yeah? wow. I'd still like to see that. On Fri, Sep 30, 2011 at 12:34 PM, Max DiOrio <[email protected]> wrote: > I'm talking about their actually paid support technicians. > > > -----Original Message----- > From: [email protected] > [mailto:[email protected]] On Behalf Of Tony Graziano > Sent: Friday, September 30, 2011 12:28 PM > To: Discussion list for users of sipXecs software > Subject: Re: [sipx-users] pfSense QoS > > Send me a link to that post on the pfsense board if you would. > > On Fri, Sep 30, 2011 at 12:24 PM, Max DiOrio <[email protected]> wrote: >> That was the setup that pfSense support recommended. I'm surprised that >> being the firewall experts, they didn't recommend a different configuration >> like the one that Tony offered me. Our office is quite slow today and >> Monday, so Tuesday would be the first good test of call quality with higher >> volume. >> >> >> >> >> -----Original Message----- >> From: [email protected] >> [mailto:[email protected]] On Behalf Of Tony >> Graziano >> Sent: Friday, September 30, 2011 12:10 PM >> To: Discussion list for users of sipXecs software >> Subject: Re: [sipx-users] pfSense QoS >> >> He was prioritizing an IP address in his firewall. The problem with that is >> he was not prioritizing the ports only, so when he hits the webgui for the >> firewall, it cannot discern that traffic from voice traffic. >> >> On Fri, Sep 30, 2011 at 12:07 PM, <[email protected]> wrote: >>> Anyhow, increase the bandwidth for Voip, I think you have no other >>> important stuff over this line, so give it 100% (or just 80%). >>> The question is what is going on so that 1 call can be affected and 4 >>> can be OK, it must be some non-voip traffic that is eating up the bandwidth. >>> Question is also what bandwidth, internally or internet. It would be >>> nice to see whether your captures show packet loss. >>> >>> Paul >>> >>> Max DiOrio <[email protected]> wrote on 30-09-2011 17:18:57: >>> >>> >>>> Sorry, my day had already ended here at work by the time I got >>>> around to seeing these posts. >>>> >>>> I am using the pfSense admin interface remotely. The interfaces are >>>> set to default for both, which is detected as 100baseTX full-duplex. >>>> >>>> The system is a Pentium D 3.4GHz with 1GB of RAM, only 20% used. >>>> The system is oversized. >>>> >>>> I can't connect a phone to the modem since it's in bridged mode. >>>> Phones are connected to my DLink PoE switch and straight into >>>> pfsense. Our office is almost always quiet. Only one or two calls >>>> going on at any given moment. We only have 10 phones. It's totally >>>> random when the call quality is affected. Could be on a single >>>> call, but it could be fine when 4 calls are going on at the same time. >>>> >>>> I have packet captures taken all day yesterday on the pfSense box on >>>> both the LAN and the WAN side of the box. I need to copy them over >>>> to my PC to look. I have the caps from two calls in which I know >>>> the voice quality was affected. >>>> >>>> [image removed] >>>> >>>> From: [email protected] [mailto:sipx-users- >>>> [email protected]] On Behalf Of [email protected] >>>> Sent: Friday, September 30, 2011 3:07 AM >>>> To: Discussion list for users of sipXecs software >>>> Subject: Re: [sipx-users] pfSense QoS >>>> >>>> I think Max has a local problem. Especially if he is using the >>>> pfsense gui from the local network (never saw an answer to that >>>> question if I am not mistaken). >>>> Either the pfsense box has a duplex mismatch, is heavily under >>>> dimensioned or something else is wrong. >>>> >>>> I would connect a phone to the ADSL modem (so as close to the >>>> internet, separated from the rest of the network through the pfsense >>>> box, but not the internet) and do some tests from the user phones. >>>> If you don't have traffic monitoring it would be good to look at >>>> port utilization and queueing in the switch(es). >>>> QoS is certainly nice and needed, but it only kicks in when an >>>> outgoing port