[asterisk-users] Incoming SIP Calls dumped to non-existent VM no matter what extensions.conf setup is used
Joe Wood
schmoe at gmail.com
Wed Aug 4 20:52:42 CDT 2010
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
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