Why would you use gateways and PRI's when several of the major carriers (AT&T, Global Crossing, etc.) also have products that can interface directly with SIP for the same per minute cost? We have a multisite Asterisk call center application and are routing all calls over private VPN to one central Asterisk location from where we have multiple point-to-point T1's going straight into Global Crossing. They are accepting the traffic as SIP g.729a and are handling the gateway themselves. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Callum McGillivray Sent: Friday, April 29, 2005 1:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] T1 Technology and VoIP Gateway Primer Hi Matt & everyone else, We have also been steering toward using a gateway for our large installation. Ours differs from your significantly in as much as our setup will involve 8 apartment buildings located throughout the CBD. Each apartment building will have as many as 600 extensions (rooms) with an Asterisk Server in the comms room in the basement. Incoming and Outgoing calls are going to be trunked from the Asterisk box along a fiber link back to our core exchange, where the calls will be handed off to a gateway machine (Cisco?) which will have an impressively large number of PRI's plugged into the back of it. My (very vague) examination so far tells me that I can use something along the lines of a Cisco AS5400 (a couple of which I have kicking around here in the office). Has anyone had experience in handing off / receiving calls from a Cisco AS5400 with Asterisk ? How is it done ? Matt, is this similar to the idea that you have for your project ? What Cisco hardware have you looked at so far ? How many E1/T1 lines are you going to have terminating on your setup ? Cheers, Callum Matt Roth wrote:> Michael, > >> Have you decided which PSTN-VoIP gateway you'll use? > > > > Not yet, but our preference is a Cisco gateway. Lucent, Quintum, and > AudioCodes also make TDM-VoIP gateways. > > Prior to purchasing any hardware, our entire layout will be posted to > this list in detail for review. > > Matthew Roth > http://voip-info.org/tiki-index.php?page=Running%20Asterisk%20on%20Deb > ian _______________________________________________ > 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_______________________________________________ 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 The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any.
Callum McGillivray
2005-Apr-30 23:31 UTC
[Asterisk-Users] T1 Technology and VoIP Gateway Primer
Hi Adam, Unfortunatley we are located in Australia and our chosen provider does not provide this service. In the future as our client bae grows larger, we may need to look at implmenting other carriers that provide this kind of service, but in the meantime we will be using PRI's. Cheers, Callum Adam Robins wrote:>Why would you use gateways and PRI's when several of the major carriers >(AT&T, Global Crossing, etc.) also have products that can interface >directly with SIP for the same per minute cost? > >We have a multisite Asterisk call center application and are routing all >calls over private VPN to one central Asterisk location from where we >have multiple point-to-point T1's going straight into Global Crossing. >They are accepting the traffic as SIP g.729a and are handling the >gateway themselves. > >-----Original Message----- >From: asterisk-users-bounces@lists.digium.com >[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Callum >McGillivray >Sent: Friday, April 29, 2005 1:19 AM >To: Asterisk Users Mailing List - Non-Commercial Discussion >Subject: Re: [Asterisk-Users] T1 Technology and VoIP Gateway Primer > >Hi Matt & everyone else, > >We have also been steering toward using a gateway for our large >installation. > >Ours differs from your significantly in as much as our setup will >involve 8 apartment buildings located throughout the CBD. Each >apartment building will have as many as 600 extensions (rooms) with an >Asterisk Server in the comms room in the basement. > >Incoming and Outgoing calls are going to be trunked from the Asterisk >box along a fiber link back to our core exchange, where the calls will >be handed off to a gateway machine (Cisco?) which will have an >impressively large number of PRI's plugged into the back of it. > >My (very vague) examination so far tells me that I can use something >along the lines of a Cisco AS5400 (a couple of which I have kicking >around here in the office). > >Has anyone had experience in handing off / receiving calls from a Cisco >AS5400 with Asterisk ? > >How is it done ? > >Matt, is this similar to the idea that you have for your project ? What >Cisco hardware have you looked at so far ? How many E1/T1 lines are you >going to have terminating on your setup ? > >Cheers, > >Callum > >Matt Roth wrote: > > > >>Michael, >> >> >> >>>Have you decided which PSTN-VoIP gateway you'll use? >>> >>> >> >>Not yet, but our preference is a Cisco gateway. Lucent, Quintum, and >>AudioCodes also make TDM-VoIP gateways. >> >>Prior to purchasing any hardware, our entire layout will be posted to >>this list in detail for review. >> >>Matthew Roth >>http://voip-info.org/tiki-index.php?page=Running%20Asterisk%20on%20Deb >>ian _______________________________________________ >>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 >> >> > > >_______________________________________________ >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 > >The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. > > >_______________________________________________ >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/20050501/ef9df80a/attachment.htm