<div class="gmail_quote">On Wed, Aug 4, 2010 at 8:52 PM, Joe Wood <span dir="ltr"><<a href="mailto:schmoe@gmail.com">schmoe@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
Hello.<br>
<br>
I have been beating my head over this problem for about 6 hours now.<br>
<br>
I have a SIP peer, who I register to (successfully), who should be<br>
directing all incoming calls at my [default] stanza in my<br>
extensions.conf:<br>
<br>
[ Context 'default' created by 'pbx_config' ]<br>
's' => 1. Wait(1) [pbx_config]<br>
2. Answer() [pbx_config]<br>
3. Background(welcome) [pbx_config]<br>
4. Background(and) [pbx_config]<br>
5. Background(thank-you-for-calling) [pbx_config]<br>
6. Background(conference-reservations) [pbx_config]<br>
7. Waitfor() [pbx_config]<br>
8. Hangup() [pbx_config]<br>
<br>
Unfortunately, no matter how I configure extensions.conf or sip.conf,<br>
the phone call always ends up saying: "Extension is unavailable.<br>
Please leave your message after the tone".<br>
<br>
sip.conf:<br>
<br>
[general]<br>
register => NPANXXZZZZ:PASSWORD@SERVICE_PROVIDER_IP<br>
registertimeout=29<br>
registerattempts=0<br>
defaultexpiry=60<br>
<br>
[DID_NUMBER]<br>
type=peer<br>
context=default<br>
host=SERVICE_PROVIDER_IP<br>
authuser=DID_NUMBER<br>
fromuser=DID_NUMBER<br>
fromdomain=SERVICE_PROVIDER_REALM<br>
remotesecret=SERVICE_PROVIDER_PASSWD<br>
secret=SERVICE_PROVIDER_PASSWD<br>
dtmfmode=rfc2833<br>
disallow=all<br>
allow=ulaw<br>
qualify=yes<br>
<br>
I am attempting just to get the starting point where I can direct<br>
users through my asterisk box, but it won't direct users to the 's'<br>
extention, only to some voicemail box. I've removed the voicemail<br>
config.<br>
<br>
My extensions.conf is tiny:<br>
<br>
[globals]<br>
<br>
[general]<br>
<br>
[default]<br>
exten => s,1,Wait(1)<br>
exten => s,n,Answer()<br>
exten => s,n,Background(welcome)<br>
exten => s,n,Background(and)<br>
exten => s,n,Background(thank-you-for-calling)<br>
exten => s,n,Background(conference-reservations)<br>
exten => s,n,Waitfor()<br>
exten => s,n,Hangup()<br>
<br>
<br>
What am I doing wrong here?<br>
<br>
<br>
<br>
Thanks for any help you can give.<br>
<br>
<br>
Joe<br></blockquote></div><br>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.<br><br clear="all"><br>-- <br>Thanks,<br>--Warren Selby<br><a href="http://www.selbytech.com">http://www.selbytech.com</a><br>
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