Asterisk@Home - Incoming Calls

Hi, i have asterisk running, outgoing calls are fine, internal calls are
fine. But incoming external calls have me very confused and annoyed!

When i call myself from another SIP device, or a friend does they get
service unavilable in x-lite/pro. The call is then logged in my recent
calls as unanswered, and it claims its route is 's'.

Anyone have any idea what i am going on about, and how i can fix this?

Cheers

Sean wrote:
|| Hi, i have asterisk running, outgoing calls are fine, internal calls
|| are fine. But incoming external calls have me very confused and
|| annoyed!
||
|| When i call myself from another SIP device, or a friend does they get
|| service unavilable in x-lite/pro. The call is then logged in my
|| recent calls as unanswered, and it claims its route is 's'.
||
|| Anyone have any idea what i am going on about, and how i can fix
|| this?
||
|| Cheers

Which company are you using?

I've made use of the tutorial here
http://esupport.gradwell.net/index.p...j=subcat&_i=29 to
good effect for Sipgate.

IIRC, there are a bunch of changes you need to make to stop incoming SIP
calls going to a busy extension (which A@H does by default)......., however,
following the tutorial & setting up the DDIs (DIDs) it all works fine.

Jono wrote:
> Sean wrote:
> || Hi, i have asterisk running, outgoing calls are fine, internal calls
> || are fine. But incoming external calls have me very confused and
> || annoyed!
> ||
> || When i call myself from another SIP device, or a friend does they get
> || service unavilable in x-lite/pro. The call is then logged in my
> || recent calls as unanswered, and it claims its route is 's'.
> ||
> || Anyone have any idea what i am going on about, and how i can fix
> || this?
> ||
> || Cheers
>
> Which company are you using?
>
> I've made use of the tutorial here
> http://esupport.gradwell.net/index.p...j=subcat&_i=29 to
> good effect for Sipgate.
>
> IIRC, there are a bunch of changes you need to make to stop incoming SIP
> calls going to a busy extension (which A@H does by default)......., however,
> following the tutorial & setting up the DDIs (DIDs) it all works fine.
>
>
Thanks, i've got the DID's set up, and it works inernally on the call
queue system, i will however have a good look through that link.

I'm trying to get it working on FWD at the moment:)

I think i will just throw the thing out of the window... Different
places say different things, that gradwell thing says leave incoming
thing blank, others don't.

I've spent out 8 hours, and give up

Sean wrote:
|| I think i will just throw the thing out of the window... Different
|| places say different things, that gradwell thing says leave incoming
|| thing blank, others don't.
||
|| I've spent out 8 hours, and give up

Why not apply for a free Sipgate number & see if it works then?

You'd be able to rule out FWD that way.

I'm assuming you've got all the necessary ports forwarded.

Jono wrote:
> Sean wrote:
> || I think i will just throw the thing out of the window... Different
> || places say different things, that gradwell thing says leave incoming
> || thing blank, others don't.
> ||
> || I've spent out 8 hours, and give up
>
> Why not apply for a free Sipgate number & see if it works then?
>
> You'd be able to rule out FWD that way.
>
> I'm assuming you've got all the necessary ports forwarded.
>
>

Its as as a DMZ.........will try sipgate soon,,...thanks

Outgoing not working on sipgate, and incoming not either :)

Sean wrote:
|| Outgoing not working on sipgate, and incoming not either :)

What have you set up in your Outbound Routes?

I set sipgate as only one, with x. as the dial thing

Ahh, well it's now working!

I don't know how, but it does. I asked a friend who has recently set up
asterisk to login and help. I edited the sip conf page, and it now works!

:):):)

Sean wrote:
|| Ahh, well it's now working!
||
|| I don't know how, but it does. I asked a friend who has recently set
|| up asterisk to login and help. I edited the sip conf page, and it
|| now works!
||
|| :):):)

Great stuff.

What did you change in sip.conf?