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...mods to the openssh-3.0p1 source to do this. The option name I used (for the ~/.ssh/config file) is: DisplayCommentString [yes|no] Default: no You can choose whatever name for this option you think is appropriate. Will you consider adding this prompting option? thanks, Jim Brucker <jim at buu.ac.th> Modifications: ============== Here are explanations of the changes and diffs. The "BUCC" comments are used by me -- feel free to delete them. ============== readconf.h - add (int) display_comment_str to the Options struct. diff: 104a105,106 > > int display_c...

...Gradinari) * ASTERISK-28107 - app_confbridge: Participant info labels aren't being added to the SDPs (Reported by George Joseph) * ASTERISK-28089 - function ast_sendtext() create RTP realtime packets with a trailing null byte in the payload (Reported by Emmanuel BUU) * ASTERISK-28076 - bridging: Asterisk crashes when receiving an empty realtime text frame (Reported by Emmanuel BUU) * ASTERISK-28084 - app_queue: QueueMemberStatus Event flooding AMI (Reported by Andrej) * ASTERISK-28077 - res_pjsip: improve realtime performance on...

...Gradinari) * ASTERISK-28107 - app_confbridge: Participant info labels aren't being added to the SDPs (Reported by George Joseph) * ASTERISK-28089 - function ast_sendtext() create RTP realtime packets with a trailing null byte in the payload (Reported by Emmanuel BUU) * ASTERISK-28076 - bridging: Asterisk crashes when receiving an empty realtime text frame (Reported by Emmanuel BUU) * ASTERISK-28084 - app_queue: QueueMemberStatus Event flooding AMI (Reported by Andrej) * ASTERISK-28077 - res_pjsip: improve realtime performance on...

...exei Gradinari) * ASTERISK-27980 - Caller ID cannot be changed on Attended Transfer before dialing out (Reported by Alexei Gradinari) * ASTERISK-28089 - function ast_sendtext() create RTP realtime packets with a trailing null byte in the payload (Reported by Emmanuel BUU) * ASTERISK-28076 - bridging: Asterisk crashes when receiving an empty realtime text frame (Reported by Emmanuel BUU) * ASTERISK-28084 - app_queue: QueueMemberStatus Event flooding AMI (Reported by Andrej) * ASTERISK-28077 - res_pjsip: improve realtime performance on...

...lorian Floimair) * ASTERISK-28011 - chan_sip: get_refer_info() attempted unlock mutex 'peer' without owning it! (Reported by Alec Davis) * ASTERISK-28002 - When T.140 realtime text is negociated, a lot of debug traces are generated (Reported by Emmanuel BUU) * ASTERISK-27973 - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY (Reported by Valentin Safonov) * ASTERISK-28007 - rtcp-mux is put in SDP answer regardless of offer (Reported by Torrey Searle) * ASTERISK-27997 - pjproject_bundled: Fix for Solaris builds. Do...

...ssible without your participation. Thank you! The following issues are resolved in this release: Bugs fixed in this release: ----------------------------------- * ASTERISK-28002 - When T.140 realtime text is negociated, a lot of debug traces are generated (Reported by Emmanuel BUU) * ASTERISK-27881 - PBX calls via chan_sip TCP trunk now get authentification error (Reported by Ian Gilmour) * ASTERISK-28011 - chan_sip: get_refer_info() attempted unlock mutex 'peer' without owning it! (Reported by Alec Davis) * ASTERISK-27944 - res_pjsip_t38:...

...Gradinari) * ASTERISK-28107 - app_confbridge: Participant info labels aren't being added to the SDPs (Reported by George Joseph) * ASTERISK-28089 - function ast_sendtext() create RTP realtime packets with a trailing null byte in the payload (Reported by Emmanuel BUU) * ASTERISK-28076 - bridging: Asterisk crashes when receiving an empty realtime text frame (Reported by Emmanuel BUU) * ASTERISK-28084 - app_queue: QueueMemberStatus Event flooding AMI (Reported by Andrej) * ASTERISK-28077 - res_pjsip: improve realtime performance on...

...SK-28013 - res_http_websocket: Crash when reading HTTP Upgrade requests (Reported by Sean Bright) Bugs fixed in this release: ----------------------------------- * ASTERISK-28076 - bridging: Asterisk crashes when receiving an empty realtime text frame (Reported by Emmanuel BUU) * ASTERISK-28084 - app_queue: QueueMemberStatus Event flooding AMI (Reported by Andrej) * ASTERISK-28077 - res_pjsip: improve realtime performance on CLI 'pjsip show contacts' (Reported by Alexei Gradinari) * ASTERISK-27920 - app_queue: Queue member considered in...

...idge: Participant info labels aren't being added to the SDPs (Reported by George Joseph) - [ASTERISK-28089 <https://issues.asterisk.org/jira/browse/ASTERISK-28089>] - function ast_sendtext() create RTP realtime packets with a trailing null byte in the payload (Reported by Emmanuel BUU) - [ASTERISK-28076 <https://issues.asterisk.org/jira/browse/ASTERISK-28076>] - bridging: Asterisk crashes when receiving an empty realtime text frame (Reported by Emmanuel BUU) - [ASTERISK-28084 <https://issues.asterisk.org/jira/browse/ASTERISK-28084>] - app_queue: Queue...

Hi, I'm trying to create Shared folder with subfolders "SPAM" and "NOT_Spam". Configuration: namespace public { separator = / prefix = Shared/ location = maildir:/vmail/%Ld/Shared:INDEX=/vmail/%Ld/%Ln/Shared hidden = no } Result I get (is exactly what I want): Inbox Drafts Sent Trash Shared --NOT_Spam --SPAM I can select SPAM and NOT_Spam, delete messages etc, but

Hi, I was using Dovecot (POP3,IMAP+quotas) with postfix quota "addon". But the postfix's quotas were reported differently and the quota file gets messed up all the time. So now I have also Dovecot's LDA, and everything works like charm... but... Now postfix can pass only one mail at a time to LDA. And that slows down delivery. Postfix master file: dovecot unix - n

Hello, I have X101P card. But it seems to be dead. Always app_dial.c:803 dial_exec: Unable to create channel of type 'Zap' (cause 0) I've add the line: exten => 999,1,Dial(Zap/1). But calling to 999 show the same error. Zap show channel, lspci etc show everything is normal. Could you tell me pls how can I test the card? (very urgent) Thank you very much. -------------- next part

Hello, I have X100P card but exten => _9X.,1,Dial(Zap/1/${EXTEN:1}) does not work. The error is app_dial.c:803 dial_exec: Unable to create channel of type 'Zap' (cause 0) Zap show channel, lspci etc show everything is normal. Because X100P is Intel 537 modem, so can I use it as true modem for test? Did any one use X100P as a modem? Thank you very much. -------------- next part