I'm currently looking to interconnect my Asterisk PBX system with the PSTN via a digital PRI/T1. I know a multitude of options exist for internal PCI cards (Digium/Sangoma/Rhino), I was wondering if anyone has any experience or recommendations of external PRI media gateways that support SIP. So far I've found: VegaStream Vega 400 Audiocodes Mediant 2000 MediaTrix 1531 However they are all expensive (over 3,000). Does any one have any other suggestions or experience with the above products? Thanks, Jameson -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070412/ab6ae8f4/attachment.htm
May i ask why not internal? On 4/12/07, jameson asterisk <mp3man.ast@gmail.com> wrote:> I'm currently looking to interconnect my Asterisk PBX system with the PSTN > via a digital PRI/T1. > I know a multitude of options exist for internal PCI cards > (Digium/Sangoma/Rhino), I was wondering if anyone has any experience or > recommendations of external PRI media gateways that support SIP. > > So far I've found: > VegaStream Vega 400 > Audiocodes Mediant 2000 > MediaTrix 1531 > > However they are all expensive (over 3,000). > > Does any one have any other suggestions or experience with the above > products? > > Thanks, > Jameson >
On Thu, Apr 12, 2007 at 11:59:00AM -0400, jameson asterisk wrote:> I'm currently looking to interconnect my Asterisk PBX system with the PSTN > via a digital PRI/T1. > I know a multitude of options exist for internal PCI cards > (Digium/Sangoma/Rhino), I was wondering if anyone has any experience or > recommendations of external PRI media gateways that support SIP. > > So far I've found: > VegaStream Vega 400 > Audiocodes Mediant 2000 > MediaTrix 1531Also have a look at Patton SmartNode 4960 range. They are available in various configurations/numbers of channels, some of which are upgradable to more channels at a later date: http://www.patton.com/products/pe_printable.asp?category=354 We have the ISDN2 and Analogue versions of these gateways (same software) and so far they have been very reliable, and can be configured in a variety of fail-over situations in case asterisk or the connection to the server dies, incoming calls can be automatically routed either back out on another ISDN channel or out to another SIP/analogue gateway etc. Rob -- Robert Lister - London Internet Exchange - http://www.linx.net/ robl@linx.net - tel: +44 (0)20 7645 3510 - RL786-RIPE