > show registry". But, its not showing anything. Teliax people are also
> > telling that my Asterisk server doesn't register with Teliax. So, the
> > final conclusion is "My Asterisk server doesn't register with Teliax".
> > Here I am giving my configuration files. Now, What I have to do to
> > register my Asterisk server with Teliax? Please tell me.
> >
> > SIP.CONF contents:
> >
> > [general]
> > register => xyz.abc:xxxxxxx at voip-co1.teliax.com
> > [authentication]
> > auth =  xyz.abc:xxxxxxx at voip-co1.teliax.com
>
> Double check the above two statements to ensure the userid and password
> are exactly those provided to you by teliax. There is nothing else in
> your config that impacts the register statement with the exception of
> nat'ing.
>
> It would appear from your other config statements that asterisk might be
> located behind a firewall or nat box. If so, read the documentation on
> that, and look at the asterisk/configs/sip.conf.sample file.
> Specifically the section on "NAT SUPPORT".
>
> You might also want to read more about using the diagnostic tools
> available to you within asterisk. Setting verbose and/or debug to a high
> number and copy/paste the CLI output associated with the problem. Or,
> start using the CLI with something like:
>   asterisk -rvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvv
>
> > [teliax-incoming]
> > exten => 3031234567, 1, Answer()
> > exten => 3031234567, 2, Dial(SIP/105,15)
>
> The above has nothing to do with registering with teliax, but you do not
> want to "answer" a call before ringing the sip phone. Take that
> statement out of there. When the sip phone answers an incoming call,
> asterisk will automatically send the answer to teliax.
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
Lacy Moore
Aspendora, Inc.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060814/e27a1ba1/attachment.htm