using just the port definitions? On Fri, Sep 30, 2011 at 1:02 PM, Max DiOrio <[email protected]> wrote: > Unfortunately it was during a chat sessions that wasn't saved, so I have no > record of that. However they have looked at my QoS settings multiple times > and were perfectly happy with what they were looking at. Even describing the > issues I was having. > > I am in the process of transferring 27 pcap files (501meg) from the pfSense > box to my desktop and had two active calls going without an issue. > Transferring the files at 1.4-1.5Mb/s with no calls. With 2 calls, it > throttled the qDefault down to 1.3 and quality remained good, so it's > obviously working. So far things seem better. > > > > > -----Original Message----- > From: [email protected] > [mailto:[email protected]] On Behalf Of Tony Graziano > Sent: Friday, September 30, 2011 12:35 PM > To: Discussion list for users of sipXecs software > Subject: Re: [sipx-users] pfSense QoS > > 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/ > > 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/