queues anything, so you need to be generating quite >>>> some traffic before it kicks in. >>>> (At home I have a 2Mbps/256kbps ADSL (because of distance issues) >>>> and voip-voice is A-OK without QoS, but when daughter-dear starts a >>>> Youtube it's byebye decent audio (only for me, not for the other >>>> side, so the 2Mbps is the problem, not the 256 Kbps, this also shows >>>> how much effort ITSP's put in in providing decent services >>>> (ISP=ITSP))))))))). >>>> >>>> BTW: If you have any managed switches in the flow where you did not >>>> configure QoS then the QoS is normally back to 0, normally a managed >>>> switch by default does not trust QoS settings in packets and resets >>>> it to nada. Cisco style: mls qos trust dscp and mls qos trust cos >>>> >>>> BTW2: Have you placed test calls when the network was relatively >>>> quiet (out-of-office hours)? >>>> >>>> Paul >>>> >>>> Tony Graziano <[email protected]> wrote on 30-09-2011 02:18:46: >>>> >>>> > I would do a traceroute to somewhere like googledns, 8.8.8.8, then >>>> > look at the first few hops and see if they are also on the >>>> > providers network and whst your latency to those points are and >>>> > also to the itsp gateway/proxy you use. >>>> > >>>> > I just went through a dozen sites for a customer to make >>>> > recommendations. After doing the basic reports, and wasting a lot >>>> > of time with teeny providers saying they were fine, we started >>>> > replacing the connection with something better than 2.5mb >>>> > down/512k up no name dsl providers. We saw instant results and >>>> > none of these sites are running voice services over the internet. >>>> > >>>> > Always look at the entire network and if your internet is crappy, >>>> > set expectations accordingly and have it fixed or replaced. >>>> > Whether it is used just to get to the provider pop and not the >>>> > internet doesnt matter. What matters is whether it is a quality >>>> > connection. >>>> > If you dont have the quality, there a whole lot you just cant pull of. >>>> >>>> > On Thu, Sep 29, 2011 at 2:56 PM, Max DiOrio >>>> > <[email protected]> >>>> > wrote: >>>> > Given my ISP is my ITSP, shouldn' tmy delay be lower than 16ms avg? >>>> > >>>> > >>>> > From: [email protected] [mailto:sipx-users- >>>> > [email protected]] On Behalf Of Tony Graziano >>>> > Sent: Thursday, September 29, 2011 2:52 PM >>>> > >>>> > To: Discussion list for users of sipXecs software >>>> > Subject: Re: [sipx-users] pfSense QoS >>>> > >>>> > Kick yerself in the pants or not. Make sure your QUALITY is good. >>>> > Look at your quality graphs and make sure you don't have latency. >>>> > If it was easy, everybody would be doing it. >>>> > On Thu, Sep 29, 2011 at 2:50 PM, Max DiOrio >>>> > <[email protected]> >>>> > wrote: >>>> > The thing that is kicking me in the pants is, should all this be >>>> > necessary for 10 phones on a dedicated VoIP only 9 x 1.5 >>>> > connection? The only data traffic on this ISP is administration >>>> > websites. >>>> > >>>> > >>>> > [image removed] >>>> > >>>> > From: [email protected] [mailto:sipx-users- >>>> > [email protected]] On Behalf Of Tony Graziano >>>> > Sent: Thursday, September 29, 2011 2:42 PM >>>> > >>>> > To: Discussion list for users of sipXecs software >>>> > Subject: Re: [sipx-users] pfSense QoS >>>> > >>>> > Let's not confuse qos with shaping. In the meantime I've emailed >>>> > you my beta version of the pfsense shaper wizard for sipxecs using >>>> > pfsense 2.0. I'd backup your config, and the files you are >>>> > replacing in the event you want to revert it. I'm running it without >>>> > issues. >>>> > >>>> > If the ITSP is not your ISP the qos is probably lost on them. If >>>> > not, it's a "switch" function which means a procurve/cisco/juniper >>>> > or something a little more smartie pants'ish is in your near future. >>>> > >>>> > Re-run the shaper with the same bandwdith and type and see if it >>>> > properly shapes bot call directions. This one is particular to >>>> > your provider, as it markes UDP 49152-53000 and 30000-31000 >>>> > instead of only 10000-20000 and 30000-31000 udp. >>>> > >>>> > It should ignore the IP and also use just 5060/tcp/udp and 5080udp >>>> > along with the ports above to queue the bandwidth according to the >>>> > best available. >>>> > On Thu, Sep 29, 2011 at 2:33 PM, Josh Patten <[email protected]> wrote: >>>> > Life would be much easier if you set your QoS precedence >>>> > (priority) to 5. Read on for why >>>> > >>>> > 5 in binary is 101 >>>> > 6 in binary is 110 >>>> > 7 in binary is 111 >>>> > DSCP 46, which is what all RTP traffic normally uses and what most >>>> > routers/switches/gateways/phones are set to use by default to tag/ >>>> > prioritize RTP traffic, in binary is 101 110 (notice how I put a >>>> space there) >>>> > >>>> > The first three bits are the precedence field which can be 0-7 >>>> > (000 to 111). Only 0-5 should be used for user traffic. 6 and 7 >>>> > are reserved for internetwork control and network control, respectively. >>>> > The remaining 3 bits are, in order, delay (0=normal, 1=low), >>>> > throughput (0=normal, 1=high), and reliability (0=normal, 1=high) >>>> > - >>>> > -So for DSCP 46 you have a precedence value of 5 (101), low delay >>>> > (1), high throughput (1), and normal reliability (0). Put all this >>>> > together and you have 101110 binary which is equal to 46 decimal. >>>> > >>>> > Also, if you are using sipXbridge then the VoIP traffic that is >>>> > reaching it is not being tagged. This needs to be done at the >>>> > switch level it possible. >>>> > >>>> > Try getting all your voice stuff back to precedence level 5 and >>>> > setting your QoS stuff for VoIP to precedence level 5. >>>> > >>>> > On Thu, Sep 29, 2011 at 1:09 PM, Max DiOrio >>>> > <[email protected]> >>>> > wrote: >>>> > pfSense is set with floating rules, defining any UDP traffic >>>> > destined for my sip trunks gets put into the qVoIP. Also any UDP >>>> > traffic from my VoIP server to any destination gets put into the qVoIP. >>>> > >>>> > Wouldn't this be enough to pull all RTP packets into the qVoIP >>>> > priority queue and therefore be set to a priority of 7 in this >>>> > case. and >>>> > >>>> > >>>> > I'm wondering if we're running into problems with the way the >>>> > bandwidth and SC is configured on the qVoIP queues. Shouldn't the >>>> > bandwidth be set to 86Kbits/s and an RealTime SC m2 of 860Kb? >>>> > >>>> > WAN HFSC Bandwidth 1572 Kbit/s >>>> > WAN qACK Priority 6, Bandwidth 19.602%, Link Share SC m2 of >>>> > 19.602% WAN qDefault has priority 3, Bandwidth 9.801% WAN qVoIP >>>> > priority 7, bandwidth of 32Kbits/s, with Real time SC and an m2 of 240Kb. >>>> > >>>> > LAN HFSC >>>> > LAN qLink priority 2, queue limit 500, Bandwidth 20% LAN >>>> > qInternet, Priority 1, Bandwidth 9Mbit/s, Upperlimit SC m2 of 9Mb, >>>> > LinkShare SC m2 of 9Mb LAN qInternet qACK priority 6, Bandwidth >>>> > 19.48%, Linkshare SC m2 of 19.48% LAN qInternet qVOIP priority 7, >>>> > bandwidth of 32Kbits/s, with Real time SC and an m2 of 240Kb >>>> > >>>> > >>>> > [image removed] >>>> > >>>> > From: [email protected] [mailto:sipx-users- >>>> > [email protected]] On Behalf Of Max DiOrio >>>> > Sent: Thursday, September 29, 2011 1:35 PM >>>> > >>>> > To: Discussion list for users of sipXecs software >>>> > Subject: Re: [sipx-users] pfSense QoS >>>> > >>>> > Using Polycom 550's with their default QoS settings. >>>> > >>>> > We are using sipXbridge for our ITSP connection. >>>> > >>>> > How do you recommend I set up QoS for sipXbridge? >>>> > >>>> > [image removed] >>>> > >>>> > From: [email protected] [mailto:sipx-users- >>>> > [email protected]] On Behalf Of Michael Picher >>>> > Sent: Thursday, September 29, 2011 12:36 PM >>>> > To: Discussion list for users of sipXecs software >>>> > Subject: Re: [sipx-users] pfSense QoS >>>> > >>>> > Right but the point here is that the sipXecs server does not flag >>>> > its traffic for QoS (dscp or cos). Most people don't