Hello Im managing a WAN with a lot of Universities. Some of them already installed a VoIP solution based on SER (to manage SIP clients) and Asterisk (for services and PSTN GW). The DNS routing provided by SER is working perfectly, but we want to start routing all calls thru IP transparently. We want our legacy PBXs (that are connected to Asterisk) to forward all calls to IP. The idea is to forward all calls to a central VoIP server, that has all the numbers that already are VoIP enabled, and then: - if the called number is VoIP enabled, he routes the call to that Univ. VoIP server - if the called number isnt in the list, the call goes back to the PBX and a PSTN call is dialed This way, ppl starts using the VoIP infrastructure, without even knowing what VoIP means, and the telecom bill starts decreasing. I know thats a statical and hierarchical structure and we dont want that, but is a good solution for this migration phase, where a lot of places are still using TDM systems. Now, the top of the hierarchy should be an Asterisk or SER? I dont know which of the systems is the best choice for the job. Does someone has an idea of what should we use? Thanks Joao Pereira www.fccn.pt
On Wed, 2005-11-30 at 17:45 +0000, Joao Pereira wrote:> Hello > Im managing a WAN with a lot of Universities. Some of them already > installed a VoIP solution based on SER (to manage SIP clients) and > Asterisk (for services and PSTN GW). The DNS routing provided by SER is > working perfectly, but we want to start routing all calls thru IP > transparently. > We want our legacy PBXs (that are connected to Asterisk) to forward all > calls to IP. The idea is to forward all calls to a central VoIP server, > that has all the numbers that already are VoIP enabled, and then: > - if the called number is VoIP enabled, he routes the call to that Univ. > VoIP server > - if the called number isnt in the list, the call goes back to the PBX > and a PSTN call is dialed >Have you considered enum for the voip enabled phones and failing through to either realtime or extensions.conf if enum fails? tip I found enum is easier to manage with powerdns and the mysql backend (although it can do postgress, isc bind, and other stuff for its backend, it seems faster for me and many have reported a much lower memory footprint when doing thousands of zones). That would seem to accomplish what you want and make it easier to port people over to voip as needed. Infact depending on how you configure everything, everyone could be in enum even the old legacy routes, then its a simple matter of editing what is already there. At least that has been my experience.> Now, the top of the hierarchy should be an Asterisk or SER? I dont know > which of the systems is the best choice for the job. Does someone has an > idea of what should we use? >SER tends to deal with large numbers of sip registrations better than asterisk on the same hardware. Mostly because it is specifically written for just that task. realtime may change that (I havent seen any specific studies done on load issues post realtime so I cant comment as I havent done any personally). -- 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 http://www.sacaug.org/ Sacramento Asterisk Users Group -------------- 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/20051130/dafbaf5c/attachment.pgp
You should take a look to ENUM protocol: http://www.voip-info.org/wiki/view/ENUM. It could provide a decentralized and simple solution for your requirements. Regards -----Mensaje original----- De: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] En nombre de Joao Pereira Enviado el: mi?rcoles, 30 de noviembre de 2005 18:45 Para: serusers@iptel.org; asterisk-users@lists.digium.com Asunto: [Asterisk-Users] hierarchical VoIP system Hello Im managing a WAN with a lot of Universities. Some of them already installed a VoIP solution based on SER (to manage SIP clients) and Asterisk (for services and PSTN GW). The DNS routing provided by SER is working perfectly, but we want to start routing all calls thru IP transparently. We want our legacy PBXs (that are connected to Asterisk) to forward all calls to IP. The idea is to forward all calls to a central VoIP server, that has all the numbers that already are VoIP enabled, and then: - if the called number is VoIP enabled, he routes the call to that Univ. VoIP server - if the called number isnt in the list, the call goes back to the PBX and a PSTN call is dialed This way, ppl starts using the VoIP infrastructure, without even knowing what VoIP means, and the telecom bill starts decreasing. I know thats a statical and hierarchical structure and we dont want that, but is a good solution for this migration phase, where a lot of places are still using TDM systems. Now, the top of the hierarchy should be an Asterisk or SER? I dont know which of the systems is the best choice for the job. Does someone has an idea of what should we use? Thanks Joao Pereira www.fccn.pt _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Hi there! We have kind of the same setup but are using a few number of SER boxes for the on net calls - using enum for the lookup would be a great idea so that you can get the numbers to do sip calls and move over slowly. And for the central routing voip server make the routing use SIP redirects as the central server then can handle a lot of calls as its only doing the routing decisions. Best regards jan --On Wednesday, November 30, 2005 05:45:21 PM +0000 Joao Pereira <joao.pereira@fccn.pt> wrote:> Hello > Im managing a WAN with a lot of Universities. Some of them already > installed a VoIP solution based on SER (to manage SIP clients) and > Asterisk (for services and PSTN GW). The DNS routing provided by SER is > working perfectly, but we want to start routing all calls thru IP > transparently. > We want our legacy PBXs (that are connected to Asterisk) to forward all > calls to IP. The idea is to forward all calls to a central VoIP server, > that has all the numbers that already are VoIP enabled, and then: > - if the called number is VoIP enabled, he routes the call to that Univ. > VoIP server > - if the called number isnt in the list, the call goes back to the PBX > and a PSTN call is dialed > > This way, ppl starts using the VoIP infrastructure, without even knowing > what VoIP means, and the telecom bill starts decreasing. > > I know thats a statical and hierarchical structure and we dont want that, > but is a good solution for this migration phase, where a lot of places > are still using TDM systems. > > Now, the top of the hierarchy should be an Asterisk or SER? I dont know > which of the systems is the best choice for the job. Does someone has an > idea of what should we use? > > Thanks > Joao Pereira > www.fccn.pt > > > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-- +------------------------------------------------------------------- ! Irial / YASK AB ! Att: Jan Saell ! Box 59, S-692 21 KUMLA, SWEDEN ! Tel: 019-58 25 15 Int +46-19 58 25 15 Fax +46-19 58 38 05 ! E-mail: jan@irial.com ! PGP Fingerprint: E957 23C8 9F51 0958 B9AD 7F18 404A 5DA1 F944 A08B -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20051130/92ba808e/attachment.pgp
And about the protocol used to create this hierarchical network? Should I use SIP (routing between SERs) or should I use IAX (routing between Asterisks)? About ENUM, Isnt the managing of the ENUM tree going to be very complicated and heavy when we reach the millions of users? Joao Jan Saell wrote:> Hi there! > > We have kind of the same setup but are using a few number of SER boxes > for the on net calls - using enum for the lookup would be a great idea > so that you can get the numbers to do sip calls and move over slowly. > > And for the central routing voip server make the routing use SIP > redirects as the central server then can handle a lot of calls as its > only doing the routing decisions. > > Best regards > jan > > --On Wednesday, November 30, 2005 05:45:21 PM +0000 Joao Pereira > <joao.pereira@fccn.pt> wrote: > >> Hello >> Im managing a WAN with a lot of Universities. Some of them already >> installed a VoIP solution based on SER (to manage SIP clients) and >> Asterisk (for services and PSTN GW). The DNS routing provided by SER is >> working perfectly, but we want to start routing all calls thru IP >> transparently. >> We want our legacy PBXs (that are connected to Asterisk) to forward all >> calls to IP. The idea is to forward all calls to a central VoIP server, >> that has all the numbers that already are VoIP enabled, and then: >> - if the called number is VoIP enabled, he routes the call to that Univ. >> VoIP server >> - if the called number isnt in the list, the call goes back to the PBX >> and a PSTN call is dialed >> >> This way, ppl starts using the VoIP infrastructure, without even knowing >> what VoIP means, and the telecom bill starts decreasing. >> >> I know thats a statical and hierarchical structure and we dont want >> that, >> but is a good solution for this migration phase, where a lot of places >> are still using TDM systems. >> >> Now, the top of the hierarchy should be an Asterisk or SER? I dont know >> which of the systems is the best choice for the job. Does someone has an >> idea of what should we use? >> >> Thanks >> Joao Pereira >> www.fccn.pt >> >> >> >> >> _______________________________________________ >> --Bandwidth and Colocation provided by Easynews.com -- >> >> Asterisk-Users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > > > >------------------------------------------------------------------------ > >_______________________________________________ >Serusers mailing list >Serusers@iptel.org >http://mail.iptel.org/mailman/listinfo/serusers > >
Kind of depends on what you want to do! Remember Asterisk is not a SIP proxy so if you want to be able to call a phone from another SIP phone out in the world you probably best off with ser as a sip proxy and the asterisk as gateways, features servers. We do a lot of the routing and so with both ser and asterisk and sip redirects so that works. I see IAX more as a trunking protocol between the asterisk boxes so there is a place for both. Best regards jan --On 05 December 2005 23:53 +0000 Joao Pereira <joao.pereira@fccn.pt> wrote:> And about the protocol used to create this hierarchical network? > Should I use SIP (routing between SERs) or should I use IAX (routing > between Asterisks)? > > About ENUM, Isnt the managing of the ENUM tree going to be very > complicated and heavy when we reach the millions of users? > > Joao > > Jan Saell wrote: > >> Hi there! >> >> We have kind of the same setup but are using a few number of SER boxes >> for the on net calls - using enum for the lookup would be a great idea >> so that you can get the numbers to do sip calls and move over slowly. >> >> And for the central routing voip server make the routing use SIP >> redirects as the central server then can handle a lot of calls as its >> only doing the routing decisions. >> >> Best regards >> jan >> >> --On Wednesday, November 30, 2005 05:45:21 PM +0000 Joao Pereira >> <joao.pereira@fccn.pt> wrote: >> >>> Hello >>> Im managing a WAN with a lot of Universities. Some of them already >>> installed a VoIP solution based on SER (to manage SIP clients) and >>> Asterisk (for services and PSTN GW). The DNS routing provided by SER is >>> working perfectly, but we want to start routing all calls thru IP >>> transparently. >>> We want our legacy PBXs (that are connected to Asterisk) to forward all >>> calls to IP. The idea is to forward all calls to a central VoIP server, >>> that has all the numbers that already are VoIP enabled, and then: >>> - if the called number is VoIP enabled, he routes the call to that Univ. >>> VoIP server >>> - if the called number isnt in the list, the call goes back to the PBX >>> and a PSTN call is dialed >>> >>> This way, ppl starts using the VoIP infrastructure, without even knowing >>> what VoIP means, and the telecom bill starts decreasing. >>> >>> I know thats a statical and hierarchical structure and we dont want >>> that, >>> but is a good solution for this migration phase, where a lot of places >>> are still using TDM systems. >>> >>> Now, the top of the hierarchy should be an Asterisk or SER? I dont know >>> which of the systems is the best choice for the job. Does someone has an >>> idea of what should we use? >>> >>> Thanks >>> Joao Pereira >>> www.fccn.pt >>> >>> >>> >>> >>> _______________________________________________ >>> --Bandwidth and Colocation provided by Easynews.com -- >>> >>> Asterisk-Users mailing list >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Serusers mailing list >> Serusers@iptel.org >> http://mail.iptel.org/mailman/listinfo/serusers >> >> > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-- +------------------------------------------------------------------- ! Irial / YASK AB ! Att: Jan Saell ! Box 59, S-692 21 KUMLA, SWEDEN ! Tel: 019-58 25 15 Int +46-19 58 25 15 Fax +46-19 58 38 05 ! E-mail: jan@irial.com ! PGP Fingerprint: E957 23C8 9F51 0958 B9AD 7F18 404A 5DA1 F944 A08B -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 187 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20051206/8097a4a0/attachment.pgp