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. _______________________________________________ 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/
