We are currently moving away from a wide-spread Cisco CallManager deployment to Asterisk. For many of our small sites we have the routers configured for what Cisco calls SRST so if we have a WAN failure, the router acts as a SCCP registrar. We are converting to SIP, and from what I can tell Cisco wants a license for each router to run SRST over SIP... So my question to the group is: What are you doing for survivability in these small (6-30 phone) sites? I would like to avoid deploying a lot of servers if at all possible. The requirements would be a simple, easy to manage device for the phones to register to in case of WAN failure with 1 or 2 POTS lines attached (also used for 911 calls from that site). Thanks for any suggestions! -Jonathan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090706/7917d7bc/attachment.htm
Audiocodes supports SRST on their mediapack analog gateways. Cory J. Andrews Director New Market Initiatives Sayers Media Group VoIP Supply, LLC 454 Sonwil Drive Buffalo, NY 14225 716-250-3402 OFFICE 716-630-1548 FAX 716-601-4474 MOBILE candrews at sayersmedia.com <mailto:brett at voipsupply.com> Have I exceeded your expectations? Please share your experience with my boss, Benjamin P. Sayers <mailto:bsayers at voipsupply.com> , CEO NOTICE: The information contained in this email and any document attached hereto is intended only for the named recipient(s). It is the property of the VoIP Supply, LLC and shall not be used, disclosed or reproduced without the express written consent of VoIP Supply, LLC. If you are not the intended recipient, nor the employee or agent responsible for delivering this message in confidence to the intended recipient(s), you are hereby notified that you have received this transmittal in error, and any review, dissemination, distribution or copying of this transmittal or its attachments is strictly prohibited. If you have received this transmittal and/or attachments in error, please notify me immediately by reply e-mail or telephone and then delete this message, including any attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY 14225 USA. From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Jonathan Thurman Sent: Monday, July 06, 2009 11:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Small site survivability We are currently moving away from a wide-spread Cisco CallManager deployment to Asterisk. For many of our small sites we have the routers configured for what Cisco calls SRST so if we have a WAN failure, the router acts as a SCCP registrar. We are converting to SIP, and from what I can tell Cisco wants a license for each router to run SRST over SIP... So my question to the group is: What are you doing for survivability in these small (6-30 phone) sites? I would like to avoid deploying a lot of servers if at all possible. The requirements would be a simple, easy to manage device for the phones to register to in case of WAN failure with 1 or 2 POTS lines attached (also used for 911 calls from that site). Thanks for any suggestions! -Jonathan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090706/57c9cf3e/attachment.htm
On Mon, 6 Jul 2009, Jonathan Thurman wrote:> We are currently moving away from a wide-spread Cisco CallManager deployment > to Asterisk. For many of our small sites we have the routers configured for > what Cisco calls SRST so if we have a WAN failure, the router acts as a SCCP > registrar. We are converting to SIP, and from what I can tell Cisco wants a > license for each router to run SRST over SIP... > > So my question to the group is: What are you doing for survivability in > these small (6-30 phone) sites? I would like to avoid deploying a lot of > servers if at all possible. The requirements would be a simple, easy to > manage device for the phones to register to in case of WAN failure with 1 or > 2 POTS lines attached (also used for 911 calls from that site). Thanks for > any suggestions!Deploy a lot of small asterisk based appliances... This way you can completely decentralise your setup and give each office it's own autonomous system, only needing the WAN links for inter-site calls (and maybe your backhaul to the PSTN) 30 phones will trivially work from a diskless, fanless, processor, so no need for anything too clever. If building them yourselves the cost per site ought to be under $600 for the hardware (Under ?400 where I am, so apply conversion rate) you could use one of the pre-built packages for this - pbxinaflash, or buy something like an Atcom unit, etc. Gordon
2009/7/6 Jonathan Thurman <jthurman42 at gmail.com>> We are currently moving away from a wide-spread Cisco CallManager > deployment to Asterisk. For many of our small sites we have the routers > configured for what Cisco calls SRST so if we have a WAN failure, the router > acts as a SCCP registrar. We are converting to SIP, and from what I can > tell Cisco wants a license for each router to run SRST over SIP... > > So my question to the group is: What are you doing for survivability in > these small (6-30 phone) sites? I would like to avoid deploying a lot of > servers if at all possible. The requirements would be a simple, easy to > manage device for the phones to register to in case of WAN failure with 1 or > 2 POTS lines attached (also used for 911 calls from that site).What happens for IT when WAN fails ? Are people still able to work or not ? If they are, then it should be possible to use current routers (if they have such POTS interfaces) as Media gateways and have a local resource to act as a backup Asterisk server. If they are not, having IT and Telephony to share the same backup WAN is advisable.> Thanks for any suggestions! > > -Jonathan > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > 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/20090708/7e98ee85/attachment.htm