I am not sure your firewall (Dave) is configured correctly, your NAT is not symmetric. I doubt that you can support remote users.
I agree with everything else though. ============================ 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 22:23:31 2010 Subject: Re: [sipx-users] Call drops after 1 min & 29 secs Please know I am as green as you with sipX - so take my opinions in that context... I can tell you that this was my approach - I segmented sipX on it's own subnet, mostly because it seemed that the easiest way to make sipX "happy" was to just let it be the DHCP, DNS and NTP server for the phones. I also created a separate sub-domain for the phones. Once I laid out my topology like this, sipX installed from the ISO very cleanly, I connected a Grandstream and a Polycom phone very easily and setup a gateway to Teliax without problems. I would really recommend getting that SonicWall out of the picture for now, if at all possible. It seems to be too smart for its own good. Some would disagree with this, but I feel like using a very simple firewall/router - maybe a cheap ASUS wireless router loaded with DD-WRT - and run sipX behind a simple NAT seems to work pretty darn good. Teliax is running FreeSwitch on their end and FreeSwitch seems to do a very nice job of detecting and managing NAT, so long as there is no Layer 5/7 or Application layer packet fiddling going on. Also, for now, make sure you are restricting everything to ULAW codec - just for simplicity. I personally feel like the strange audio issues you were hearing almost have to be a codec issue more than a networking thing. Dave ----- Original Message ----- From: "Stiles Watson" <wat...@datatek-net.com> To: "Discussion list for users of sipXecs software" <sipx-users@list.sipfoundry.org> Sent: Wednesday, September 8, 2010 5:02:53 PM GMT -06:00 US/Canada Central Subject: Re: [sipx-users] Call drops after 1 min & 29 secs Well, I've done the reinstall and am having other issues so I can not really answer these questions. Dave is right about the logs I was talking about. The calls did show up in the sipx logs and not in the SW "SIP Call Status" logs. I do not doubt that there is some fundamental thing that I am misunderstanding. I have, in the past, setup an NEC hybrid IP/PBX and actually knew more than the phone guys the CLEC kept sending out to fix their own problems, so while I'm by no stretch of the imagination an expert, I'm not an idiot. I very much appreciate the time everyone has given me (especially Tony and Dave) and I understand that I may frustrate the old-hats, but I promise I've read a good bit of the docs and I always search the archives before I ask a question here. Re-install issue: First, let me say that I did the reinstall twice with the same result. The issue I'm having is that the DHCP test keeps failing, but there is a functioning DHCP server on the new subnet. If I plug a laptop in to the switch attached to the subnet, I get a valid IP address. The sonicwall is acting as the DHCP server for all subnets and I did not create a DHCP server on the sipx server. There are currently 4 subnets defined on the sonicwall and DHCP is working fine for all of them. I've checked ifcfg-eth0 and I've run netstat -r on the sipx server and everything is correct. netstat -r produces Kernel IP routing table Destination Gateway Genmask Flags MSS Window irtt Iface 192.168.25.0 * 255.255.255.0 U 0 0 0 eth0 169.254.0.0 * 255.255.0.0 U 0 0 0 eth0 default 192.168.25.101 0.0.0.0 UG 0 0 0 eth0 192.168.25.101 is the correct gateway running DHCP assigning dynamic IP in the range of 192.168.25.102 to 192.168.25.254. As you can see below, the server is assigned an IP in the static range. ifcfg-eth0: DEVICE=eth0 BOOTPROTO=none DHCPCLASS= HWADDR=(MAC address removed to protect the innocent) ONBOOT=YES IPADDR=192.168.25.11 NETMASK=255.255.255.0 BROADCAST=192.168.25.255 NETWORK=192.168.25.0 Anyone know what's going on? Stiles Dave Redmore wrote: > Yes, it seems like the core issue is the SonicWall doing "something" > to the SIP traffic beyond simple NAT. The audio issue seems like > possibly some strange Codec implementation more than network issues? > > What Codec is being used for these calls? I did notice that Teliax > will connect calls to the PSTN with G722, which seems odd, but did work. > > Dave > > ----- Original Message ----- > From: "Tony Graziano" <tgrazi...@myitdepartment.net> > To: "Discussion list for users of sipXecs software" > <sipx-users@list.sipfoundry.org> > Sent: Wednesday, September 8, 2010 1:32:52 PM GMT -06:00 US/Canada Central > Subject: Re: [sipx-users] Call drops after 1 min & 29 secs > > If that is the case, then his firewall is either doing SPI or ALG to > show that, and it "should not" be involved even at the call status level. > > On Wed, Sep 8, 2010 at 2:24 PM, Dave Redmore > <dave.redm...@spigotsystems.com > <mailto:dave.redm...@spigotsystems.com>> wrote: > > Just to clarify - I think you are each talking about different > things - I THINK Stiles is saying that Int-to-Int calls (Extension > to Extension) do not show up in the SonicWall "call status logs" - > not the sixX CDR records??? This would be logical as internal > calls would never hit the firewall. > > 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: Wednesday, September 8, 2010 1:12:50 PM GMT -06:00 US/Canada > Central > Subject: Re: [sipx-users] Call drops after 1 min & 29 secs > > That's screwed up. I've never seen that in a local configuration. > If the call was answered (either by voicemail or the other > extension) it should show in the CDR. RINGING with no answer does > not show in CDR. > > Makes me wonder what kind of configuration the phones have. Are > there any filters on a firewall for the local ethernet subnet > anywhere? > > On Wed, Sep 8, 2010 at 2:06 PM, Stiles Watson > <wat...