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...ies in /tmp Product: Portable OpenSSH Version: -current Platform: ix86 OS/Version: Linux Status: NEW Severity: normal Priority: P2 Component: sshd AssignedTo: openssh-unix-dev at mindrot.org ReportedBy: robertlaferla at attbi.com ------- You are receiving this mail because: ------- You are the assignee for the bug, or are watching the assignee.

http://bugzilla.mindrot.org/show_bug.cgi?id=264 ------- Additional Comments From robertlaferla at attbi.com 2002-06-05 07:07 ------- sshd leaves files of the form "ssh-XX*" in /tmp. it should clean these up on shutdown or startup. ------- You are receiving this mail because: ------- You are the assignee for the bug, or are watching the assignee.

I have encountered the following problem with the latest Asterisk source (as of 4/23/2006): Someone calls me on my PSTN line, it then dials my analog extension (I have both SIP and analog phones where all analog phones are a shared extension.) After a while, I get a busy signal. How can I further diagnose this? What could be the problem?

I have been experiencing a problem where after someone calls me from an analog line, the phone call is terminated after a period of time (anywhere from 15 seconds to 15 minutes) The phone that I use to answer the call is an Aastra 9133i SIP phone. There are several other SIP extensions on the network as well as a few analog extensions on a shared FXS line. When a call comes in the

The volume/traffic on this list has been getting pretty heavy. I find it hard to follow certain discussions and there are some that I am not interested in. Perhaps, we could split the list into two: One for discussing hardware (client phones and cards) and one for the software (configuration, problems, etc...) Or some other better scheme that someone can propose.

I'm curious why the number of jobs out there requiring Asterisk seems to be pretty low. After looking around dice, monster, careerbuilder etc, I was surprised to find no more than 3-4 employment opportunities with Asterisk throughout the US. Is it really that low? There seems to be a job of opportunities for Cisco and other vendors solutions (duh... GUI's are good... duh). I wonder if

Is there some way * can trim the trailing silence in a voicemail message? There's the "maxsilence" setting for silence detection which is related to what I'm asking but not the same. Let's say I set the maxsilence to 8 seconds. During the recording of a voicemail, if someone doesn't say anything for 8 seconds, the recording ends. However, the recording still has

Anyone using the AlarmReceiver? Does it work? Mine doesn't seem to communicate properly. How can I tweak the DTMF settings? Is it in the zaptel.conf or somewhere else?? -- Executing AlarmReceiver("Zap/1-1", "") in new stack > AlarmReceiver: Setting read and write formats to ULAW > AlarmReceiver: Answering channel > AlarmReceiver:

I have been trying to adjust the gain as per this document without any success: http://lists.digium.com/pipermail/asterisk-users/2004-November/071301.html I have a PSTN and VoIP (SIP) connection via *. I disabled all echo cancel/training in zapata.conf and set tx/rxgain to 0. I then changed my extensions.conf so that when I call the VoIP line from the PSTN line, it plays the Milliwatt

If you have many old voicemail messages, to get to the most recent one, you have to keep hitting "6" until you reach the last one. It would be better if you could hit "4" from the first message to get to the last message and/or have a digit that takes you the first and last messages respectively. Anyone have any patches for this?

I am looking for docs on how to diagnose and adjust the rx/tx gain in zapata.conf. The wiki has a link to this article but it no longer exists on the server. http://lists.digium.com/pipermail/asterisk-users/2004-November/071301.html

It seems like asterisk-addons in SVN has been broken for the last few weeks: gcc -DHAVE_CONFIG_H -I. -I. -I. -I./ooh323c/src -I./ooh323c/src/h323 - DGNU -D_GNU_SOURCE -D_REENTRANT -D_COMPACT -c src/chan_h323.c -MT chan_h323.lo -MD -MP -MF .deps/chan_h323.TPlo -fPIC -DPIC -o .libs/ chan_h323.lo src/chan_h323.c: In function 'ooh323_new': src/chan_h323.c:250: error: too few arguments

I cannot access my voicemail and get the following warning in my console: [Nov 25 10:26:43] WARNING[5628]: app.c:935 ast_lock_path: Failed to lock path '/var/spool/asterisk/voicemail/default/8900/Old': File exists I have also noticed that Asterisk will crash several minutes later after this warning message. I am using the latest SVN 1.4 branch of Asterisk (Revision 48007) and

There was a stale lock file in the mailbox directory. This is a bug though. Asterisk should clean up all lock files on startup. Lastly, I can't explain the intermittent crash and wasn't able to catch it using gdb either.

If you own Aastra phones, here's a group dedicated to your specific needs. BTW - The Asterisk-users mailing list is great but it has way too much volume to be useful for specific problems. It needs to be broken up into smaller more manageable lists. Homepage: http://groups.google.com/group/aastra-asterisk-users Group email: aastra-asterisk-users@googlegroups.com

Is there a zaptel mailing list? Here's the error: CC [M] zaptel-1.4/xpp/xbus-core.o zaptel-1.4/xpp/xbus-core.c: In function ?debugfs_open?: zaptel-1.4/xpp/xbus-core.c:171: error: ?struct inode? has no member named ?u?

Please explain the relationship between modules from the driver (wctdm), the /etc/zaptel.conf file and zapata.conf. Specifically, if I have a FXS module 0 and FXO module 1, what should be used in zaptel.conf and what should be used in zapata.conf? Then finally, in extensions.conf, what is the Zap channel for dialing out? Zap/? % dmesg Module 0: Installed -- AUTO FXS/DPO Module 1:

After so many rings when the party does not answer, my SIP phone says Call Failed. Why doesn't it just keep ringing? Here's the dial plan rule: exten => _NXXXXXXXXX,1,Dial(SIP/${EXTEN}@sip.myprovider.com,,r) exten => _NXXXXXXXXX,n,Hangup()

Using the latest SVN of zaptel and asterisk, I can no longer receive incoming analog calls. The caller just hears it ringing but Asterisk doesn't pick up. I am seeing these error messages: [Apr 9 09:38:02] WARNING[16564]: chan_zap.c:6470 ss_thread: CallerID returned with error on channel 'Zap/2-1' [Apr 9 09:38:50] WARNING[16570]: chan_zap.c:6470 ss_thread: CallerID