similar to: All extensions now cannot loggin!!!!

Hi, I'm running *@home 0.5 exactly as it comes off the .iso image and have configured an extension (206) using AMP for a grandstream 102. Have checked sip_additional.conf looks ok, but I can't get the incoming cli on the 102 to read anything other than 't ri', apparently from other posts the phone is trying to display 'asterisk'! I've got a couple of pc's

If you define a sip peer, wheather or not you put an entry in the field OUTBOUND CID, if you dial an external extension (let's say an extension on another asterisk server, connected via IAX2 connection) the callerid received by the foreign asterisk is device <YOURNUMBER>: i.e device <567> If you take a look at etc/asterisk/sip_additional.conf, you can see under the SIP extension

Download V 0.4 here http://sourceforge.net/project/showfiles.php?group_id=123387 burn it to an .iso install into asterisk box (be warned it deletes everything on the hard drive but this is what you want right :) it will automatically install Asterisk AMP FOP and Web Meetme read the FAQ here http://asteriskathome.sourceforge.net/faq.html basically if you are using a X100P all you need to do

Wiley, There are a couple of issues that we saw while not using this option. 1) sip authentication failures as Asterisk is not able to reach Polycom phones. A typical problem description is here: http://lists.digium.com/pipermail/asterisk-users/2004-December/079251.ht ml 2) DTMF issues for Transfers, Hold or simply to dial extensions. This problem is more pronounced when you are using

_____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of khalid touati Sent: Tuesday, April 13, 2010 1:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Time variables in system application Hi Guys, i have a weird thing here: when using time variables (%F & %T) in a shell script, out

Hi, I am simulating the sending of fax using sendfax through voip to reach an Asteria server via ZAP/1 ( PSTN phone line ) which then route call to a fax machine at ZAP/2. It seems like I am not able to establish any handshake with the physical fax machine using the sendfax program. Does anyone know why that happens and how to fix it? The scenario will be deployed in remote location in the

Egad, not again with Broadvoice! Anyhow, I recently installed AAH and configured my TDM11B and got that and some SIP phones working. I still have some issues to work out, etc, but my current problem is Broadvoice. I have checked out all of the online resources, including the recent list exchange about the recent changes made by Broadvoice. However, the one thing I have found to be consitent in

Hi All, I'm trying to connect to a SIP carrier who never connected with Asterisk. I managed to connect with a sipura phone or a grandstream, no problem. When I configure asterisk, I'm able to send out calls to the carrier no problems, however, receiving calls doesn't work, and I keep getting the following messages: <-- SIP read from 69.xx.xx.xx:5060: INVITE

Here's something that should be easy for RUBY pro's. Here is a script: 1.times do r = $agi.exec('DIAL', SIP/voipuser&Zap/32&Zap/33&Zap/34&Zap/35) r = $agi.get_variable('DIALSTATUS') # $agi.set_variable(' WHOANSWERED

Hi All, Asterisk 1.4.12 on CentOS 5 Yesterday and today I got the following warnings in /var/log/asterisk/messages: WARNING[2066] chan_sip.c: Got authentication request (401) on unknown REGISTER to '<sip:account at sip.voipuser.org>;tag=d8f15e1f30efddd35168b07dba9d540e.3922' The corresponding bits in sip.conf are: register =>

Hi, I have a Samba 3 PDC How do you restrict users from loggin in on more than one machine at the same time? Thanks --Siju

Is there a way to have rails log the sql statement that it runs. It would be much easier for me to debug my code if I could actually see the sql that active record is using. Charlie recentrambles.com -> use Time not DateTime -------------- next part -------------- An HTML attachment was scrubbed... URL:

Sorry I don't think my first trial worked I am just starting to use samba, and I keep running into the same problem time and again. I have installed SAMBA 2.0.2 correctly and have (i think) a correct smb.conf file. Now when I try to log on to one of my shares it will not let me in. I get the following errors when logging in: [dutchboy@lisa dutchboy]$ smbclient \\\\lisa\\dutchboy doing

This is a follow-up on a thread from last year requesting that openssh indicate which authorized key was accepted during a login as opposed to just logging that a key was accepted... Here's the old message: It is possible for ~user/.ssh/authorized_keys to have multiple entries. It would be quite helpful if openssh would enhance the log to indicate WHICH key was

It is possible to make dovecot do not log any request from a determined host? Because, we gonna use ldirectord for foward request and etc. He makes an connection every 2 sec i guess, and dovecot logs its. The logs get too many info that we don need! Any ideias ? Thanks! * *[]'sf.rique

I''ve been in and out of everything I can find, and can''t find a way to make the logger stop adding ANSI code to the log. Is there a logger.cant_see_white_on_gray method? :-) TIA! Regards, JJ

Hi all, We are using ssh for file transfer through SCP and SFTP in a FreeBSD box, and it works fine as expected. But from some days ago the customers are requesting the logs of the transmissions, and we?ve just realized that the sshd daemon doesn?t log the files copied to our server nor the downloaded files. I?ve tried with all the debug levels of the sshd daemon, but nothing. It logs a

Hi! I am switching a mail server setup to a new machine which has RHL Fedore Core 2 installed, which uses Dovecot. I managed to get Squirrelmail to work on my personal account, but when logging in with some other accounts I get the following error as response: ERROR : Connection dropped by imap-server. Query: Capability I did a web search on this and found a message from Tim Middleton who

JID Pri S Owner Number Pages Dials TTS Status 58 123 S root 008675533661 0:2 4:12 02:12 No carrier detected Here is the asterisk output: [Mar 28 01:54:00] NOTICE[16753]: chan_iax2.c:6025 update_registry: Restricting registration for peer 'iaxmodem' to 60 seconds (requested 50) -- ast_get_srv: SRV lookup for '_sip._udp.voipuser.org' mapped to host