Hello, I've been using Asterisk for a while now. For a large project I think about using SER, too. But although I have studied the SER tutorial, I'm not quite sure, how Asterisk and SER work together, how Asterisk know about clients that are registered at the SER and so on. Can anyone of you recommend a document or tutorial that explains this stuff for a dummy like me ? Thanks in advance. Ralf -- ___________________________________________________ Play 100s of games for FREE! http://games.mail.com/
Dear Ralf We have a few large installations that are using Asterisk and SER managed by our Open Source software AstBill. It is working exelent. Basically Asterisk is handeling the PSTN and Voicemail part. The authentication in Asterisk is done using ANI/CLI. This setup is not very well documented yet so we have to work together for you to get it running. But it is a very powerfull and stable combination. Just contact me off list and I give you more info. Are Casilla -- http://astartelecom.com - Independent VOIP Telecoms Broker. Asterisk and Drupal Consultants http://astbill.com - Billing, Routing and Management software for Asterisk and VOIP AstBill DEMO: http://demo.astbill.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051025/094c9841/attachment.htm
trixter aka Bret McDanel
2005-Oct-25 00:48 UTC
[Asterisk-Users] Asterisk & SER for dummies ?
On Tue, 2005-10-25 at 08:27 +0100, Are wrote:> The authentication in Asterisk is done using ANI/CLI. >Same way as broadvoice, wonder if using that setup if I set my caller id to someone else will it cause the INVITE that broadvoice does (broadvoice will invite the person registered as that account if you try to make a call on their CID, asterisk ignores that invite, I am not so sure if all devices will) -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20051025/3b046835/attachment.pgp
Good Question. We have tested it with any combination we can think about and it is working safely. There is no way (we know about) that you can pass toll free calls. :-) Basically SER is configured to only accept clients that have the same callerid as account numbers so SER refuse to pass the call if you try to be smart. Asterisk only passes the call if you have a valid account and the request is handed over from the SER server. Asterisk determine the max length of the call based on the Users Account balance in AstBill. Are Casilla -- http://astartelecom.com - Independent VOIP Telecoms Broker. Asterisk and Drupal Consultants http://astbill.com - Billing, Routing and Management software for Asterisk and VOIP AstBill DEMO: http://demo.astbill.com On 10/25/05, trixter aka Bret McDanel <trixter@0xdecafbad.com> wrote:> > On Tue, 2005-10-25 at 08:27 +0100, Are wrote: > > The authentication in Asterisk is done using ANI/CLI. > > > Same way as broadvoice, wonder if using that setup if I set my caller id > to someone else will it cause the INVITE that broadvoice does > (broadvoice will invite the person registered as that account if you try > to make a call on their CID, asterisk ignores that invite, I am not so > sure if all devices will) > > -- > Trixter http://www.0xdecafbad.com Bret McDanel > UK +44 870 340 4605 Germany +49 801 777 555 3402 > US +1 360 207 0479 or +1 516 687 5200 > FreeWorldDialup: 635378 > > > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.1 (GNU/Linux) > > iD8DBQBDXeNg+1olxlzQw5cRApWJAJ4sXCutFLLuAk26jzumrS/ioMiZ3ACfa8zZ > IBWJRwuEQ1RN9EqRvajQG/c> =DzJ5 > -----END PGP SIGNATURE----- > > > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com <http://Easynews.com>-- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051025/da56dec5/attachment.htm
Hi Have you got SER up and running If so then get asterisk up and running Then make sure ser can route to asterisk , search in google for routing to voicemail from ser, lots of people do that Now the call will be in asterisk, you will need to allow ser to pass calls, and vice versa ser needs to be told that asterisk is friendly. This is of course not the best user guide, but there isnt really one I have seen. Get the first two points up and running, post again... Iqbal Ralf Mueller wrote:>Hello, > >I've been using Asterisk for a while now. For a large project I think about using SER, too. >But although I have studied the SER tutorial, I'm not quite sure, how Asterisk and SER work >together, how Asterisk know about clients that are registered at the SER and so on. > >Can anyone of you recommend a document or tutorial that explains this stuff for a dummy like me ? > >Thanks in advance. > >Ralf > > >
Are, Are you using a common database for SER and Asterisk? How are you keeping the accounts synced? Does this setup cause any complications with AstBill? regards, David On 10/25/05, Are <london3@gmail.com> wrote:> Good Question. > > We have tested it with any combination we can think about and it is working > safely. There is no way (we know about) that you can pass toll free calls. > :-) > > Basically SER is configured to only accept clients that have the same > callerid as account numbers so SER refuse to pass the call if you try to be > smart. Asterisk only passes the call if you have a valid account and the > request is handed over from the SER server. Asterisk determine the max > length of the call based on the Users Account balance in AstBill. > > Are Casilla -- > http://astartelecom.com - Independent VOIP Telecoms Broker. Asterisk and > Drupal Consultants > http://astbill.com - Billing, Routing and Management software for Asterisk > and VOIP > AstBill DEMO: http://demo.astbill.com > > > > On 10/25/05, trixter aka Bret McDanel <trixter@0xdecafbad.com> wrote: > > > > On Tue, 2005-10-25 at 08:27 +0100, Are wrote: > > > The authentication in Asterisk is done using ANI/CLI. > > > > > Same way as broadvoice, wonder if using that setup if I set my caller id > > to someone else will it cause the INVITE that broadvoice does > > (broadvoice will invite the person registered as that account if you try > > to make a call on their CID, asterisk ignores that invite, I am not so > > sure if all devices will) > > > > -- > > Trixter http://www.0xdecafbad.com Bret McDanel > > UK +44 870 340 4605 Germany +49 801 777 555 3402 > > US +1 360 207 0479 or +1 516 687 5200 > > FreeWorldDialup: 635378 > > > > > > -----BEGIN PGP SIGNATURE----- > > Version: GnuPG v1.4.1 (GNU/Linux) > > > > > iD8DBQBDXeNg+1olxlzQw5cRApWJAJ4sXCutFLLuAk26jzumrS/ioMiZ3ACfa8zZ > > IBWJRwuEQ1RN9EqRvajQG/c> > =DzJ5 > > -----END PGP SIGNATURE----- > > > > > > _______________________________________________ > > --Bandwidth and Colocation sponsored by Easynews.com -- > > > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > >