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
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: miércoles, 30 de noviembre de 2005 18:45 Para: serusers@lists.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
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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
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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
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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
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Gustavo García Bernardo -
Jan Saell -
Joao Pereira