external calls register, internal calls do not.
Tony Graziano wrote: > The externals calls don't register in cdr? Definitely a config issue. Are > you sure your sip alg is off in the sonicwall? > ============================ > Tony Graziano, Manager > Telephone: 434.984.8430 > Fax: 434.984.8431 > > Email: tgrazi...@myitdepartment.net > > LAN/Telephony/Security and Control Systems Helpdesk: > Telephone: 434.984.8426 > Fax: 434.984.8427 > > Helpdesk Contract Customers: > http://www.myitdepartment.net/gethelp/ > > ----- Original Message ----- > From: Tony Graziano <tgrazi...@myitdepartment.net> > To: 'sipx-users@list.sipfoundry.org' <sipx-users@list.sipfoundry.org> > Sent: Wed Sep 08 13:30:15 2010 > Subject: Re: [sipx-users] Call drops after 1 min & 29 secs > > If you have an internal-internal call where sipx is not inbetween (no remote > user, no internet, aa, voicemail or siptrunk) you can "unplug" the ethernet > to sipx with an internal-internal call as long as media is established. All > other bets are off. > ============================ > Tony Graziano, Manager > Telephone: 434.984.8430 > Fax: 434.984.8431 > > Email: tgrazi...@myitdepartment.net > > LAN/Telephony/Security and Control Systems Helpdesk: > Telephone: 434.984.8426 > Fax: 434.984.8427 > > Helpdesk Contract Customers: > http://www.myitdepartment.net/gethelp/ > > ----- Original Message ----- > From: sipx-users-boun...@list.sipfoundry.org > <sipx-users-boun...@list.sipfoundry.org> > To: Discussion list for users of sipXecs software > <sipx-users@list.sipfoundry.org> > Sent: Wed Sep 08 13:13:57 2010 > Subject: Re: [sipx-users] Call drops after 1 min & 29 secs > > Tony, > > I've looked at the sipx server and cpu usage never gets above 27% > utilization and the sonicwall (SW) never gets above 10% utilization so > neither of those are an issue. > > The SW is able to prioritize ports and traffic as well as assign > individual ports and port groups to separate subnets. It handles VLANs > as well. > > Incidentally, the hard and soft phones are on the same subnet and are > plugged into the same network switch. That switch is up-linked to the > switch which the sipX server is connected to (which is, in turn, > connected to the SW). If after a call connects between these two > extensions, if I turn off the SW, the call should stay up should it not? > This is the way other network traffic works as long as its on the same > subnet. This leads me to think that it is not a SW issue at all (ext to > ext calls do not even register in the SW SIP call status logs) - just a > guess, I haven't tested it. > > Besides, if it is a network issue, why would sip to cell (both att and > verizon) & sip to POTS be perfectly clear but sip to vonage & internal > ext to internal ext have outbound voice issues? > > I'll do the reinstall and move it to its own subnet and report back. > > Stiles > > Tony Graziano wrote: > >> You would be best to use a reinstall. Especially if the system is a >> very basic install. You might consider looking at your sonicwall AGAIN >> and making sure all the SIP stuff is turned off and that consistent >> NAT is on. >> >> I don't know if there is a way to prioritize certain ports with the >> sonicwall. I have made a bandwidth shaping script available for >> pfsense that many people seem to use without issue. >> >> I don't assume the issue is sipx. Take a look at >> >> top >> >> in sipx and see if CPU is high or swap is being used. >> >> It is more likely a firewall cpu related issue. >> >> I normally also suggest putting sipx on its own vlan, and prioritizing >> the LAN traffic on the vlan (if you have those capabilities. You >> should also look to see if there are generic ethernet errors >> (collisions, crc, etc.) on the sipx ethernet port and on your lan >> segment where sipx and any pc's/phones are in use. >> >> I somewhat think a rebuild is unnecessary. >> >> On Wed, Sep 8, 2010 at 12:00 PM, Stiles Watson <wat...