20 Nov
2003
20 Nov
'03
3:58 a.m.
Hello,
maybe i'm too stupid to understand your problems completly, but wouldn't it be
easier to run your SER with aliases the other way round?
I mean create and run only numerical users, but add alphabetical aliases.
(AFAIK nothing in SER insists on alphabetical usernames or numerical
aliases.) If a request with an alphabetical username in request URI comes in
convert it to numeric via aliases.
But in any case you could just forward the request to your * box without
writing extensions.
Greets
Nils
On Thursday 20 November 2003 02:42, Jim Burwell wrote:
I'm presently working up an Asterisk voice mail
set up which runs in a
SER environment myself.
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
"sip:text_user_name@domain", where Asterisk is really designed for
traditional numerical phone numbers.
By "typical" SER set up, I mean one in which users are defined as
"sip:testname@domain" 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.
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.
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.
For example, user dials "5502" on his phone, SER translates this into
"sip:joeuser@somedomain.com", 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 "sip:joeuser@somedomain.com" 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:
exten => _.,1,AGI(map-ser-aliases-contact-to-ast-extension.agi)
exten => _.,2,GotoIf($[${rewriteext} = NOTFOUND]?3:4)
exten => _.,3,Goto(vm-prompt,s,1)
exten => _.,4,Voicemail(u${rewriteext})
exten => _.,5,Wait(1)
exten => _.,6,Hangup
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.
This is working pretty good in my test environment, but I havn't put it
into actual production yet.
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.
Let me know if any of you guys have interest in this...
- Jim