From 3a3a254bec9a447efaa988687b9890baab6a7426 Mon Sep 17 00:00:00 2001 From: 4-FLOSS-Free-Libre-Open-Source-Software <46166740+4-FLOSS-Free-Libre-Open-Source-Software@users.noreply.github.com> Date: Tue, 21 Jan 2020 14:02:59 +0100 Subject: let gui set MAX_PTIME cutoff 80ms --- src/gui/userprofileform.ui | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'src/gui/userprofileform.ui') diff --git a/src/gui/userprofileform.ui b/src/gui/userprofileform.ui index 4a11efa..f3683b8 100644 --- a/src/gui/userprofileform.ui +++ b/src/gui/userprofileform.ui @@ -823,7 +823,7 @@ This field is mandatory. 10 - 50 + 80 10 -- cgit v1.2.3 From 30eac220c69dabd715d6357d2df2f39b91b08fe2 Mon Sep 17 00:00:00 2001 From: Jose Riha Date: Thu, 20 Feb 2020 14:20:13 +0100 Subject: Fix spelling errors --- src/gui/userprofileform.ui | 26 +++++++++++++------------- 1 file changed, 13 insertions(+), 13 deletions(-) (limited to 'src/gui/userprofileform.ui') diff --git a/src/gui/userprofileform.ui b/src/gui/userprofileform.ui index 4a11efa..ef2f900 100644 --- a/src/gui/userprofileform.ui +++ b/src/gui/userprofileform.ui @@ -2494,7 +2494,7 @@ This format is what most SIP phones use. - Indicates if the Replaces-extenstion is supported. + Indicates if the Replaces-extension is supported. Replaces @@ -2585,7 +2585,7 @@ This format is what most SIP phones use. - An attended call transfer should use the contact URI as a refer target. A contact URI may not be globally routable however. Alternatively the AoR (Address of Record) may be used. A disadvantage is that the AoR may route to multiple endpoints in case of forking whereas the contact URI routes to a single endoint. + An attended call transfer should use the contact URI as a refer target. A contact URI may not be globally routable however. Alternatively the AoR (Address of Record) may be used. A disadvantage is that the AoR may route to multiple endpoints in case of forking whereas the contact URI routes to a single endpoint. Attended refer to AoR (Address of Record) @@ -3079,7 +3079,7 @@ When you choose this option you have to create static address mappings in your N <p> -Often the format of the telphone numbers you need to dial is different from the format of the telephone numbers stored in your address book, e.g. your numbers start with a +-symbol followed by a country code, but your provider expects '00' instead of the '+', or you are at the office and all your numbers need to be prefixed with a '9' to access an outside line. Here you can specify number format conversion using Perl style regular expressions and format strings. +Often the format of the telephone numbers you need to dial is different from the format of the telephone numbers stored in your address book, e.g. your numbers start with a +-symbol followed by a country code, but your provider expects '00' instead of the '+', or you are at the office and all your numbers need to be prefixed with a '9' to access an outside line. Here you can specify number format conversion using Perl style regular expressions and format strings. </p> <p> For each number you dial, Twinkle will try to find a match in the list of match expressions. For the first match it finds, the number will be replaced with the format string. If no match is found, the number stays unchanged. @@ -4076,7 +4076,7 @@ The values of all SIP headers of the outgoing INVITE are passed in environment v - When ZRTP/SRTP is enabled, then Twinkle will try to encrypt the audio of each call you originate or receive. Encryption will only succeed if the remote party has ZRTP/SRTP support enabled. If the remote party does not support ZRTP/SRTP, then the audio channel will stay unecrypted. + When ZRTP/SRTP is enabled, then Twinkle will try to encrypt the audio of each call you originate or receive. Encryption will only succeed if the remote party has ZRTP/SRTP support enabled. If the remote party does not support ZRTP/SRTP, then the audio channel will stay unencrypted. &Enable ZRTP/SRTP encryption @@ -4206,23 +4206,23 @@ The values of all SIP headers of the outgoing INVITE are passed in environment v <p> If your provider offers the message waiting indication service, then Twinkle can show you when new voice mail messages are waiting. Ask your provider which type of message waiting indication is offered. </p> -<H3>Unsollicited</H3> +<H3>Unsolicited</H3> <p> -Asterisk provides unsollicited message waiting indication. +Asterisk provides unsolicited message waiting indication. </p> -<H3>Sollicited</H3> +<H3>Solicited</H3> <p> -Sollicited message waiting indication as specified by RFC 3842. +Solicited message waiting indication as specified by RFC 3842. </p> - Unsollicited + Unsolicited - Sollicited + Solicited @@ -4261,9 +4261,9 @@ Sollicited message waiting indication as specified by RFC 3842. - + - Sollicited MWI + Solicited MWI @@ -4328,7 +4328,7 @@ Sollicited message waiting indication as specified by RFC 3842. - For sollicited MWI, an endpoint subscribes to the message status for a limited duration. Just before the duration expires, the endpoint should refresh the subscription. + For solicited MWI, an endpoint subscribes to the message status for a limited duration. Just before the duration expires, the endpoint should refresh the subscription. 999999 -- cgit v1.2.3