Differences between current version and revision by previous author of AsteriskSipPhoneSetup.
Other diffs: Previous Major Revision, Previous Revision, or view the Annotated Edit History
Newer page: | version 7 | Last edited on Saturday, March 12, 2005 9:58:33 am | by AdamWentworth | |
Older page: | version 6 | Last edited on Tuesday, November 23, 2004 4:54:19 pm | by JohnMcPherson | Revert |
@@ -78,5 +78,6 @@
nat=yes
qualify=no
reinvite=no
canreinvite=no
+fromdomain=proxy01.sipphone.com ; I was unable to dial out to PSTN using SIP Minutes without this line, Failed INVITE :)
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