realize this. >>>> > >>>> > Also, if you use sipXbridge it's on the sipXecs server. >>>> > >>>> > Mike >>>> > On Thu, Sep 29, 2011 at 12:00 PM, Max DiOrio >>>> > <[email protected]> >>>> > wrote: >>>> > All VoIP traffic is showing up in the correct qVoIP queue in both >>>> directions. >>>> > >>>> > pfSense support set up floating rules to match traffic based on IP >>>> > address. >>>> > >>>> > I don't have access to the switch right now, I'm working on that, >>>> > but it's just a typical DLink smart switch, which I believe passes >>>> > QoS through. >>>> > >>>> > [image removed] >>>> > >>>> > From: [email protected] [mailto:sipx-users- >>>> > [email protected]] On Behalf Of Tony Graziano >>>> > Sent: Thursday, September 29, 2011 11:32 AM >>>> > >>>> > To: Discussion list for users of sipXecs software >>>> > Subject: Re: [sipx-users] pfSense QoS >>>> > >>>> > >>>> > On Thu, Sep 29, 2011 at 11:13 AM, Max DiOrio >>>> > <[email protected]> >>>> > wrote: >>>> > Tony, >>>> > >>>> > Here's a response from my ITSP: >>>> > >>>> > "Hm, that's odd. If there's no other traffic and the DSL circuit >>>> > is clean then you shouldn't have any packet loss. Why do you have >>>> > such a large circuit if you're only doing voice? >>>> > >>>> > They didn't really say that did they? Give me a break. How many >>>> > calls simultaneous *85'ish k sets your needed bandwidth. >>>> > >>>> > >>>> > Sounds like there's something else going on. A packet capture >>>> > would be helpful, and at least tell us if you're receiving all the >>>> > packets or not. Sometimes packet loss can be the result of late >>>> > arrivals to the jitter buffer - what's the jitter buffer set to? >>>> > Normally defaults are at 50ms. >>>> > >>>> > Regardless, we're coming from 64.246.135.202, SIP is on UDP 5080, >>>> > and RTP is on UDP ports 49152 - 53000." >>>> > >>>> > >>>> > OK, so we need to add ports 49152-53000 to this for them. I'll >>>> > whip you up a set set of files and send them to you. >>>> > >>>> > Let me know what your pfSense 2.0 wizard is now. >>>> > >>>> > Jitter buffer in sipXecs for 711u is 40ms isn't it? Would 10ms >>>> > added to the jitter buffer help? >>>> > >>>> > My users are having problems today actually. And this time, the >>>> > people outside are the ones hearing the choppy audio. My users >>>> > say what they hear is fine. >>>> > >>>> > I'm working with pfSense on an issue with the QoS queues that >>>> > appear to be broken. If I'm using the pfSense administration page >>>> > and transmitting HTTP data like refreshing the RRD page, while on >>>> > a call, the call audio gets choppy. The HTTP data is going >>>> > through the default Queue however, so I'm not sure why my VoIP >>>> > queue priority isn't taking over. >>>> > >>>> > It most likely because the calls are not recognized in the >>>> > status/ queues. You should see voice calls shaped in both >>>> > directions in in in/out queue when someone is talking. >>>> > >>>> > >>>> > Now this isn't a huge deal as I'm almost never in the admin pages >>>> > of pfSense or sipXecs, but if the queue isn't working for this, >>>> > maybe it's not working under load. I am seeing a lot of queue >>>> > drops in VoIP queue on occasion though. Love the RRD graphs. >>>> > >>>> > >>>> > Drops make me think there is a quality isssue/latency. Look at the >>>> > wan interface in question and look at the quality graph. Is ou >>>> > switch set for qos internally? >>>> > >>>> > Thanks! >>>> > >>>> > [image removed] >>>> > >>>> > From: [email protected] [mailto:sipx-users- >>>> > [email protected]] On Behalf Of Max DiOrio >>>> > Sent: Friday, September 23, 2011 1:38 PM >>>> > >>>> > To: Discussion list for users of sipXecs software >>>> > Subject: Re: [sipx-users] pfSense QoS >>>> > >>>> > That makes sense. I'll find out what the ports they use are and >>>> > get back to you. >>>> > >>>> > >>>> > [image removed] >>>> > >>>> > From: [email protected] [mailto:sipx-users- >>>> > [email protected]] On Behalf Of Tony Graziano >>>> > Sent: Friday, September 23, 2011 1:15 PM >>>> > To: Discussion list for users of sipXecs software >>>> > Subject: Re: [sipx-users] pfSense QoS >>>> > >>>> > OK. I updated MY scripts for 1.2.3, 2.0 is "very different" too. >>>> > What I found is that I need to also specifiy the media ports the >>>> > ITSP uses when they send me calls. >>>> > >>>> > So while I shape ports 5060, 5080, 30000-31000, I am not >>>> > necessarily applying anything to calls I receive from the ITSP if >>>> > they specify duifferent ports (which they will). >>>> > >>>> > So find out what media ports they specify and I can help you get >>>> > the new wizard working. As an example, maybe your outbound calls >>>> > are fine, but you might be having more quality issues with calls >>>> > from the itsp to you... >>>> > On Fri, Sep 23, 2011 at 11:59 AM, Max DiOrio >>>> > <[email protected]> >>>> > wrote: >>>> > CornerStone Telephone. They're a decent sized local carrier. >>>> > They cover all of New York State, Western Mass, and Northern PA. >>>> > >>>> > >>>> > [image removed] >>>> > >>>> > From: [email protected] [mailto:sipx-users- >>>> > [email protected]] On Behalf Of Tony Graziano >>>> > Sent: Friday, September 23, 2011 11:28 AM >>>> > >>>> > To: Discussion list for users of sipXecs software >>>> > Subject: Re: [sipx-users] pfSense QoS >>>> > >>>> > who is the itsp? >>>> > On Fri, Sep 23, 2011 at 11:18 AM, Max DiOrio >>>> > <[email protected]> >>>> > wrote: >>>> > Yes, it is a 2.0 install. pfSense said that it can't be the QoS >>>> > causing the problems as it doesn't show any dropped packets for >>>> > the queues, so I'm not overflowing them. >>>> > >>>> > I have UDP 30000-31000, UDP 5080, TCP/UDP 5060 NAT'd to the sipX >>>> > server. I believe I followed your wizard when I set this up back >>>> > in August. Have you updated it at all? >>>> > >>>> > With choppy audio and dropped calls I'm not sure how it would be a >>>> > port problem. I would think port problems would manifest >>>> > themselves as no audio/no connection issues. >>>> > >>>> > [image removed] >>>> > >>>> > From: [email protected] [mailto:sipx-users- >>>> > [email protected]] On Behalf Of Tony Graziano >>>> > Sent: Friday, September 23, 2011 11:05 AM >>>> > To: Discussion list for users of sipXecs software >>>> > Subject: Re: [sipx-users] pfSense QoS >>>> > >>>> > pfsense does floating rules but my guess is not all of the ports >>>> > are properly specified. Is this a 2.0 install? If so, you might >>>> > want my new wizard for setting that up. >>>> > On Fri, Sep 23, 2011 at 10:57 AM, Max DiOrio >>>> > <[email protected]> >>>> > wrote: >>>> > I was hoping someone can take a quick look at my QoS settings and >>>> > let me know if anything is off. My users are complaining that >>>> > they're having choppy audio and an occasional dropped call. Our >>>> > ITSP/ISP says everything is fine on their end. The QoS settings >>>> > were tweaked by pfSense. We have a DSL connection that's supposed >>>> to be 9/1.5 >>>> > >>>> > I'm just trying to figure out a way to pin the troubles back on >>>> the ITSP/ISP. >>>> > >>>> > <shaper> >>>> > <queue> >>>> > <interface>lan</interface> >>>> > <name>lan</name> >>>> > <scheduler>HFSC</scheduler> >>>> > <bandwidth/> >>>> > <bandwidthtype/> >>>> > <enabled>on</enabled> >>>> > <queue> >>>> > <name>qLink</name> >>>> > <interface>lan</interface> >>>> > <qlimit>500</qlimit> >>>> > <priority>2</priority> >>>> > <bandwidth>20</bandwidth> >>>> > <bandwidthtype>%</bandwidthtype> >>>> > <enabled>on</enabled> >>>> > <default>on</default> >>>> > <ecn>on</ecn> >>>> > </queue> >>>> > <queue> >>>> > <name>qInternet</name> >>>> > <interface>lan</interface> >>>> > <bandwidth>9</bandwidth> >>>> > <bandwidthtype>Mb</bandwidthtype> >>>> > <enabled>on</enabled> >>>> > <ecn>on</ecn> >>>> > <linkshare3>9Mb</linkshare3> >>>> > <linkshare>on</linkshare> >>>> > <upperlimit3>9Mb</upperlimit3> >>>> > <upperlimit>on</upperlimit> >>>> > <queue> >>>> > <name>qACK</name> >>>> > <interface