@datatek-net.com >> <mailto:wat...@datatek-net.com>> wrote: >> >> Another interesting note: ext to ext calls do not drop at all, BUT >> the voice does become increasingly drawn out over time. This is >> occurring on a call between the Polycom IP 335 hardphone and a >> kphone softphone running on Ubuntu. >> >> Since this call is not going out to Teliax, this has to be either >> a sipX issue or a network issue. It seems as long as the call is >> sip-to-sip there is an issue, sip-to-POTS seems to have no voice >> issues. >> >> I'm going to move the sipx server to its own subnet, reconfigure >> and try this all again. From looking a previous posts, to change >> IPs requires a reinstall, correct? >> >> Stiles >> >> Stiles Watson wrote: >> >>> Premature rejoicing - problem is not solved. Right after I made >>> the change, I voluntarily ended the next call at 2min. But every >>> call after that has disconnected at 1min 29sec as before. >>> >>> Other interesting things to note. If I make a call through >>> sipx/teliax to a cell phone the call sounds fine. However, if I >>> make a call to someone who is using Vonage, the person called >>> sounds fine to me, but my voice is slow and drawn out (the person >>> on the other end said I sounded drugged or drunk). >>> >>> On the sonicwall I also checked "Enable SIP Back-to-Back User >>> Agent (B2BUA) support," but no noticeable difference occurred (by >>> default it is unchecked). >>> >>> Other SIP settings on the sonicwall (all are using defaults): >>> >>> Permit non-SIP packets on signaling port: disabled >>> SIP Signaling inactivity time out (seconds): 1800 >>> SIP Media inactivity time out (seconds): 120 >>> Additional SIP signaling port (UDP) for transformations (optional): 0 >>> >>> Stiles >>> >>> Tony Graziano wrote: >>> >>>> I think in sonicwall it is called consistent nat. >>>> >>>> >>>> To enable Consistent NAT, select the Enable Consistent NAT >>>> setting and click Apply. This checkbox >>>> is disabled by default. >>>> >>>> >>>> On Wed, Sep 8, 2010 at 10:31 AM, Tony Graziano >>>> <tgrazi...@myitdepartment.net >>>> <mailto:tgrazi...@myitdepartment.net>> wrote: >>>> >>>> your firewall is a REALLY important pice of the puzzle. >>>> Thanks for finally telling us what it is. >>>> >>>> In the sonicwall: >>>> >>>> 1. Open web administration interface >>>> 2. Select VoIP from the left menu >>>> 3. Check/uncheck Enable SIP Transformations >>>> 4. Click Accept >>>> >>>> >>>> Then try your call again and see if it disconnects at the 90 >>>> second call timer. The call is because both sides have never >>>> agreed the connection is "OK". So, the FIRST thing to do is >>>> make sure your firewall is configured to disable the SIP ALG >>>> and provide symmetric nat. >>>> >>>> 1. DISABLE SIP ALG (see above). >>>> 2. Make sure the NAT is "SYMMETRIC" NAT on the sonicwall. >>>> >>>> Sonicwall lovers feel free to share "how-to" on the >>>> sonicwall, especially how to deploy symmetric nat >>>> >>>> >>>> On Wed, Sep 8, 2010 at 10:22 AM, Tony Graziano >>>> <tgrazi...