Differences between version 43 and revision by previous author of SIPDirectory.
Other diffs: Previous Major Revision, Previous Revision, or view the Annotated Edit History
| Newer page: | version 43 | Last edited on Monday, July 14, 2008 2:28:28 pm | by BruceKingsbury | Revert |
| Older page: | version 42 | Last edited on Monday, October 30, 2006 10:52:34 am | by CraigMckenna | Revert |
@@ -1,6 +1,7 @@
PLEASE - if you ring the people on this list the time difference may mean they are asleep.. so dont expect to get an answer.. but please leave a message saying you are ringing to test your configuration and you are following the instructions on the wiki.. The more people that leave messages the more likely we will improve the information here. and it is more likely we will return the call or leave a message for you.
+<?plugin OldStyleTable
<?plugin OldStyleTable
| Name | [FWD] (1-393)| sipphone.com (1-747) [1] | MetaNet [2] | IAXTEL | Notes
| KyleCarter | 233007 | 3863351 | | | [http://fwd.pulver.com/myicon.php?userid=233007&.jpg]
| MattBrown | 239981 | 3863374 | | | [http://fwd.pulver.com/myicon.php?userid=239981&.jpg]
@@ -9,17 +10,18 @@
| AndrewMitchell | 248024 | 2443392 | | | [http://fwd.pulver.com/myicon.php?userid=248024&.jpg]
| JohnMcPherson | 242260 | | john@10.21.1.100 | | currently investigating NAT issues with fwd
| LindsayDruett%%% Line 1 | 266015 | | | | [http://fwd.pulver.com/myicon.php?userid=266015&.jpg]%%%[Cisco ATA-186|http://www.lindsay.wired.net.nz/projects/sip/]
| LindsayDruett%%% Line 2 | 433606 | | | | [http://fwd.pulver.com/myicon.php?userid=433606&.jpg]%%%[Cisco ATA-186|http://www.lindsay.wired.net.nz/projects/sip/]
-| BruceKingsbury | | 6183397 | | | ATA-186 line
-| CairoKingsbury | | 2445132 | | | ATA-186 line 1
-| VernePavreal | | 6265272 | | | Using Asterisk and GrandStream Budgetone
+| BruceKingsbury | | 6183397 | | |
+| CairoKingsbury | | 2445132 | | |
+| VernePavreal | | 6265272 | | | Using Asterisk and Linksys ATA
| AaronMartin (Work) | 42325| | | | [http://fwd.pulver.com/myicon.php?userid=42325&.jpg] Using Asterisk with 12x Grandstream Phones
| AaronMartin (Home) | 70362 | | | | [http://fwd.pulver.com/myicon.php?userid=70362&.jpg]
| DanielLawson | 271486 | 6694095 | | (700) 423-6888 | [http://fwd.pulver.com/myicon.php?userid=271486&.jpg]
| PerryLorier | 284789 | | | | [http://fwd.pulver.com/myicon.php?userid=284789&.jpg]
| ReneBartosh | 393754 | | | | [http://fwd.pulver.com/myicon.php?userid=393754&.jpg]Currently using asterisk with a X100P, a PAP2 and italk.
?>
-[1] on sipphone.com you can dial ** to hear your own number
+?>
+#
[|ftnt_
1]~[[1|#ftnt_ref_1]~]
on sipphone.com you can dial ** to hear your own number
-[2] you will probably need to make sure you aren't connecting via a proxy (or registrar) such a fwd.pulver.com:
+#
[|ftnt_
2]~[[2|#ftnt_ref_2]~]
you will probably need to make sure you aren't connecting via a proxy (or registrar) such a fwd.pulver.com:
