similar to: Directory problem

Hello I'm having a difficult time finding precisely what to put in sip.conf and extensions.conf (and possibly other files) to get a working configuration to connect an Asterisk (1.4) server to a VoIP provider with the Asterisk server + SIP clients located in a private LAN behind a NAT router: http://img560.imageshack.us/img560/3749/asterisknat.png Would someone have a full, direct (ie.

Hi, Summary of my problem: Remote X forwarding is apperently randomly impossible for different display numbers. At the end of this mail you will find a recipe for how to reproduce this behaviour easily. I use SuSE 10.2 with the following openssh version: OpenSSH_4.4p1, OpenSSL 0.9.8d 28 Sep 2006 Clients (Linux and Windows (Cygwin)) connect to the server with X-Forwarding enabled

> Hi Gareth Blades & Doug, > > Thanks so much for for the feedback. I have searched on lot of documents > but couldn't able to find clear answer regarding it. > > I hope you guys replies are very much help all in aterisk community. > > > Thanks & Regards, > > Vidura Senadeera, > > Network Engineer, > > Debug Solutions > > Sri Lanka .

Does any body know if it is possible to group SIP channels just like it is possible with Zap channels? I have a group of FXO gateways (Sipura 3000's) and I would like to treat them as a group the same as I would Zap channels. Does anyone know if this is this possible?

Hi list, I enabled the transfer function in my dialplan, but I may configure it wrong because sometime when I call a SIP extension number from one FXS port, the SIP side will hear word "transfer", I hear nothing, after that, the call conversation is fine.I'v had this problem for a long time, could not get clue where I configure it wrong. here is my related config part: sip.conf:

This is on Solaris 10 x86, do not see this behavior on Solaris 10 sparc. Seen on multiple machines. Sshd debug: debug1: server_input_channel_open: ctype session rchan 256 win 16384 max 16384 debug1: input_session_request debug1: channel 0: new [server-session] debug2: session_new: allocate (allocated 0 max 10) debug3: session_unused: session id 0 unused debug1: session_new: session 0 debug1:

With best regards Abdul Ahad Anwer Khan, M.Sc(CME, in progress) University of Applied Sciences Offenburg Germany Phone:+497814748226 Mobile:+4917623468462 From: abdulahadanwer at hotmail.com To: asterisk-users-bounces at lists.digium.com Subject: New in asterisk Date: Sun, 27 Sep 2009 14:50:59 +0600 Hello All I am a student and doing my thesis which is related to asterisk. I am new in

Rowland, I've done almost all permissions change, I forgot bind-dns directory. Now, the named service still doesn't start and journalctl -xe showed me that this occurs because permission denied to run dlz_bind9_9.so. I've checked out and the lib and directory /usr/local/samba/lib/bind9/ have execute permission to named group. The output of ls command, journalctl -xe and

I think I might have a bug here, but I''m not sure. I just cannot seem to win with X11 forwarding and OpenSSH-1.2.2! The shell works fine though. When I had the Linux 2.2.11 kernel running, sshd was saying: debug: Received request for X11 forwarding with auth spoofing. debug: Socket family 10 not supported [X11 disp create] debug: channel 0: new [X11 inet listener] debug: Forking

Sorry for missing the importance of these earlier. These vulnerabilities were first disclosed this January. There are seven vulnerabilities reported in the icoutils package, in the 'wrestool' program. Unfortunately because libguestfs downloads untrusted guest content and processes it with 'wrestool -x' on the host, libguestfs is vulnerable to these. This could lead to host

http://bugzilla.mindrot.org/show_bug.cgi?id=173 Summary: sshd does not listen on tcp-socket, though GatewayPorts yes Product: Portable OpenSSH Version: 3.1p1 Platform: UltraSparc OS/Version: SunOS Status: NEW Severity: normal Priority: P2 Component: sshd AssignedTo:

I've noticed a burst of about 12000 of these mbox-rewrite msgs in the log, spread over about 10 minutes. They are always for the same person (XXX'd out here), and have not occurred before or since. There's no evidence of device problems in the message log. Jan 10 17:46:43 ls1 imap-login: Login: XXX [::ffff:10.2.0.1] Jan 10 17:46:43 ls1 imap(XXX): file mbox-rewrite.c: line 429

Hello to all When my SNOM (300 or 320) loses Internet connectivity, it loses its Asterisk registration (ok, thats normal). But when the phone is back online, he doesn't try to register in Asterisk. I believe this happens to avoid flooding the private LANs when the Internet link is lost.... but the problem is that the phones don't try to re-register in the future.... Sometimes it stays

Hi, I just read a pressrelease from VON that Digium will soon be releaseing a couple of new cards. What got me interested was: "The TE420P and TE415P support 128ms of G.168 (2002)-compliant echo cancellation across their entire 128 channels." Does anyone know when thease will be released and what they will cost when released? Thanks!

Is that a bios setting (I don't recall seeing it) or an OS setting? I run a lot of Via C3 machines (they are so nifty) but don't remember seeing this. -----Original Message----- From: Technical Support [mailto:support@ocg.ca] Sent: Tuesday, September 06, 2005 10:43 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion'; shmaltz@gmail.com Subject: RE: [Asterisk-Users]

With 3.3p1 built on AIX 4.3.2: $ ssh [blah] Couldn't set usrinfo: Not owner debug1: Calling cleanup 0x20019080(0x200219a0) debug3: mm_request_send entering: type 27 debug1: Calling cleanup 0x20018dd4(0x0) Connection to songohan closed by remote host. Connection to songohan closed. Output from sshd -d -d -d: ... debug3: tty_parse_modes: 92 0 debug3: tty_parse_modes: 93 0

Since Monday I didn't see much traffic.

Can anyone recommend a call accounting solution with rating for post paid billing that works well with asterisk using the account code or any other info from the CDR? I don't want the billing software to any phone calls for me, therefore any solution that modifies my extensions.conf is out, nor does it have to allow for customers the ability to log in to check their usage/balances. I have

Hello *, sorry if i missed an article which already solves my problem. I need a working configuration of OpenSSH for Solaris 7 (SunOS 5.7) with AFS support. PAM support to use the AFS PAm module `pam_afs.so' and TCP-Wrapper support would be nice. It would be nice to have similar configurations for Solaris 6 (SunOS 5.6), Solaris 8 (SunOS 5.8) and Solaris 2.5.1. I have tested a few

I have a customer that wants to get switchvox, since I have never used it, I would like to hear some feedback from active users of switchvox. In specific: 1. Does it use realtime or conf files 2. Is it possible to change it manually? 3. Is SSH access to login to console/shell available? 4. Are you or your customers happy with the user interface? TIA