@myitdepartment.net >>>> <mailto:tgrazi...@myitdepartment.net>> wrote: >>>> >>>> I think you need to disable the sip alg on the sonicwall. >>>> >>>> >>>> On Wed, Sep 8, 2010 at 10:19 AM, Stiles Watson >>>> <wat...@datatek-net.com <mailto:wat...@datatek-net.com>> >>>> wrote: >>>> >>>> That was my initial sipX setup as well (except I had >>>> Auth User set equal to User). >>>> >>>> On the Teliax side under device settings did you do >>>> either of the following? >>>> >>>> * enable DNIS so they send the number instead of >>>> the user in the SIP INVITE? >>>> * enter your pubilc IP >>>> >>>> The reason I ask is because the "User part of INVITE >>>> SIP URI is a phone number" checkbox under the sipX >>>> ITSP Account settings defaults to 'enabled', but >>>> unless you enable DNIS on the Teliax side, this is >>>> not the case (unless I'm misunderstanding the >>>> something works). >>>> >>>> Firewall: >>>> >>>> I'm using a Sonicwall NSA 240. I have NAT policies >>>> which forward ports UDP 5080, UDP&TCP 5060-5061 & >>>> UDP 30000-31000 untranslated to the sipX server >>>> (we're a small shop so everything is running on one >>>> server). Are you saying that the invite actually >>>> comes to UDP port 37678? >>>> >>>> >>>> Stiles >>>> >>>> Dave Redmore wrote: >>>> >>>>> My settings for the gateway are all default - Under >>>>> "Configuration", I defined "Address" as >>>>> "den.teliax.net <http://den.teliax.net>" - Under >>>>> "CallerID" I set the "Default Caller ID" to my >>>>> incoming phone number - under "ITSP Account" I >>>>> defined "Username" ("Authentication Username" is >>>>> left blank), "Password" and checked "Register on >>>>> Initialization". Everything else is defaulted. >>>>> >>>>> When I do a packet capture on the WAN port of the >>>>> pfSense - I see Teliax sending me OPTION pings to >>>>> the NAT'd port number (37678 in this case). When I >>>>> look at the State table I see active states from >>>>> sipX:5080 -> pfSense:37678 -> den.teliax.net:5060 >>>>> <http://den.teliax.net:5060>. Incoming Invite is >>>>> to the external port (37678). >>>>> >>>>> So, it looks like FreeSwitch on Teliax end is doing >>>>> its NAT compensation magic and pfSense is staying >>>>> out of the way. >>>>> >>>>> Interestingly, when I looked at the packet capture >>>>> and state tables - in addition to the connection >>>>> from sipXbridge on port 5080 - there is also a >>>>> connection maintained from sipXecs on port 5060 >>>>> (which in this case is being NAT'd to port 5041). >>>>> So, I am getting OPTION pings to port 37678 >>>>> (translated to 5080), to which sipXbridge respondes >>>>> "406 Not Acceptable" and OPTION pings to port 5041 >>>>> (translated to 5060) to which sipX responses "200 >>>>> Okay". The "Request URI" for the OPTION ping to >>>>> sipXbridge looks like "sip:teliaxusername@(Ext. IP >>>>> Address):37678;transport=udp;fs_nat=yes". The >>>>> "Request URI" for the OPTION ping to sipX looks >>>>> like "sip:s@(Ext IP Address):5041;fs_nat=yes". >>>>> >>>>> Dave >>>>> >>>>> >>>>> ----- Original Message ----- >>>>> From: "Tony Graziano" >>>>> <tgrazi...@myitdepartment.net> >>>>> <mailto:tgrazi...@myitdepartment.net> >>>>> To: sipx-users@list.sipfoundry.org >>>>> <mailto:sipx-users@list.sipfoundry.org> >>>>> Sent: Tuesday, September 7, 2010 6:20:04 PM GMT >>>>> -06:00 US/Canada Central >>>>> Subject: Re: [sipx-users] Call drops after 1 min & >>>>> 29 secs >>>>> >>>>> Then it would be good to have a template for them. >>>>> Can you detail an example >>>>> of your gateway? Are they sending on port 5080? >>>>> What did you have to do to >>>>> get them to send on port 5080? >>>>> ============================ >>>>> Tony Graziano, Manager >>>>> Telephone: 434.984.8430 >>>>> Fax: 434.984.8431 >>>>> >>>>> Email: tgrazi...