It depends on what you are trying to accomplish. For budgeting purposes it is the same cost and same device, with a slightly different configuration.
On Mon, Mar 15, 2010 at 5:59 PM, Andrew Cotter <[email protected]> wrote: > In this config are you talking about the Ingate SIParator or Ingate > Firewall. Sounds like what I want to do! > > Andrew > >> -----Original Message----- >> From: Tony Graziano [mailto:[email protected]] >> Sent: Monday, March 15, 2010 5:47 PM >> To: Picher, Michael >> Cc: Andrew Cotter; Todd Hodgen; Sipx-users list >> Subject: Re: [sipx-users] One last attempt - AT&T IP Flex >> >> It's nice not running trunking on the server and using an >> independent device. You can make sipx dialplan changes and >> restart services without interfering with a call in progress. >> >> With our typical ingate configuration, ETH1 gets a public IP >> address, ETH0 gets an ip on the same subnet as the sipx >> system. Assuming your phones are on a separate vlan, it works >> very well. >> >> Since the ingate can accept both trunking and remote users on >> port 5060, you don't have any extra work in configuring DNS >> or an out of the ordinary proxy configuration. >> >> Remote users with FTP to their "home" server for voice are >> also supportable using polycom phones, though safefguards >> should be applied. We do this all the time and have very good results. >> >> We also would be able to configure it to support multi-site >> dialing plans (one system to another) without adding the >> siptrunking role to sipx. Makes it very tidy. >> >> Ingates are small, and have flash drives for reliability, low >> power consumption and so forth. >> >> On Mon, Mar 15, 2010 at 5:08 PM, Picher, Michael >> <[email protected]> wrote: >> > I'm with everyone else and recommending Ingates at this point for >> > flexibility / reliability. >> > >> > Mike >> > >> >> -----Original Message----- >> >> From: [email protected] [mailto:sipx-users- >> >> [email protected]] On Behalf Of Andrew Cotter >> >> Sent: Monday, March 15, 2010 4:02 PM >> >> To: 'Todd Hodgen'; 'Sipx-users list' >> >> Subject: Re: [sipx-users] One last attempt - AT&T IP Flex >> >> >> >> 15 in FL are not that big of a deal. Only 4-5 maybe use >> the phones >> >> significantly. The site is a warehouse mostly so there are people >> >> receiving equipment, auditing, testing, packing, and >> shipping. Most >> >> calls by >> > the >> >> warehouse staff are internal. The rest are in sales and a >> moderate >> >> phone users. >> >> >> >> They are typically in the mid 50-60 ms ping time. >> >> >> >> I don't know of any other AT&T trunk offering. Been using >> flowroute >> > in >> >> AZ >> >> for a while on asterisk which has worked really well. >> >> >> >> Andrew >> >> >> >> > -----Original Message----- >> >> > From: Todd Hodgen [mailto:[email protected]] >> >> > Sent: Monday, March 15, 2010 3:51 PM >> >> > To: 'Andrew Cotter'; 'Sipx-users list' >> >> > Subject: RE: [sipx-users] One last attempt - AT&T IP Flex >> >> > >> >> > You may be able to get out of those contracts if they >> can't provide >> >> > the provisioning that you need, and it really is quite >> >> simple. >> >> > >> >> > Sounds like you just need SIP trunks really. There are >> several on >> >> > this list that provide SIP trunks from other providers that have >> >> > been certified to work with sipXecs, which would make >> life simpler >> >> > for you, and potentially save you the cost of additional >> hardware. >> >> > >> >> > If your traffic is staying on net with AT&T, I would >> think trying >> >> > some sipx to sipx calls between two locations might be a >> good judge >> >> > of the type of service you will get across those links. >> You could >> >> > run Ping Plotter between two locations as well to see how much >> >> > delay runs between them for a good understanding of the >> underlying >> >> > network. >> >> > >> >> > BTW, the 15 users in Florida would be a concern for me over a >> >> > single T-1 unless you are running some compression on >> those