@datatek-net.com <mailto:wat...@datatek-net.com>> wrote: > > 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 > <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: Tony Graziano <tgrazi...@myitdepartment.net > <mailto:tgrazi...@myitdepartment.net>> > > To: 'sipx-users@list.sipfoundry.org > <mailto:sipx-users@list.sipfoundry.org>' > <sipx-users@list.sipfoundry.org > <mailto: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 > <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: 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> > >> <mailto: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> > >>>> <mailto: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> > >>>> <mailto: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> <mailto: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> > <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> > >>>>> <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>> > >>>>> <mailto:tgrazi...@myitdepartment.net > <mailto:tgrazi...@myitdepartment.net>> > >>>>> To: sipx-users@list.sipfoundry.org > <mailto:sipx-users@list.sipfoundry.org> > >>>>> <mailto: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> > >>>>> <mailto: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> > >>>>> > <mailto: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>> > >>>>> > <mailto: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>> > >>>>> <mailto: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>> > >>>>> <mailto: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>> > >>>>> <mailto: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> <mailto: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> > >>>>> > >>>>> > <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>> > >>>>> [mailto: > >>>>> 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> > >>>>> <mailto: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> > >>>>> <mailto: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> > >>>>> <mailto: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> > >>>>> <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> > >>>>> <mailto: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> > >>>>> > <mailto:tgrazi...@voice.myitdepartment.net > <mailto:tgrazi...@voice.myitdepartment.net>> > >>>>> Fax: 434.984.8431 > >>>>> > >>>>> Email: tgrazi...@myitdepartment.net > <mailto:tgrazi...@myitdepartment.net> > >>>>> <mailto: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> > >>>>> > <mailto: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> > >>>>> <mailto: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> > >>>>> <mailto: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> > >>>>> > <mailto:tgrazi...@voice.myitdepartment.net > <mailto:tgrazi...@voice.myitdepartment.net>> > >>>>> Fax: 434.984.8431 > >>>>> > >>>>> Email: tgrazi...@myitdepartment.net > <mailto:tgrazi...@myitdepartment.net> > >>>>> <mailto: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> > >>>>> > <mailto: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> > >>>>> > <mailto:tgrazi...@voice.myitdepartment.net > <mailto:tgrazi...@voice.myitdepartment.net>> > >>>>> Fax: 434.984.8431 > >>>>> > >>>>> Email: tgrazi...@myitdepartment.net > <mailto:tgrazi...@myitdepartment.net> > >>>>> <mailto: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> > >>>>> > <mailto: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> > >>>>> <mailto: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> > >>>>> > <mailto:tgrazi...@voice.myitdepartment.net > <mailto:tgrazi...@voice.myitdepartment.net>> > >>>>> Fax: 434.984.8431 > >>>>> > >>>>> Email: tgrazi...@myitdepartment.net > <mailto:tgrazi...@myitdepartment.net> > >>>>> <mailto: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> > >>>>> > <mailto: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> > >>>>> <mailto: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> > >>>>> <mailto: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> > >>>>> <mailto: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> > >>>> <mailto: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> > >>>> <mailto:tgrazi...@voice.myitdepartment.net > <mailto:tgrazi...@voice.myitdepartment.net>> > >>>> Fax: 434.984.8431 > >>>> > >>>> Email: tgrazi...@myitdepartment.net > <mailto:tgrazi...@myitdepartment.net> > >>>> <mailto: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> > >>>> <mailto: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> > >>>> <mailto:tgrazi...@voice.myitdepartment.net > <mailto:tgrazi...@voice.myitdepartment.net>> > >>>> Fax: 434.984.8431 > >>>> > >>>> Email: tgrazi...@myitdepartment.net > <mailto:tgrazi...@myitdepartment.net> > >>>> <mailto: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> > >>>> <mailto: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> > >>>> <mailto:tgrazi...@voice.myitdepartment.net > <mailto:tgrazi...@voice.myitdepartment.net>> > >>>> Fax: 434.984.8431 > >>>> > >>>> Email: tgrazi...@myitdepartment.net > <mailto:tgrazi...@myitdepartment.net> > >>>> <mailto: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> > >>>> <mailto: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> > >>>> <mailto: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> > >>> <mailto: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> > <mailto: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> > >> <mailto:tgrazi...@voice.myitdepartment.net > <mailto:tgrazi...@voice.myitdepartment.net>> > >> Fax: 434.984.8431 > >> > >> Email: tgrazi...@myitdepartment.net > <mailto:tgrazi...@myitdepartment.net> > <mailto: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> > >> <mailto: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. > > > _______________________________________________ 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/