@myitdepartment.net >>>>> <mailto:tgrazi...@myitdepartment.net> >>>>> >>>>> LAN/Telephony/Security and Control Systems Helpdesk: >>>>> Telephone: 434.984.8426 >>>>> Fax: 434.984.8427 >>>>> >>>>> Helpdesk Contract Customers: >>>>> http://www.myitdepartment.net/gethelp/ >>>>> >>>>> ----- Original Message ----- >>>>> From: sipx-users-boun...@list.sipfoundry.org >>>>> <mailto:sipx-users-boun...@list.sipfoundry.org> >>>>> <sipx-users-boun...@list.sipfoundry.org> >>>>> <mailto:sipx-users-boun...@list.sipfoundry.org> >>>>> To: Discussion list for users of sipXecs software >>>>> <sipx-users@list.sipfoundry.org> >>>>> <mailto:sipx-users@list.sipfoundry.org> >>>>> Sent: Tue Sep 07 19:17:14 2010 >>>>> Subject: Re: [sipx-users] Call drops after 1 min & >>>>> 29 secs >>>>> >>>>> I can report that I have 4.2.1 installed and >>>>> working very nicely with >>>>> Teliax. I have configured a gateway using very >>>>> "plain vanilla" settings and >>>>> it worked pretty much "right out of the box". >>>>> Incoming calls and outgoing. >>>>> MOH and transfers all seem to work fine. I >>>>> currently have a Grandstream >>>>> GXP-2020 and Polycom 301 on that system for >>>>> testing/evaluation and will >>>>> probably put it into "production" in the next day >>>>> or two. I have sipX >>>>> sitting behind a pfSense firewall. I am using the >>>>> Denver proxy for incoming >>>>> calls and outgoing route to their Chicago proxy. >>>>> >>>>> >>>>> I am limited in choices for ITSPs that can provide >>>>> local DIDs and have been >>>>> working with Teliax for about 4-5 years. I >>>>> personally find them to be pretty >>>>> good and a decent value when using the PAYG services. >>>>> >>>>> >>>>> Dave >>>>> >>>>> ----- Original Message ----- >>>>> From: "Tony Graziano" >>>>> <tgrazi...@myitdepartment.net> >>>>> <mailto:tgrazi...@myitdepartment.net> >>>>> To: "Discussion list for users of sipXecs software" >>>>> <sipx-users@list.sipfoundry.org> >>>>> <mailto:sipx-users@list.sipfoundry.org> >>>>> Sent: Tuesday, September 7, 2010 5:40:35 PM GMT >>>>> -06:00 US/Canada Central >>>>> Subject: Re: [sipx-users] Call drops after 1 min & >>>>> 29 secs >>>>> >>>>> That still references using port 5060 and ip >>>>> authentication. He would need >>>>> to ensure they support using the public IP at port >>>>> 5080. It sounds like he >>>>> may have to get them to do that for him manually. >>>>> >>>>> >>>>> On Tue, Sep 7, 2010 at 6:29 PM, Todd Hodgen < >>>>> thod...@verizon.net <mailto:thod...@verizon.net> > >>>>> wrote: >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> There have been some discussions about this ITSP on >>>>> the list in the past. >>>>> >>>>> >>>>> >>>>> I did find this one. >>>>> >>>>> http://forum.sipfoundry.org/index.php?t=msg&goto=44468&S=9a2fe924342a700db212b8481e97cc22#msg_44468 >>>>> >>>>> <http://forum.sipfoundry.org/index.php?t=msg&goto=44468&S=9a2fe924342a700db212b8481e97cc22#msg_44468> >>>>> >>>>> >>>>> >>>>> Not sure if this fixes your problems, but it does >>>>> reference a dashboard that >>>>> you may want to access for some configuration >>>>> options. I’d search more of >>>>> the archives as well for people that have >>>>> referenced this ITSP and have >>>>> successfully gotten it working. >>>>> >>>>> >>>>> >>>>> >>>>> From: sipx-users-boun...