Re: [Serusers] SER and Asterisk as voicemail

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 > >