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I'm presently working up an Asterisk voice mail set up which runs in a
SER environment myself.<br>
<br>
One problem I ran into was that in the "typical" SER set up is that SER
calls Asterisk with a textual SIP URI in the form of
<a class="moz-txt-link-rfc2396E" href="sip:text_user_name@domain">"sip:text_user_name@domain"</a>, where Asterisk is really designed for
traditional numerical phone numbers. <br>
<br>
By "typical" SER set up, I mean one in which users are defined as
<a class="moz-txt-link-rfc2396E" href="sip:testname@domain">"sip:testname@domain"</a> and numerical phone numbers are simply SER
aliases which point to the textual SIP URIs. This is kind of nice
since a user is identified as a name instead of a number, and it can
match the users email address, etc.<br>
<br>
The problem with this sort of set up is that when you configure SER to
call Asterisk for voice mail, even if a numerical extension was dialed
for that user, SER translates it to the "name" form and uses it in the
INVITE message to Asterisk (even though it leaves the To: header as the
originally dialed numeric #). Asterisk's SIP module takes this INVITE
as the dialed extension, so you wind up with Asterisk searching for a
user name instead of a numeric extension. You CAN actually set up
textual extensions to match these types of calls, but that's kinda
"wrong" in Asterisk, and there's a possibility that the text user names
could collide with some of Asterisk's reserved text extensions. Also,
to do this you must then have an entry for every single user in
extensions.conf, and map them individually to numeric extensions. Not
very good.<br>
<br>
To solve this problem, I wound up using Asterisk's AGI interface to
write a Perl script which when all numeric matches fail, is called and
maps the SIP user name back into a numeric extension by calling the SER
aliases database. <br>
<br>
For example, user dials "5502" on his phone, SER translates this into
<a class="moz-txt-link-rfc2396E" href="sip:joeuser@somedomain.com">"sip:joeuser@somedomain.com"</a>, and procedes with the routing. Then,
when the user doesn't answer the phone (or there's no location info for
him), it gets routed to the Asterisk VM server. Asterisk then sees the
dialed extension as "joeuser" instead of the original "5502", and falls
through all the numerical extension matches to my AGI call in the
extensions file. This then looks up <a class="moz-txt-link-rfc2396E" href="sip:joeuser@somedomain.com">"sip:joeuser@somedomain.com"</a> in
the aliases table, and returns "5502" in a Asterisk variable using
AGI. Then the call can be properly routed to a voice mail box. Here's
what the actual extensions.conf rules look like:<br>
<blockquote><tt>exten =>
_.,1,AGI(map-ser-aliases-contact-to-ast-extension.agi)<br>
exten => _.,2,GotoIf($[${rewriteext} = NOTFOUND]?3:4)<br>
exten => _.,3,Goto(vm-prompt,s,1)<br>
exten => _.,4,Voicemail(u${rewriteext})<br>
exten => _.,5,Wait(1)<br>
exten => _.,6,Hangup<br>
</tt></blockquote>
<br>
If the extension is found, the variable is set to the numeric extension
(if there are more than one, it picks the lowest numerical extension
returned by the database). If no matching entry is found, or the entry
is textual, the variable is set to "NOTFOUND" and the extensions.conf
matching code can do the appropriate thing.<br>
<br>
This is working pretty good in my test environment, but I havn't put it
into actual production yet.<br>
<br>
I really wish Asterisk had a similar set up for the voicemail.conf
file. For now, I plan to write another Perl script which can generate
entries for this file by reading info from the SER database.<br>
<br>
Let me know if any of you guys have interest in this...<br>
<br>
- Jim<br>
<br>
<br>
Jesus Rodriguez wrote:<br>
<blockquote type="cite" cite="mid20031114170207.E28117@camel.jerocu.net">
<pre wrap="">On Fri, 14 Nov 2003, Alessio Focardi wrote:
Hi Alessio,
</pre>
<blockquote type="cite">
<pre wrap="">I imagined the redirection part just like you say, but from what I have understood about asterisk I will have to set up
an extension and a mailbox for every ser user. This looks "time consuming" :)
</pre>
</blockquote>
<pre wrap=""><!---->
Not really. You don't need an entry in extensions.conf for every mailbox/user.
You can use something like this:
[default]
;mapping from 34... to 93390...
exten => _34XXXXXXXX,1,Goto(9339059${EXTEN:8},1)
;voicemail extensions
exten => _9339059XX,1,Wait(2)
exten => _9339059XX,2,Voicemail2(u${EXTEN})
And then in voicemail.conf :
933905903 => 1003,User one, <a class="moz-txt-link-abbreviated" href="mailto:user1@voztele.com">user1@voztele.com</a>
933905904 => 1004,User two, <a class="moz-txt-link-abbreviated" href="mailto:user2@voztele.com">user2@voztele.com</a>
933905905 => 1005,User three, <a class="moz-txt-link-abbreviated" href="mailto:user3@voztele.com">user3@voztele.com</a>
933905906 => 1006,User four, <a class="moz-txt-link-abbreviated" href="mailto:user4@voztele.com">user4@voztele.com</a>
</pre>
<blockquote type="cite">
<pre wrap="">Also: What about playback of recorded messages ?
</pre>
</blockquote>
<pre wrap=""><!---->
Works ok, You can configure the codec/s you allow and the audio format of the
recorded message.
</pre>
<blockquote type="cite">
<pre wrap="">If anyone has asterisk config files to share ... just to get an idea!
</pre>
</blockquote>
<pre wrap=""><!---->
Hope this helps.
</pre>
<blockquote type="cite">
<pre wrap="">BAI> Hi,
BAI> I have some experience with setting ser to use Asterisk as voicemail
BAI> system and it worked without any problem. What you have to do is to
BAI> configure ser to redirect all call for unregistered or busy users to the
BAI> address (IP:port) of your asterisk voicemail box. No special issues... :-)
BAI> Bogdan
BAI> Alessio Focardi wrote:
</pre>
<blockquote type="cite">
<blockquote type="cite">
<pre wrap="">HI,
is there anyone using Asterisk as voicemail backend for SER ?
I would like to know if this is possible and what can I expect from
this integration ...
Tnx !
</pre>
</blockquote>
</blockquote>
<pre wrap="">
--
Best regards,
Alessio <a class="moz-txt-link-freetext" href="mailto:alessiof@interconnessioni.it">mailto:alessiof@interconnessioni.it</a>
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</pre>
</blockquote>
<pre wrap=""><!---->
Saludos
JesusR.
-------------------------------
Jesus Rodriguez
VozTelecom Sistemas, S.L.
<a class="moz-txt-link-abbreviated" href="mailto:jesusr@voztele.com">jesusr@voztele.com</a>
<a class="moz-txt-link-freetext" href="http://www.voztele.com">http://www.voztele.com</a>
Tel. 902360305
-------------------------------
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</pre>
</blockquote>
<br>
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</pre>
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