@list.sipfoundry.org >>>>> <mailto:sipx-users-boun...@list.sipfoundry.org> >>>>> [mailto: >>>>> sipx-users-boun...@list.sipfoundry.org >>>>> <mailto:sipx-users-boun...@list.sipfoundry.org> ] >>>>> On Behalf Of Tony Graziano >>>>> Sent: Tuesday, September 07, 2010 3:16 PM >>>>> >>>>> To: Discussion list for users of sipXecs software >>>>> >>>>> Subject: Re: [sipx-users] Call drops after 1 min & >>>>> 29 secs >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> If your firewall has a packet capture facility, you >>>>> can do a pcap on the WAN >>>>> interface and see what they are sending. >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> I would suspect if anyone has a working teliax >>>>> config they will share it. >>>>> >>>>> >>>>> On Tue, Sep 7, 2010 at 6:15 PM, Tony Graziano < >>>>> tgrazi...@myitdepartment.net >>>>> <mailto:tgrazi...@myitdepartment.net> >>>>> > wrote: >>>>> >>>>> I think unless you are wed to them, it would be >>>>> easier to switch to a >>>>> "normal" provider. Supported providers in the >>>>> templates usually take 5 >>>>> minutes to setup. I HOPE your firewall is doing >>>>> manual versus automatic NAT. >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> I looked at Teliax and they seem "residentially" >>>>> focused, and really >>>>> expensive for business plans. >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> On Tue, Sep 7, 2010 at 6:12 PM, Stiles Watson < >>>>> wat...@datatek-net.com >>>>> <mailto:wat...@datatek-net.com> > >>>>> wrote: >>>>> >>>>> >>>>> Unfortunately, there is no way in the Teliax portal >>>>> to even see if you are >>>>> registered, much less what port. >>>>> >>>>> The reason I had 5060 forwarded to sipx was this >>>>> was how I had Trixbox CE >>>>> setup and working. There is nothing in my Teliax >>>>> setup which I changed to >>>>> force 5060. >>>>> >>>>> Thanks for the pdf. With the exception of the SIP >>>>> port, I think I have >>>>> everything setup correctly. I changed my NAT rules >>>>> to forward 5080 instead >>>>> of 5060 and the call acted exactly the same. >>>>> >>>>> I've also asked Teliax if they have config info for >>>>> sipX and they said no, >>>>> but many are using the two together successfully. >>>>> Here is their exact >>>>> response: >>>>> >>>>> "We do not have a have a configuration for them. >>>>> However, I know that many >>>>> customers have used SIPXECS without a problem. The >>>>> main information you need >>>>> is the username, secret, and host that you are >>>>> registering to." >>>>> >>>>> I've asked them what port they are sending the >>>>> INVITE on and am waiting on a >>>>> response. >>>>> >>>>> Any other suggestions/thoughts? >>>>> >>>>> Stiles >>>>> >>>>> Tony Graziano wrote: >>>>> >>>>> >>>>> >>>>> It means they are not acking the call. I suspect >>>>> this is because sipxbridge >>>>> may not be involved in the call, and only sipxproxy >>>>> is, which would be >>>>> problematic for a lot of call scenarios (like >>>>> transfers). >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> I'm confused though, because it seems you are >>>>> breaking "rule #1" when using >>>>> sipxbridge... you are having the calls sent to port >>>>> 5060 instead of 5080. >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> When you register with teliax, can you see on their >>>>> portal what port you are >>>>> registering on? Can you confirm they are sending to >>>>> you on a specific port? >>>>> If so, what port? >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> You should peek at this: >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> http://blog.myitdepartment.net/wp-content/uploads/2009/11/Call-Setup-Example-sipXecs-through-ITSP1.pdf >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> Somehow I don't believe you are doing it quite like >>>>> that. >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> On Tue, Sep 7, 2010 at 5:18 PM, Stiles Watson < >>>>> wat...