[asterisk-users] Incoming SIP Calls dumped to non-existent VM no matter what extensions.conf setup is used

[Warren Selby]

On Wed, Aug 4, 2010 at 8:52 PM, Joe Wood <schmoe at gmail.com> wrote:

> Hello.
>
> I have been beating my head over this problem for about 6 hours now.
>
> I have a SIP peer, who I register to (successfully), who should be
> directing all incoming calls at my [default] stanza in my
> extensions.conf:
>
> [ Context 'default' created by 'pbx_config' ]
>  's' =>            1. Wait(1)
>  [pbx_config]
>                    2. Answer()
> [pbx_config]
>                    3. Background(welcome)
>  [pbx_config]
>                    4. Background(and)
>  [pbx_config]
>                    5. Background(thank-you-for-calling)
>  [pbx_config]
>                    6. Background(conference-reservations)
>  [pbx_config]
>                    7. Waitfor()
>  [pbx_config]
>                    8. Hangup()
> [pbx_config]
>
> Unfortunately, no matter how I configure extensions.conf or sip.conf,
> the phone call always ends up saying: "Extension is unavailable.
> Please leave your message after the tone".
>
> sip.conf:
>
> [general]
> register => NPANXXZZZZ:PASSWORD at SERVICE_PROVIDER_IP
> registertimeout=29
> registerattempts=0
> defaultexpiry=60
>
> [DID_NUMBER]
> type=peer
> context=default
> host=SERVICE_PROVIDER_IP
> authuser=DID_NUMBER
> fromuser=DID_NUMBER
> fromdomain=SERVICE_PROVIDER_REALM
> remotesecret=SERVICE_PROVIDER_PASSWD
> secret=SERVICE_PROVIDER_PASSWD
> dtmfmode=rfc2833
> disallow=all
> allow=ulaw
> qualify=yes
>
> I am attempting just to get the starting point where I can direct
> users through my asterisk box, but it won't direct users to the 's'
> extention, only to some voicemail box. I've removed the voicemail
> config.
>
> My extensions.conf is tiny:
>
> [globals]
>
> [general]
>
> [default]
> exten => s,1,Wait(1)
> exten => s,n,Answer()
> exten => s,n,Background(welcome)
> exten => s,n,Background(and)
> exten => s,n,Background(thank-you-for-calling)
> exten => s,n,Background(conference-reservations)
> exten => s,n,Waitfor()
> exten => s,n,Hangup()
>
>
> What am I doing wrong here?
>
>
>
> Thanks for any help you can give.
>
>
> Joe
>

You don't have any extensions in your default context that match the
extension that your sip peer is dialing in on.  's' is not a default
extension for SIP...try using _X., and see what you get.  Bump up the CLI
(core set verbose 10) and then repost a failed called attempt.  Some SIP
providers also use a + symbol in front of their inbound calls, so you may
need to use _+X., instead.


-- 
Thanks,
--Warren Selby
http://www.selbytech.com
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