30 Nov
2005
30 Nov
'05
6:45 p.m.
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
30 Nov
30 Nov
7:10 p.m.
New subject: [Serusers] RE: [Asterisk-Users] hierarchical VoIP system
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(a)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(a)lists.iptel.org; asterisk-users(a)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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11:32 p.m.
New subject: [Serusers] Re: [Asterisk-Users] hierarchical VoIP system
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(a)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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To UNSUBSCRIBE or update options visit:
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--
+-------------------------------------------------------------------
! 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(a)irial.com
! PGP Fingerprint: E957 23C8 9F51 0958 B9AD 7F18 404A 5DA1 F944 A08B
6 Dec
6 Dec
12:53 a.m.
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(a)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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Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
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------------------------------------------------------------------------
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Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
8:42 p.m.
New subject: [Serusers] Re: [Asterisk-Users] hierarchical VoIP system
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(a)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:
--
+-------------------------------------------------------------------
! 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(a)irial.com
! PGP Fingerprint: E957 23C8 9F51 0958 B9AD 7F18 404A 5DA1 F944 A08B
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(a)fccn.pt> wrote:
_______________________________________________
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http://lists.digium.com/mailman/listinfo/asterisk-users 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:
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4 comments
3 participants
participants (3)
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Gustavo García Bernardo -
Jan Saell -
Joao Pereira