@datatek-net.com >>>>> <mailto:wat...@datatek-net.com> > >>>>> wrote: >>>>> >>>>> >>>>> Running >>>>> >>>>> • sipXecs v 4.2.1 >>>>> • ITSP is Teliax >>>>> • SIP ports 5060 & 5061 are routed to sipX server >>>>> • RTP ports 30000-31000 are routed to sipX server >>>>> • Polycom IP 335 hardphone >>>>> >>>>> >>>>> I'm able to place incoming and outgoing calls >>>>> through Teliax, but calls >>>>> consistently drop after 1 min. 29 sec. >>>>> >>>>> Teliax device config change attempts: >>>>> >>>>> • Enable DNIS (teliax sends number in sip >>>>> invite instead of user) >>>>> >>>>> >>>>> >>>>> >>>>> • result: calls still drop after 1 min. 29 >>>>> sec., but made call >>>>> routing easier via a custom DID! >>>>> >>>>> • Entered public IP under "Your IP" >>>>> >>>>> >>>>> >>>>> • This is optional and resulted in not >>>>> being able to make inbound >>>>> calls (I read in the archives that this is >>>>> recommended with Teliax - is >>>>> there a sipX config change needed to make this work?) >>>>> >>>>> sipX config for teliax SIP trunk Gateway: >>>>> >>>>> • Configuration >>>>> >>>>> >>>>> >>>>> >>>>> • Enabled: yes • Name: teliax >>>>> • SBC Route: sipXbridge-1 >>>>> • Address: den.teliax.net >>>>> <http://den.teliax.net> (this has to match with the >>>>> proxy setting >>>>> in your teliax account) >>>>> • Port: 0 >>>>> • Transport protocol: Auto >>>>> • Location: all >>>>> • Shared: yes >>>>> >>>>> >>>>> • Caller ID >>>>> >>>>> >>>>> >>>>> • Default Caller ID: set this to the number >>>>> from Teliax • >>>>> use default for all other settings >>>>> >>>>> >>>>> • Dial Plan >>>>> >>>>> >>>>> >>>>> • Enabled and added both Local & Long >>>>> Distance dial plans to this >>>>> gateway >>>>> >>>>> • ITSP Account >>>>> >>>>> >>>>> >>>>> • Username: use teliax username • >>>>> Authentication Username: >>>>> same as Username >>>>> • Password: use teliax device password >>>>> • Register on init: yes >>>>> • ITSP server address: same as >>>>> Config-->Address above >>>>> • Use public address for call setup: yes (I >>>>> tried both yes and no, >>>>> calls completed either way and did not effect >>>>> disconnect problem) >>>>> • Strip private headers: default >>>>> • Use default asserted identity: default >>>>> • Asserted identity: default >>>>> • Use default preferred identity: default >>>>> • Preferred identity: default >>>>> • User part of INVITE SIP URI is a phone >>>>> number: NO >>>>> • ITSP Registrar Address: default >>>>> • ITSP Registrar Port: default >>>>> • Registration interval: default >>>>> • Session Timer Interval: default >>>>> • Method to use for SIP keepalive: Empty >>>>> SIP message (also tried >>>>> None) >>>>> • Method to use for RTP keepalive: Replay >>>>> last sent packet (also >>>>> tried None) >>>>> • Route by To Header: default >>>>> >>>>> >>>>> Any thoughts as to why the calls would drop after 1 >>>>> min. 29 sec.? >>>>> >>>>> Stiles >>>>> >>>>> >>>>> _______________________________________________ >>>>> sipx-users mailing list >>>>> sipx-users@list.sipfoundry.org >>>>> <mailto:sipx-users@list.sipfoundry.org> >>>>> List Archive: >>>>> http://list.sipfoundry.org/archive/sipx-users/ >>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> ====================== >>>>> Tony Graziano, Manager >>>>> Telephone: 434.984.8430 >>>>> sip: tgrazi...