On Fri, 14 Nov 2003, Alessio Focardi wrote:

Hi Alessio,

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" :)

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, user1@voztele.com 933905904 => 1004,User two, user2@voztele.com 933905905 => 1005,User three, user3@voztele.com 933905906 => 1006,User four, user4@voztele.com

Also: What about playback of recorded messages ?

Works ok, You can configure the codec/s you allow and the audio format of the recorded message.

If anyone has asterisk config files to share ... just to get an idea!

Hope this helps.

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:

...

...

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 !

-- Best regards, Alessio mailto:alessiof@interconnessioni.it

---

Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

Saludos JesusR.

------------------------------- Jesus Rodriguez VozTelecom Sistemas, S.L. jesusr@voztele.com http://www.voztele.com Tel. 902360305 -------------------------------

On Fri, 14 Nov 2003, Alessio Focardi wrote:

Hello,

JR> Not really. You don't need an entry in extensions.conf for every mailbox/user. JR> You can use something like this:

JR> [default] JR> ;mapping from 34... to 93390... exten =>> _34XXXXXXXX,1,Goto(9339059${EXTEN:8},1)

I'm not getting it, sorry, surely because I know nothing about asterisk.

If my numbering plan for ser is from 0000000 to 9999999 how this tranwlates in Asterisk ?

Sorry, this was not the best example as we are using a combination of user id and pstn number. This does not apply with your numbering plan.

JR> ;voicemail extensions exten =>> _9339059XX,1,Wait(2) exten =>> _9339059XX,2,Voicemail2(u${EXTEN})

Idem ! :)

Idem too.. :)

JR> And then in voicemail.conf :

933905903 =>> 1003,User one, user1@voztele.com 933905904 =>> 1004,User two, user2@voztele.com 933905905 =>> 1005,User three, user3@voztele.com 933905906 =>> 1006,User four, user4@voztele.com

One entry for every user, like I was fearing ... Is this cfg parsed only at Asterisk start ? I mean, if its parsed dinamycally maybe I can write a script to update it every time I add an user to ser....

I meant that you don't need to create one entry per user in extensions.conf .

I think it's only readed at startup but you can use the "reload" command when needed without disturbing current connections/conversations/sessions.

By the way, you have mysql support in voicemail2 when compiled with USEMYSQLVM .

Saludos JesusR.

------------------------------- Jesus Rodriguez VozTelecom Sistemas, S.L. jesusr@voztele.com http://www.voztele.com Tel. 902360305 -------------------------------

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

Jesus Rodriguez wrote:

On Fri, 14 Nov 2003, Alessio Focardi wrote:

Hi Alessio,

...

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" :)

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, user1@voztele.com 933905904 => 1004,User two, user2@voztele.com 933905905 => 1005,User three, user3@voztele.com 933905906 => 1006,User four, user4@voztele.com

...

Also: What about playback of recorded messages ?

Works ok, You can configure the codec/s you allow and the audio format of the recorded message.

...

If anyone has asterisk config files to share ... just to get an idea!

Hope this helps.

...

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:

...

...

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 !

-- Best regards, Alessio mailto:alessiof@interconnessioni.it

---

Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

Saludos JesusR.

---

Jesus Rodriguez VozTelecom Sistemas, S.L. jesusr@voztele.com http://www.voztele.com Tel. 902360305

---

---

Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

Yep. If the existing set up I want to put this VM server into was set up such that all users, subscribers, and phones were set up with numeric user names instead of email-style textual ones, there would be no need for this at all.

But the production environment I'm integrating Asterisk into already has everything set up with text style names. This is easier than going around and changing everyone's phone config, SERweb logins, etc.

- Jim

Nils Ohlmeier wrote:

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