Doug, We currently are using Digium TE410P boards directly into each Asterisk server. I've been researching various gateways, up to DS3 capacity, to convert PRI to SIP and then allocate the SIP among multiple Asterisk servers. I've looked at Cisco AS5400 ($$$$), Lucent APX 1000 ($$$), and Quintum Tenor CMS ($$).
Thanks, Adam -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: Friday, December 09, 2005 10:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk Dial Failover Adam, An Audicodes Mediant 2000 gateway with a couple of PRI's. Why? Doug. -----Original Message----- From: Adam Robins [mailto:[EMAIL PROTECTED] Sent: Friday, December 09, 2005 7:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk Dial Failover What are you using to terminate the PSTN calls and do the SIP transcoding? -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan k. Creasy Sent: Friday, December 09, 2005 8:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk Dial Failover I chose this method and have been happy with the results. -Jonathan -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Friday, December 09, 2005 7:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk Dial Failover Your other option is to setup the OpenSER boxes in a truly redundant configuration using Linux HA (www.linux-ha.org). That way you setup all your PSTN calls to forward to one shared virtual IP between the boxes. One of the boxes is the Master, the other is the Slave. There is a heartbeat between the boxes that goes at a configurable rate. If the Master fails then the Slave will take over and it can even be configured for sub-second failover. I think there is a article on voip-info.org about this, but don't have time to look it up. Good luck and let us know what you choose to do. Ryan > All, > > I have an Asterisk system that sends PSTN calls to an OpenSER system to be > routed. I have a command like this in my extensions.conf: > > exten => 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr) > > There's actually two OpenSER systems for redundancy. I'm trying to find a > way to have Asterisk attempt to route the call to one OpenSER system, and > if it's down, fallback to another. > > Any first thoughts on how to achieve this? > > I can't have Asterisk do a DNS SRV lookup because Asterisks SRV lookups > are broken. If I issue a series of Dial commands, such as this: > > exten => 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr) > exten => 1_.,2,Dial(SIP/[EMAIL PROTECTED],20,tr) > > ... what seems to happen is that when proxy1 is down, Asterisk waits the > full 20 seconds before returning control. Also, This 20s includes the time > is takes for the other end to answer, so if I put a small value of say 5s > in there, the dial command will probably give up before someone answers at > the other end. Neither is workable. > > Asterisk SHOULD be able to distinguish between a TRYING and no response. > In the event it gets no TRYING response to a dial command within a > specified timeout it should return control and flag an error. If on the > other hand it does get a TRYING response (and maybe a RINGING too) it > should continue to wait until the 20s has expired. > > I can't use dynamic DNS (ie putting two A records for a hostname in DNS) > because Asterisk reads the extensions.conf on startup and also seems to > cache what the host maps to on startup. Subsequent calls to the host > always return the same IP address. > > But... in general... how have people implemented this? > > Help appreciated! > Doug > > > > > > > > _______________________________________________ > --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 > _______________________________________________ --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 _______________________________________________ --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 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. _______________________________________________ --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 _______________________________________________ --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 _______________________________________________ --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