@voice.myitdepartment.net >>>>> <mailto:tgrazi...@voice.myitdepartment.net> >>>>> Fax: 434.984.8431 >>>>> >>>>> Email: tgrazi...@myitdepartment.net >>>>> <mailto:tgrazi...@myitdepartment.net> >>>>> >>>>> LAN/Telephony/Security and Control Systems Helpdesk: >>>>> Telephone: 434.984.8426 >>>>> sip: helpd...@voice.myitdepartment.net >>>>> <mailto:helpd...@voice.myitdepartment.net> >>>>> 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 >>>>> sipx-users@list.sipfoundry.org >>>>> <mailto:sipx-users@list.sipfoundry.org> List Archive: >>>>> http://list.sipfoundry.org/archive/sipx-users/ >>>>> >>>>> >>>>> _______________________________________________ >>>>> sipx-users mailing list >>>>> sipx-users@list.sipfoundry.org >>>>> <mailto:sipx-users@list.sipfoundry.org> >>>>> List Archive: >>>>> http://list.sipfoundry.org/archive/sipx-users/ >>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> ====================== >>>>> Tony Graziano, Manager >>>>> Telephone: 434.984.8430 >>>>> sip: tgrazi...@voice.myitdepartment.net >>>>> <mailto:tgrazi...@voice.myitdepartment.net> >>>>> Fax: 434.984.8431 >>>>> >>>>> Email: tgrazi...@myitdepartment.net >>>>> <mailto:tgrazi...@myitdepartment.net> >>>>> >>>>> LAN/Telephony/Security and Control Systems Helpdesk: >>>>> Telephone: 434.984.8426 >>>>> sip: helpd...@voice.myitdepartment.net >>>>> <mailto:helpd...@voice.myitdepartment.net> >>>>> 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 >>>>> sip: tgrazi...@voice.myitdepartment.net >>>>> <mailto:tgrazi...@voice.myitdepartment.net> >>>>> Fax: 434.984.8431 >>>>> >>>>> Email: tgrazi...@myitdepartment.net >>>>> <mailto:tgrazi...@myitdepartment.net> >>>>> >>>>> LAN/Telephony/Security and Control Systems Helpdesk: >>>>> Telephone: 434.984.8426 >>>>> sip: helpd...@voice.myitdepartment.net >>>>> <mailto:helpd...@voice.myitdepartment.net> >>>>> 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 >>>>> sipx-users@list.sipfoundry.org >>>>> <mailto:sipx-users@list.sipfoundry.org> >>>>> List Archive: >>>>> http://list.sipfoundry.org/archive/sipx-users/ >>>>> >>>>> >>>>> >>>>> -- >>>>> ====================== >>>>> Tony Graziano, Manager >>>>> Telephone: 434.984.8430 >>>>> sip: tgrazi...@voice.myitdepartment.net >>>>> <mailto:tgrazi...@voice.myitdepartment.net> >>>>> Fax: 434.984.8431 >>>>> >>>>> Email: tgrazi...@myitdepartment.net >>>>> <mailto:tgrazi...@myitdepartment.net> >>>>> >>>>> LAN/Telephony/Security and Control Systems Helpdesk: >>>>> Telephone: 434.984.8426 >>>>> sip: helpd...@voice.myitdepartment.net >>>>> <mailto:helpd...@voice.myitdepartment.net> >>>>> 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 >>>>> sipx-users@list.sipfoundry.org >>>>> <mailto:sipx-users@list.sipfoundry.org> List Archive: >>>>> http://list.sipfoundry.org/archive/sipx-users/ >>>>> _______________________________________________ >>>>> sipx-users mailing list >>>>> sipx-users@list.sipfoundry.org >>>>> <mailto:sipx-users@list.sipfoundry.org> >>>>> List Archive: >>>>> http://list.sipfoundry.org/archive/sipx-users/ >>>>> >>>>> ------------------------------------------------------------------------ >>>>> _______________________________________________ >>>>> sipx-users mailing list >>>>> sipx-users@list.sipfoundry.org >>>>> <mailto:sipx-users@list.sipfoundry.org> List >>>>> Archive: http://list.sipfoundry.org/archive/sipx-users/ >>>>> >>>> _______________________________________________ >>>> sipx-users mailing list >>>> sipx-users@list.sipfoundry.org >>>> <mailto:sipx-users@list.sipfoundry.org> >>>> List Archive: >>>> http://list.sipfoundry.org/archive/sipx-users/ >>>> >>>> >>>> >>>> >>>> -- >>>> ====================== >>>> Tony Graziano, Manager >>>> Telephone: 434.984.8430 >>>> sip: tgrazi...