similar to: Verbose setting changed?

Can anyone recommend some phone sets that are adsi compliant and work well with asterisk?

Hi guys, I'm having a problem rsyncing between two directories on local disk. I wish to sync the local directory /local/CA/eTrustDirectory/dxserver/ldif to the other local directory /local/CA/archive. 'rsync -vvvr --progress /local/CA/eTrustDirectory/dxserver/ldif /local/CA/archive' pauses at: make_file(4,ldif/MELLDRBIB11-ldsau004wm1/ldbau004v1_20060330.0204.raw)

d# rsync -vcrlpogtz . rsync@domain.ltd:/export/home/rsync/ rsync@domain.ltd's password: sh: rsync: rsync: connection unexpectedly closed (0 bytes read so far) rsync error: error in rsync protocol data stream (code 12) at io.c(189) d# Can someone tell me what am I doing wrong? Thanks

Hi, I am using asterisk 1.6.0.10 For debugging i set verbosity to 10 with asterisk -vvvvvvvvvvr.. now i am trying to set it lower but.. when i type asterisk -r it starts with Connected to Asterisk 1.6.0.10 currently running on asterisk1 (pid = 2408) Verbosity is at least 10 when i try set verobisty 1 or similar commands.. i think this command is obselete in 1.6 .. set verbose 1 No such command

Hello. Well I finally rx'd my dev kit (new batch of TDM's apparently. I'm on Mandrake 9.1 There were no hardware install instructions, it would have been nice to know whether the 4-way power connector was to be used or was for some other future or expansion purpose. It came with a floppy disc, no label and it wasn't even write protected. The only readme file was

Unless you suppress all output with the -q option, rsync will always print an initial "building/receiving file list ... done" line. In my opinion, this is a bit superfluous. When I want to see a progress indication, I can use the --progress option. Another issue is the 3-line transfer speed and speedup factor report at the end. So every rsync invocation produces at least four lines of

Does anybody else have the feeling that custom messages (Verbose(1,...)) do not stand out enough on the CLI? We're sending messages like "Extension 123 is unknown" to the output and that should tell the user why a call to 123 fails but users fre- quently crank up the verbosity to 3 or 10 so our messages go unnoticed in many cases. My idea was to use terminal escape sequences to make

Hi all, after upgrading from 13.7 to 14.2, asterisk cli (asterisk -r) don't show what's happens. I've trying setting debug and verbose to 100 but nothing, no show. All commands works as expected but i can't what's happens on my asterisk server. asterisk*CLI> core show settings PBX Core settings ----------------- Version: 14.2.0 Build Options:

Hello all, During discussion of https://github.com/networkupstools/nut/issues/1782 me and Greg uncovered a diametral difference of opinion about the verbosity of NUT programs in general, and of `upsmon -K` (checking for POWERDOWNFLAG during shutdown integration) in particular. To me as a sysadmin type often (in the past at least) having to troubleshoot the tails and ends of systems'

Hello all, During discussion of https://github.com/networkupstools/nut/issues/1782 me and Greg uncovered a diametral difference of opinion about the verbosity of NUT programs in general, and of `upsmon -K` (checking for POWERDOWNFLAG during shutdown integration) in particular. To me as a sysadmin type often (in the past at least) having to troubleshoot the tails and ends of systems'

I have a situation where our VOIP provider is running *, my office is running *, and my house is running *. I have an extension at the office so that if a call comes in from the VOIP provider and they select that extension, the call will be sent to my home * box and ring my phone. That works fine. I set "notransfer=no" in the iax.conf file at the office so that the office system can

Hi folks, How I can change default level in asterisk from 3 level to 7level, using the script /etc/init.d/asterisk

I had a similar problem yesterday. I would go to the root of my site and I would get a Too Many Redirects message. It seems like there was an infinite loop. After struggling for over an hour yesterday late at night, it seemed like I fixed it. But now, when I want to create a New User, it''s redirecting me to the Sign In screen. Seems like I didn''t fix the problem completely. I

I have a pjsip install that is not pulling it's realtime registrations from an ODBC database. When I have them directly pull from a MySQL database everything seems to work. I am having trouble finding a requirements document for the pjsip realtime registrations. Is there some kind of special config that has to be used to trigger the connection for realtime registrations over ODBC?

Hi all. I am actually using ActiveRecord outside of Rails but I don''t see that it matters for this question. I want to automatically create a registration entry when I add a new user to my application. This works fine, but it''s not automatic: class User < ActiveRecord :: Base has_many :registrations end class Test user = User.new user.registrations.create end

Hi all, for an unknown reason, I find my asterisk server down every morning as a result of failing to restart during the night because of a segmentation fault. The error message is as follows: Waiting for inactivity to perform restart... Executing last minute cleanups == Destroying any remaining musiconhold processes Asterisk cleanly ending (0). -- Remote UNIX connection Waiting for

Hi all, i made a test scenario with two windoze machines: On the first one callgen323 is running in listening mode On the second one, callgen323 strarting 25 calls to the asterisk pbx, and the asterisk calls the first windoze machine. But after the second one make a few calls (mostly after 11 - 14) asterisk crashes with the only message : Segmentation fault. Are this to many calls for oh323

-----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Joshua Colp Sent: Sunday, October 04, 2015 12:44 PM To: asterisk-users at lists.digium.com Subject: Re: [asterisk-users] pjsip realtime registrations not pulling from ODBC On 15-10-04 01:42 PM, Bryant Zimmerman wrote: >

On 15-10-04 09:54 AM, Bryant Zimmerman wrote: > I have a pjsip install that is not pulling it's realtime registrations > from an ODBC database. > When I have them directly pull from a MySQL database everything seems to > work. > I am having trouble finding a requirements document for the pjsip > realtime registrations. > Is there some kind of special config that has to be

Hi, There is any possibility to dial a specific extension and then enter in your own mailbox (the one defined for that specific SIP phone) without asking for the exxtension number but only for the password? I want to be the same extension for all phones, not a specific one for each of them. It is possible to have a time stamp in the recorded message? I want to know when the message has been