>lan</interface> >>>> > <priority>6</priority> >>>> > <bandwidth>19.48</bandwidth> >>>> > <bandwidthtype>%</bandwidthtype> >>>> > <enabled>on</enabled> >>>> > <ecn>on</ecn> >>>> > <linkshare3>19.48%</linkshare3> >>>> > <linkshare>on</linkshare> >>>> > </queue> >>>> > <queue> >>>> > <name>qVoIP</name> >>>> > <interface>lan</interface> >>>> > <priority>7</priority> >>>> > <bandwidth>32</bandwidth> >>>> > <bandwidthtype>Kb</bandwidthtype> >>>> > <enabled>on</enabled> >>>> > <ecn>on</ecn> >>>> > <realtime3>240Kb</realtime3> >>>> > <realtime>on</realtime> >>>> > </queue> >>>> > </queue> >>>> > </queue> >>>> > <queue> >>>> > <interface>wan</interface> >>>> > <name>wan</name> >>>> > <scheduler>HFSC</scheduler> >>>> > <bandwidth>1572</bandwidth> >>>> > <bandwidthtype>Kb</bandwidthtype> >>>> > <enabled>on</enabled> >>>> > <queue> >>>> > <name>qACK</name> >>>> > <interface>wan</interface> >>>> > <priority>6</priority> >>>> > <bandwidth>19.602</bandwidth> >>>> > <bandwidthtype>%</bandwidthtype> >>>> > <enabled>on</enabled> >>>> > <ecn>on</ecn> >>>> > <linkshare3>19.602%</linkshare3> >>>> > <linkshare>on</linkshare> >>>> > </queue> >>>> > <queue> >>>> > <name>qDefault</name> >>>> > <interface>wan</interface> >>>> > <priority>3</priority> >>>> > <bandwidth>9.801</bandwidth> >>>> > <bandwidthtype>%</bandwidthtype> >>>> > <enabled>on</enabled> >>>> > <default>on</default> >>>> > <ecn>on</ecn> >>>> > </queue> >>>> > <queue> >>>> > <name>qVoIP</name> >>>> > <interface>wan</interface> >>>> > <priority>7</priority> >>>> > <bandwidth>32</bandwidth> >>>> > <bandwidthtype>Kb</bandwidthtype> >>>> > <enabled>on</enabled> >>>> > <ecn>on</ecn> >>>> > <realtime3>240Kb</realtime3> >>>> > <realtime>on</realtime> >>>> > </queue> >>>> > </queue> >>>> > </shaper> >>>> > >>>> > >>>> > [image removed] >>>> > >>>> > >>>> > _______________________________________________ >>>> > sipx-users mailing list >>>> > [email protected] >>>> > List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>>> > >>>> >>>> > >>>> > -- >>>> > ====================== >>>> > Tony Graziano, Manager >>>> > Telephone: 434.984.8430 >>>> > sip: [email protected] >>>> > Fax: 434.465.6833 >>>> > >>>> > Email: [email protected] >>>> > >>>> > LAN/Telephony/Security and Control Systems Helpdesk: >>>> > Telephone: 434.984.8426 >>>> > sip: [email protected] >>>> > >>>> > Helpdesk Contract Customers: >>>> > http://support.myitdepartment.net >>>> > >>>> > Blog: >>>> > http://blog.myitdepartment.net >>>> > >>>> > Linked-In Profile: >>>> > http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 >>>> > >>>> > Ask about our Internet faxservices! >>>> > >>>> > >>>> > _______________________________________________ >>>> > sipx-users mailing list >>>> > [email protected] >>>> > List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>>> > >>>> >>>> > >>>> > -- >>>> > ====================== >>>> > Tony Graziano, Manager >>>> > Telephone: 434.984.8430 >>>> > sip: [email protected] >>>> > Fax: 434.465.6833 >>>> > >>>> > Email: [email protected] >>>> > >>>> > LAN/Telephony/Security and Control Systems Helpdesk: >>>> > Telephone: 434.984.8426 >>>> > sip: [email protected] >>>> > >>>> > Helpdesk Contract Customers: >>>> > http://support.myitdepartment.net >>>> > >>>> > Blog: >>>> > http://blog.myitdepartment.net >>>> > >>>> > Linked-In Profile: >>>> > http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 >>>> > >>>> > Ask about our Internet faxservices! >>>> > >>>> > >>>> > _______________________________________________ >>>> > sipx-users mailing list >>>> > [email protected] >>>> > List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>>> > >>>> >>>> > >>>> > -- >>>> > ====================== >>>> > Tony Graziano, Manager >>>> > Telephone: 434.984.8430 >>>> > sip: [email protected] >>>> > Fax: 434.465.6833 >>>> > >>>> > Email: [email protected] >>>> > >>>> > LAN/Telephony/Security and Control Systems Helpdesk: >>>> > Telephone: 434.984.8426 >>>> > sip: [email protected] >>>> > >>>> > Helpdesk Contract