@voice.myitdepartment.net >>>> <mailto:tgrazi...@voice.myitdepartment.net> >>>> Fax: 434.984.8431 >>>> >>>> Email: tgrazi...@myitdepartment.net >>>> <mailto:tgrazi...@myitdepartment.net> >>>> >>>> LAN/Telephony/Security and Control Systems Helpdesk: >>>> Telephone: 434.984.8426 >>>> sip: helpd...@voice.myitdepartment.net >>>> <mailto:helpd...@voice.myitdepartment.net> >>>> 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 >>>> sip: tgrazi...@voice.myitdepartment.net >>>> <mailto:tgrazi...@voice.myitdepartment.net> >>>> Fax: 434.984.8431 >>>> >>>> Email: tgrazi...@myitdepartment.net >>>> <mailto:tgrazi...@myitdepartment.net> >>>> >>>> LAN/Telephony/Security and Control Systems Helpdesk: >>>> Telephone: 434.984.8426 >>>> sip: helpd...@voice.myitdepartment.net >>>> <mailto:helpd...@voice.myitdepartment.net> >>>> 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 >>>> sip: tgrazi...@voice.myitdepartment.net >>>> <mailto:tgrazi...@voice.myitdepartment.net> >>>> Fax: 434.984.8431 >>>> >>>> Email: tgrazi...@myitdepartment.net >>>> <mailto:tgrazi...@myitdepartment.net> >>>> >>>> LAN/Telephony/Security and Control Systems Helpdesk: >>>> Telephone: 434.984.8426 >>>> sip: helpd...@voice.myitdepartment.net >>>> <mailto:helpd...@voice.myitdepartment.net> >>>> 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 >>>> sipx-users@list.sipfoundry.org >>>> <mailto:sipx-users@list.sipfoundry.org> >>>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>>> >>> ------------------------------------------------------------------------ >>> _______________________________________________ sipx-users >>> mailing list sipx-users@list.sipfoundry.org >>> <mailto:sipx-users@list.sipfoundry.org> List Archive: >>> http://list.sipfoundry.org/archive/sipx-users/ >>> >> _______________________________________________ >> sipx-users mailing list >> sipx-users@list.sipfoundry.org <mailto:sipx-users@list.sipfoundry.org> >> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >> >> >> >> >> -- >> ====================== >> Tony Graziano, Manager >> Telephone: 434.984.8430 >> sip: tgrazi...@voice.myitdepartment.net >> <mailto:tgrazi...@voice.myitdepartment.net> >> Fax: 434.984.8431 >> >> Email: tgrazi...@myitdepartment.net <mailto:tgrazi...@myitdepartment.net> >> >> LAN/Telephony/Security and Control Systems Helpdesk: >> Telephone: 434.984.8426 >> sip: helpd...@voice.myitdepartment.net >> <mailto:helpd...@voice.myitdepartment.net> >> 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 >> sipx-users@list.sipfoundry.org >> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >> > _______________________________________________ > sipx-users mailing list > sipx-users@list.sipfoundry.org > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > _______________________________________________ > sipx-users mailing list > sipx-users@list.sipfoundry.org > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > > > _______________________________________________ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
