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It is registering with the sip provider. When I do a CLI> sip show
registry, it shows as registered.<br>
<br>
I really think my problem is about contexts. For some reason, I think
I got the flow of how a call comes in and goes out works.<br>
<br>
Is this approximately right?:<br>
<br>
Incoming SIP Calls:<br>
PSTN -> SIP Provider -> ASTERISK -> "incoming" context in
[general] of sip.conf -> matching [incoming] in extensions.conf
-> sent do wherever the dialpan says to go.<br>
<br>
Outgoing SIP Calls:<br>
Phone -> asterisk > outgoing context in [ext] of sip.conf ->
matching [outgoing] context in extensions.con -> sent to do whatever
the dialpan says to do.<br>
<br>
If that's wrong, please let me know :D<br>
<br>
Cheers<br>
<br>
<br>
Carlos Alperin wrote:
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<p class="MsoNormal"><font color="navy" face="Arial" size="2"><span
style="font-size: 10pt; font-family: Arial; color: navy;">That
shouldn’t be complicate, but it
looks like you ‘re not registering with your provider. However, without
the configuration files, it is not much to do for help you.<o:p></o:p></span></font></p>
<p class="MsoNormal"><font color="navy" face="Arial" size="2"><span
style="font-size: 10pt; font-family: Arial; color: navy;"><o:p> </o:p></span></font></p>
<p class="MsoNormal"><font color="navy" face="Arial" size="2"><span
style="font-size: 10pt; font-family: Arial; color: navy;">Carlos
Alperin<o:p></o:p></span></font></p>
<p class="MsoNormal"><font color="navy" face="Arial" size="2"><span
style="font-size: 10pt; font-family: Arial; color: navy;"><o:p> </o:p></span></font></p>
<div>
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color="black" face="Times New Roman" size="3"><span
style="font-size: 12pt; color: windowtext;">
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<p class="MsoNormal"><b><font color="black" face="Tahoma" size="2"><span
style="font-size: 10pt; font-family: Tahoma; color: windowtext; font-weight: bold;">From:</span></font></b><font
color="black" face="Tahoma" size="2"><span
style="font-size: 10pt; font-family: Tahoma; color: windowtext;">
<a class="moz-txt-link-abbreviated" href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>
[<a class="moz-txt-link-freetext" href="mailto:asterisk-users-bounces@lists.digium.com">mailto:asterisk-users-bounces@lists.digium.com</a>] <b><span
style="font-weight: bold;">On Behalf Of </span></b>Pikoro<br>
<b><span style="font-weight: bold;">Sent:</span></b> Wednesday,
October 26, 2005
11:01 PM<br>
<b><span style="font-weight: bold;">To:</span></b>
<a class="moz-txt-link-abbreviated" href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a><br>
<b><span style="font-weight: bold;">Subject:</span></b>
[Asterisk-Users] Simple
SIP only Asterisk Configuration</span></font><font color="black"><span
style="color: windowtext;"><o:p></o:p></span></font></p>
</div>
<p class="MsoNormal"><font color="black" face="Times New Roman"
size="3"><span style="font-size: 12pt;"><o:p> </o:p></span></font></p>
<p class="MsoNormal" style="margin-bottom: 12pt;"><font color="black"
face="Times New Roman" size="3"><span style="font-size: 12pt;">Ok,
I've been messing
with asterisk for the last 3 weeks and I just can't seem to figure this
out.<br>
<br>
Everything I've read seems to state that asterisk will work "out of the
box" with only minor config changes when being used only for SIP to SIP
calls.<br>
<br>
The problem I am having is I cannot make outbound calls or receive
incoming
calls over my sip-provider. Asterisk registers properly, and internal
communications seem to work fine.<br>
<br>
I have, at one time or another, had either outgoing only, or incoming
only, but
never both at once. Unfortunately, I didn't know what I did to make
either of those work since I had made multiple adjustments and had done
a
reload after each change. For some reason, the incoming calls only
started working after restarting the computer so it could have been any
of 50
things I had changed.<br>
<br>
I am back to the sample config files. Is there any kind of walkthrough
for a sip only setup? I have seen SIP only touched on briefly, with
most
of the documentation leaning torwards IAX communications.<br>
<br>
Here is what I am trying to accomplish:<o:p></o:p></span></font></p>
<ol start="1" type="1">
<li class="MsoNormal" style=""><font color="black"
face="Times New Roman" size="3"><span style="font-size: 12pt;">Asterisk
server registers with our sip-provider for sip to pstn local and
international calls<o:p></o:p></span></font></li>
<li class="MsoNormal" style=""><font color="black"
face="Times New Roman" size="3"><span style="font-size: 12pt;">Internal
extensions 0, 200-210 can call eachother (of course)<o:p></o:p></span></font></li>
<li class="MsoNormal" style=""><font color="black"
face="Times New Roman" size="3"><span style="font-size: 12pt;">Extensions
200-205 are in a Tech Support Queue<o:p></o:p></span></font></li>
<li class="MsoNormal" style=""><font color="black"
face="Times New Roman" size="3"><span style="font-size: 12pt;">Extensions
206-210 are in a Customer Support Queue<o:p></o:p></span></font></li>
<li class="MsoNormal" style=""><font color="black"
face="Times New Roman" size="3"><span style="font-size: 12pt;">Extension
0 is the operator or menu system (I guess this would be s?)<o:p></o:p></span></font></li>
<li class="MsoNormal" style=""><font color="black"
face="Times New Roman" size="3"><span style="font-size: 12pt;">All
phones (for now) are x-ten soft phones<o:p></o:p></span></font></li>
<li class="MsoNormal" style=""><font color="black"
face="Times New Roman" size="3"><span style="font-size: 12pt;">Each
extension has voice mail<o:p></o:p></span></font></li>
<li class="MsoNormal" style=""><font color="black"
face="Times New Roman" size="3"><span style="font-size: 12pt;">When a
customer calls during office hours, they are presented with a menu,
press 1 for CS, press 2 for TS, or dial the extension you wish to
reach, etc...<o:p></o:p></span></font></li>
<li class="MsoNormal" style=""><font color="black"
face="Times New Roman" size="3"><span style="font-size: 12pt;">Calls
can be forwarded to other extensions<o:p></o:p></span></font></li>
<li class="MsoNormal" style=""><font color="black"
face="Times New Roman" size="3"><span style="font-size: 12pt;">On-hold
music is implemented<o:p></o:p></span></font></li>
</ol>
<p class="MsoNormal" style="margin-bottom: 12pt;"><font color="black"
face="Times New Roman" size="3"><span style="font-size: 12pt;">I can
handle doing
everything on the list except for #1.<br>
<br>
If anyone can offer any suggestions, it would make me, and my boss,
very happy.<br>
<br>
Thanks in Advance<br>
Aaron<o:p></o:p></span></font></p>
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