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...//bugs.launchpad.net/ubuntu/+source/openssh/+bug /663455 OS/Version: All Status: NEW Severity: minor Priority: P2 Component: ssh AssignedTo: unassigned-bugs at mindrot.org ReportedBy: cjwatson at debian.org David Lublink reported the following as an Ubuntu bug: int key_perm_ok(int fd, const char *filename) { [...] error("Permissions 0%3.3o for '%s' are too open.", (u_int)st.st_mode & 0777, filename); error("It is recommended that your private key files a...

Hey, I have a voip provider that uses mgcp and I would like to connect that provider to my asterisk. Anyone succeed in doing this? Thanks, David -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060218/aea2042a/attachment.htm

Hi, Consider the following situation : <SIP/asterisk-0000001d>AGI Rx << WAIT FOR DIGIT 3000 <SIP/asterisk-0000001d>AGI Tx >> 200 result=48 <SIP/asterisk-0000001d>AGI Rx << WAIT FOR DIGIT 3000 <SIP/asterisk-0000001d>AGI Tx >> 200 result=48 <SIP/asterisk-0000001d>AGI Rx << WAIT FOR DIGIT 3000 <SIP/asterisk-0000001d>AGI Tx >>

Hello, I am searching for a DTMF issue on my setup ( 2 years and counting ), and I am wondering why rtp.c has code to mute DTMF ( the rtp->dtmfmute variable ), but this same mechanism does not exist in dahdi. I am sending a DTMF over SIP w/ RTP & RFC2833 to the asterisk box with the dahdi card. The dahdi card sends it out on the PRI line. Trouble is, the DTMF is echoed back and the

Hello, Here is my setup : Telephone 1 ( GXP 2000 ) Telephone 2 ( SPA942 ) Asterisk 1.4.17 ( same behaviour on Asterisk 1.4.23.1 ) Scenario: I don't like the default asterisk transfer feature, so I am trying to write my own. What I did : 1. Added to dynamic features #3 with AGI pointing to my php script 2. PHP script asks the user to enter his/her extension 3. PHP connects to Asterisk

Hello, I installed Asterisk 1.6.2.17.3 ( latest as of yesterday ) and had multiple problems with DTMF. I have two machines, we'll call them asterisk and asterisk-pri. Asterisk does IVR and asterisk-pri has a PRI card in it and connects to the PSTN. The two servers communicate via SIP with RFC2833. I setup logger.conf on both machines to display DTMF to the console. Both are built from

Hello, I am running asterisk 1.4. For argument's sake I have 4 telephones. 2 support video, 2 do not. Calls between phones work fine and codecs are properly negociated. I have videosupport=yes in sip.conf and when the two video phones communicate I have video. I call meet me with this command EXEC MEETME 1234|d SIP looks like this : -- AGI Script Executing Application: (MeetMe)