calls, >> >> > assuming you run general internet traffic over that circuit also. >> >> > >> >> > Does AT&T offer other SIP trunks that are not part of their IP >> >> > Perplex, maybe IP non-Flex that is simpler and more configurable? >> >> > >> >> > >> >> > >> >> > -----Original Message----- >> >> > From: Andrew Cotter [mailto:[email protected]] >> >> > Sent: Monday, March 15, 2010 12:31 PM >> >> > To: 'Todd Hodgen'; 'Sipx-users list' >> >> > Subject: RE: [sipx-users] One last attempt - AT&T IP Flex >> >> > >> >> > "BTW, IP FLEX doesn't seem to have much FLEX." - That made me >> >> > chuckle! >> >> > >> >> > Why AT&T? They are providing our internet at all 4 sites. >> >> > We have a dozen or so home office types as well, but I am not >> >> > concerned with them as of yet. >> >> > We are in contract with AT&T, but I have already spoken with the >> >> > sales rep that I may want to drop IP Flex at the two smaller >> >> > locations where it has not been installed yet. >> >> > Fiber at HQ with 60 users >> >> > T1 in FL - 15 users >> >> > T1 in AZ - 5 users >> >> > T1 in IL - 3 users >> >> > >> >> > No MPLS between sites, but IP Flex is supposed to allow >> for on-net >> >> > calling between sites. This lets AT&T handle the QoS >> without the >> >> > cost to us for MPLS. Not much site-to-site calling is going on, >> >> > but some is. >> >> > >> >> > HQ is the only site I have tried SipX with and it is the most >> >> > complex by far. Our datacenter is also at HQ. Network is ok >> >> > internally and calls route as expected. Separate VLAN for our >> >> > network internally for the phones, Cisco SIP handoff, Audiocodes >> >> > MP118, and SipX. >> >> > >> >> > Would people suggest not getting IP Flex at the smaller >> locations >> >> > and run SIP over IPSEC VPN tunnels between CT and AZ/IL? >> Not much >> >> > QoS on the public internet, but AT&T circuits on both ends so I >> >> > might have a better shot with this. >> >> > >> >> > I can go into more detail if it would help. >> >> > >> >> > Andrew >> >> > >> >> > >> >> > > -----Original Message----- >> >> > > From: Todd Hodgen [mailto:[email protected]] >> >> > > Sent: Monday, March 15, 2010 3:13 PM >> >> > > To: 'Andrew Cotter'; 'Sipx-users list' >> >> > > Subject: RE: [sipx-users] One last attempt - AT&T IP Flex >> >> > > >> >> > > If you could explain your network in more detail, there may >> >> > be several >> >> > > solutions. >> >> > > >> >> > > For instance, Is AT&T providing an MPLS network to connect >> >> > these sites >> >> > > together? Could you use site to site dialing, and then use a >> >> > > different provider for the SIP trunks over the MPLS network? >> >> > > >> >> > > IS there a contractual reason why you have to use AT&T, or is >> >> > > that just a preference you have. There are many other >> providers >> >> > that can >> >> > > support standard sip. >> >> > > >> >> > > BTW, IP FLEX doesn't seem to have much FLEX. >> >> > > >> >> > > -----Original Message----- >> >> > > From: [email protected] >> >> > > [mailto:[email protected]] On Behalf Of >> > Andrew >> >> > > Cotter >> >> > > Sent: Monday, March 15, 2010 12:06 PM >> >> > > To: 'Sipx-users list' >> >> > > Subject: [sipx-users] One last attempt - AT&T IP Flex >> >> > > >> >> > > So I have finally gotten word from AT&T labs that they >> will not >> >> > > be able to support SipX and fix our transfer issue. >> >> > > >> >> > > We have a SIP handoff that is direct (switch in the middle) >> >> > from their >> >> > > Cisco router onsite. I asked for them to send signaling on >> >> > port 5080 >> >> > > (sipXbridge) but that was a no go. >> >> > > Then I asked if they can do some sort of NAT translation >> >> > for incoming >> >> > > data from their end, through the router, and into port >> >> > 5080. Again, >> >> > > no go as they tested this in the labs. >> >> > > B2BUA on the Cisco, nope. >> >> > > >> >> > > So... I am left with probably having to leave my sipX >> setup, that >> > I >> >> > > have come to know and love, behind. >> >> > > >> >> > > A