<div dir="ltr">Stephen,<div style="padding-right: 1px; "> What exactly are you trying to accomplish? If you want basic call in/out you're just about there. Changes need to be made in your extensions.conf. Your phones, by default, are in the [default] context. In other words when making a call it looks for extensions here. To allow outbound calling include your outgoing context within the default. To include it, at the bottom of the default context add "include => outgoing" either of these should allow outgoing calling. As for incoming, add a Goto as follows.</div>
<div style="padding-right: 1px; "><br></div><div style="padding-right: 1px; "><span class="Apple-style-span" style="border-collapse: collapse; ">[inbound]<br>exten => 9045622082,1,Answer</span></div><div style="padding-right: 1px; ">
<span class="Apple-style-span" style="border-collapse: collapse;">exten => 9045622082,n,Goto(default,101,1)</span></div><div style="padding-right: 1px; "><span class="Apple-style-span" style="border-collapse: collapse;"><br>
</span></div><div style="padding-right: 1px; "><span class="Apple-style-span" style="border-collapse: collapse;">That equates to "goto the default context, extension 101, at the 1st priority" which is your Dial command.</span></div>
<div><br></div>Best Regards,<div>Darren Severino</div><div><br></div><div><br><div class="gmail_quote">On Sat, Oct 4, 2008 at 1:30 PM, Stephen Reese <span dir="ltr"><<a href="mailto:rsreese@gmail.com">rsreese@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">I have a Asterisk server setup and I am able to connect to the server<br>
using a soft client 'x-lite' and call and leave a message on my second<br>
extension 102. I have setup a Vitelity account and add what I believe<br>
to be the correct information to my sip.conf and extension.conf. I<br>
would like to setup incoming and outgoing calls with voicemail<br>
support. I've searched all over but many of the full configurations<br>
that are available are a bit complex. Any tips or recommendations to<br>
get up and running would be great.<br>
<br>
sip.conf<br>
Code:<br>
<br>
[general]<br>
register => <a href="http://rsreese:pass@inbound18.vitelity.net:5060" target="_blank">rsreese:pass@inbound18.vitelity.net:5060</a><br>
context=default ; Default context for incoming calls<br>
realm=<a href="http://ns1.neocipher.net" target="_blank">ns1.neocipher.net</a> ; Realm for digest authentication<br>
bindport=5060 ; UDP Port to bind to (SIP standard<br>
port is 5060)<br>
bindaddr=<a href="http://0.0.0.0" target="_blank">0.0.0.0</a> ; IP address to bind to (<a href="http://0.0.0.0" target="_blank">0.0.0.0</a> binds to all)<br>
srvlookup=yes ; Enable DNS SRV lookups on outbound calls<br>
domain=<a href="http://neocipher.net" target="_blank">neocipher.net</a> ; Set default domain for this host<br>
[101]<br>
type=friend ; allows incoming and outgoing calls<br>
username=101<br>
secret=test81<br>
mailbox=101<br>
callerid="Stephen" <101><br>
host=dynamic<br>
dtmfmode=rfc2833<br>
canreinvite=no<br>
reinvite=no<br>
disallow=all<br>
allow=gsm<br>
[102]<br>
type=friend ; allows incoming and outgoing calls<br>
username=102<br>
secret=test81<br>
mailbox=102<br>
callerid=("Bob" <101>)<br>
host=dynamic<br>
dtmfmode=rfc2833<br>
canreinvite=yes<br>
allowguest=yes<br>
insecure=very<br>
promiscredir=yes<br>
musicclass=default ; Sets the default music on hold class<br>
for all SIP calls<br>
[authentication]<br>
[vitel-inbound] ;(exact format/casing required)<br>
type=friend<br>
host=<a href="http://inbound18.vitelity.net" target="_blank">inbound18.vitelity.net</a><br>
context=inbound ;(ext-did or from-trunk for A@H)<br>
username=rsreese<br>
secret=pass<br>
allow=all<br>
insecure=very<br>
canreinvite=no<br>
[vitel-outbound] ;(exact format/casing required)<br>
type=friend<br>
host=<a href="http://outbound.vitelity.net" target="_blank">outbound.vitelity.net</a><br>
context=inbound ;(ext-did or from-trunk for A@H)<br>
username=rsreese<br>
fromuser=rsreese<br>
trustrpid=yes<br>
sendrpid=yes<br>
secret=pass<br>
allow=all<br>
canreinvite=no<br>
<br>
<br>
extensions.conf<br>
Code:<br>
<br>
[general]<br>
static=yes<br>
writeprotect=yes<br>
<br>
[globals]<br>
<br>
[default]<br>
<br>
exten => 101,1,Dial(SIP/101,20)<br>
exten => 101,2,Voicemail(102)<br>
<br>
exten => 102,1,Dial(SIP/102,20)<br>
exten => 102,2,Voicemail(102)<br>
<br>
exten=>*98,1,VoiceMailMain(${CALLERIDNUM}@${CONTEXT}) ;This<br>
automatically calls the right mailbox using the ${CALLERIDNUM}<br>
variable in the current context (var ${CONTEXT}).<br>
<br>
[outgoing]<br>
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@vitel-outbound)<br>
exten => _011.,1,Dial(SIP/${EXTEN}@vitel-outbound)<br>
<br>
exten => _911,1,Dial(SIP/911@vitel-outbound)<br>
<br>
[inbound]<br>
exten => 9045622082,1,Answer<br>
<br>
<br>
voicemail.conf<br>
Code:<br>
<br>
[general]<br>
format=wav49|gsm|wav<br>
serveremail=asterisk<br>
attach=yes<br>
skipms=3000<br>
maxsilence=10<br>
silencethreshold=128<br>
maxlogins=3<br>
emaildateformat=%A, %B %d, %Y at %r<br>
sendvoicemail=yes ; Context to Send voicemail from [option 5<br>
from the advanced menu]<br>
[zonemessages]<br>
eastern=America/New_York|'vm-received' Q 'digits/at' IMp<br>
central=America/Chicago|'vm-received' Q 'digits/at' IMp<br>
central24=America/Chicago|'vm-received' q 'digits/at' H N 'hours'<br>
military=Zulu|'vm-received' q 'digits/at' H N 'hours' 'phonetic/z_p'<br>
[default]<br>
101 => 123,Stephen Rese,<a href="mailto:rsreese@gmail.com">rsreese@gmail.com</a><br>
102 => 123,Bob Dole,<a href="mailto:rsreese@gmail.com">rsreese@gmail.com</a><br>
1234 => 4242,Example Mailbox,root@localhost<br>
[other]<br>
1234 => 5678,Company2 User,root@localhost<br>
<br>
_______________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
<br>
AstriCon 2008 - September 22 - 25 Phoenix, Arizona<br>
Register Now: <a href="http://www.astricon.net" target="_blank">http://www.astricon.net</a><br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
</blockquote></div><br></div></div>