Customers: >>>> > http://support.myitdepartment.net >>>> > >>>> > Blog: >>>> > http://blog.myitdepartment.net >>>> > >>>> > Linked-In Profile: >>>> > http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 >>>> > >>>> > Ask about our Internet faxservices! >>>> > >>>> > >>>> > _______________________________________________ >>>> > sipx-users mailing list >>>> > [email protected] >>>> > List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>>> > >>>> > >>>> > >>>> > -- >>>> > ====================== >>>> > Tony Graziano, Manager >>>> > Telephone: 434.984.8430 >>>> > sip: [email protected] >>>> > Fax: 434.465.6833 >>>> > >>>> > Email: [email protected] >>>> > >>>> > LAN/Telephony/Security and Control Systems Helpdesk: >>>> > Telephone: 434.984.8426 >>>> > sip: [email protected] >>>> > >>>> > Helpdesk Contract Customers: >>>> > http://support.myitdepartment.net >>>> > >>>> > Blog: >>>> > http://blog.myitdepartment.net >>>> > >>>> > Linked-In Profile: >>>> > http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 >>>> > >>>> > Ask about our Internet Fax services! >>>> > >>>> > >>>> > _______________________________________________ >>>> > sipx-users mailing list >>>> > [email protected] >>>> > List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>>> > >>>> >>>> > >>>> > -- >>>> > Michael Picher >>>> > eZuce >>>> > Director of Technical Services >>>> > O.978-296-1005 X2015 >>>> > M.207-956-0262 >>>> > @mpicher <http://twitter.com/mpicher> www.ezuce.com >>>> > >>>> > _______________________________________________ >>>> > sipx-users mailing list >>>> > [email protected] >>>> > List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>>> > >>>> > >>>> > >>>> > -- >>>> > Josh Patten >>>> > eZuce >>>> > Solutions Architect >>>> > O.978-296-1005 X2050 >>>> > M.979-574-5699 >>>> > >>>> > _______________________________________________ >>>> > sipx-users mailing list >>>> > [email protected] >>>> > List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>>> > >>>> >>>> > >>>> > -- >>>> > ====================== >>>> > Tony Graziano, Manager >>>> > Telephone: 434.984.8430 >>>> > sip: [email protected] >>>> > Fax: 434.465.6833 >>>> > >>>> > Email: [email protected] >>>> > >>>> > LAN/Telephony/Security and Control Systems Helpdesk: >>>> > Telephone: 434.984.8426 >>>> > sip: [email protected] >>>> > >>>> > Helpdesk Contract Customers: >>>> > http://support.myitdepartment.net >>>> > >>>> > Blog: >>>> > http://blog.myitdepartment.net >>>> > >>>> > Linked-In Profile: >>>> > http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 >>>> > >>>> > Ask about our Internet Fax services! >>>> > >>>> > >>>> > _______________________________________________ >>>> > sipx-users mailing list >>>> > [email protected] >>>> > List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>>> > >>>> >>>> > >>>> > -- >>>> > ====================== >>>> > Tony Graziano, Manager >>>> > Telephone: 434.984.8430 >>>> > sip: [email protected] >>>> > Fax: 434.465.6833 >>>> > >>>> > Email: [email protected] >>>> > >>>> > LAN/Telephony/Security and Control Systems Helpdesk: >>>> > Telephone: 434.984.8426 >>>> > sip: [email protected] >>>> > >>>> > Helpdesk Contract Customers: >>>> > http://support.myitdepartment.net >>>> > >>>> > Blog: >>>> > http://blog.myitdepartment.net >>>> > >>>> > Linked-In Profile: >>>> > http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 >>>> > >>>> > Ask about our Internet Fax services! >>>> > >>>> > >>>> > _______________________________________________ >>>> > sipx-users mailing list >>>> > [email protected] >>>> > List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>>> > >>>> >>>> > >>>> > -- >>>> > ====================== >>>> > Tony Graziano, Manager >>>> > Telephone: 434.984.8430 >>>> > sip: [email protected] >>>> > Fax: 434.465.6833 >>>> > >>>> > Email: [email protected] >>>> > >>>> > LAN/Telephony/Security and Control Systems Helpdesk: >>>> > Telephone: 434.984.8426 >>>> > sip: [email protected] >>>> > >>>> > Helpdesk Contract