final question for the masses: >> >> > > >> >> > > Would having AT&T swap out the SIP handoff for a PRI handoff >> >> > > potentially fix my transfer issues if I put a gateway in? If >> >> > > this would work and I can convince AT&T to convert the SIP >> >> > handoff to a PRI >> >> > > handoff, what solution would you suggest (patton, >> >> > audiocodes, etc.) to >> >> > > handle a single PRI. I have >> >> > > 4 sites spread throughout the US and would need something >> >> > fairly cost >> >> > > effective for 2 of them since there are 5 or less employees >> >> > at those >> >> > > sites. >> >> > > I am sure I will have more questions if people come back >> >> > saying this >> >> > > might resolve the issues. >> >> > > >> >> > > >> >> > > Parting thoughts. >> >> > > In light of the position I am now in I am forced to >> begin to look >> >> > > elsewhere at commercial products. I wanted to share >> my thoughts >> > on >> >> > > the comparison of sipx and a well known commercial product >> >> > out there. >> >> > > After getting a demo of one solution that the salesperson >> >> > was touting >> >> > > as an extremely easy interface, so simple a cave man can >> >> > set it up, I >> >> > > was amazed at how much I was left desiring the >> simplicity of SipX. >> >> > > The screens were cluttered, the interface was fairly well >> >> > organized, >> >> > > but the voicemail and admin console still resided on a windows >> >> > > machine. Not what I want. >> >> > > >> >> > > Yes it was a nice system in terms of failover and distribution, >> > but >> >> > > they pretty much insist that we swap out our phones >> (polycom) for >> >> > > their own phones. Also, for a VoIP system they almost left me >> >> > > speechless when they said I could only use one SIP >> trunk provider >> >> > > unless I bought an InGate. >> >> > > VoIP... SIP... Won't support it? Wow! Don't even get me >> >> > started on >> >> > > the Windows application or the Outlook piece that I repeatedly >> > told >> >> > > them we would not be using. >> >> > > >> >> > > Thank you again for everyone's help and suggestions >> over the past >> >> > > month in trying to make this work. If I can slip in a plug for >> > the >> >> > > project during my talk at the Computerworld OSBC later >> this week >> >> > > I will. >> >> > > >> >> > > >> >> > > Andrew >> >> > > >> >> > > _______________________________________________ >> >> > > sipx-users mailing list [email protected] List >> >> > > Archive: http://list.sipfoundry.org/archive/sipx-users >> >> > > Unsubscribe: >> > http://list.sipfoundry.org/mailman/listinfo/sipx-users >> >> > > sipXecs IP PBX -- http://www.sipfoundry.org/ >> >> > > >> >> > >> >> >> >> _______________________________________________ >> >> sipx-users mailing list [email protected] >> List Archive: >> >> http://list.sipfoundry.org/archive/sipx-users >> >> Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users >> >> sipXecs IP PBX -- http://www.sipfoundry.org/ >> > _______________________________________________ >> > sipx-users mailing list [email protected] List >> Archive: >> > http://list.sipfoundry.org/archive/sipx-users >> > Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users >> > sipXecs IP PBX -- http://www.sipfoundry.org/ >> > >> >> >> >> -- >> ====================== >> Tony Graziano, Manager >> Telephone: 434.984.8430 >> Fax: 434.984.8431 >> >> Email: [email protected] >> >> LAN/Telephony/Security and Control Systems Helpdesk: >> Telephone: 434.984.8426 >> Fax: 434.984.8427 >> >> Helpdesk Contract Customers: >> http://www.myitdepartment.net/gethelp/ >> >> Why do mathematicians always confuse Halloween and Christmas? >> Because 31 Oct = 25 Dec. >> > > -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: [email protected] LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ Why do mathematicians always confuse Halloween and Christmas? Because 31 Oct = 25 Dec. _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users sipXecs IP PBX -- http://www.sipfoundry.org/