Customers: >>>> > http://support.myitdepartment.net >>>> > >>>> > Blog: >>>> > http://blog.myitdepartment.net >>>> > >>>> > Linked-In Profile: >>>> > http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 >>>> > >>>> > Ask about our Internet Fax services! >>>> > _______________________________________________ >>>> > sipx-users mailing list >>>> > [email protected] >>>> > List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>>> [attachment "Max DiOrio.vcf" deleted by Paul Scheepens/EPO] >>>> _______________________________________________ >>>> sipx-users mailing list >>>> [email protected] >>>> List Archive: >>> http://list.sipfoundry.org/archive/sipx-users/ >>> _______________________________________________ >>> sipx-users mailing list >>> [email protected] >>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>> >> >> >> >> -- >> ====================== >> Tony Graziano, Manager >> Telephone: 434.984.8430 >> sip: [email protected] >> Fax: 434.465.6833 >> >> Email: [email protected] >> >> LAN/Telephony/Security and Control Systems Helpdesk: >> Telephone: 434.984.8426 >> sip: [email protected] >> >> Helpdesk Contract Customers: >> http://support.myitdepartment.net >> Blog: >> http://blog.myitdepartment.net >> >> Linked-In Profile: >> http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 >> Ask about our Internet Fax services! >> _______________________________________________ >> sipx-users mailing list >> [email protected] >> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >> >> CONFIDENTIALITY NOTICE: >> >> The information contained in this message may be privileged and >> confidential. If you are NOT the intended recipient, please notify the >> sender immediately with a copy to [email protected] and destroy this >> message. Please be aware that email communication can be intercepted in >> transmission or misdirected. Your use of email to communicate protected >> health information to us indicates that you acknowledge and accept the >> possible risks associated with such communication. Please consider >> communicating any sensitive information by telephone, fax or mail. If you >> do not wish to have your information sent by email, please contact the >> sender immediately. >> >> _______________________________________________ >> sipx-users mailing list >> [email protected] >> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >> > > > > -- > ====================== > Tony Graziano, Manager > Telephone: 434.984.8430 > sip: [email protected] > Fax: 434.465.6833 > > Email: [email protected] > > LAN/Telephony/Security and Control Systems Helpdesk: > Telephone: 434.984.8426 > sip: [email protected] > > Helpdesk Contract Customers: > http://support.myitdepartment.net > Blog: > http://blog.myitdepartment.net > > Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 > Ask about our Internet Fax services! > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > > CONFIDENTIALITY NOTICE: > > The information contained in this message may be privileged and confidential. > If you are NOT the intended recipient, please notify the sender immediately > with a copy to [email protected] and destroy this message. Please be > aware that email communication can be intercepted in transmission or > misdirected. Your use of email to communicate protected health information > to us indicates that you acknowledge and accept the possible risks associated > with such communication. Please consider communicating any sensitive > information by telephone, fax or mail. If you do not wish to have your > information sent by email, please contact the sender immediately. > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ >
-- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: [email protected] Fax: 434.465.6833 Email: [email protected] LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected] Helpdesk Contract Customers: http://support.myitdepartment.net Blog: http://blog.myitdepartment.net Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/
