similar to: Question: handling fully-qualified SIP dial requests

Bug in the register= code; see details below from the developer of "ser" (SIP Express Router) Apparently, ACKs don't need to be sent on OK's to REGISTERs. Plus, malformed data somewhere... no details on that, though. JT >Date: Sun, 23 Feb 2003 23:54:07 +0100 >To: John Todd <jtodd at loligo.com> >From: Jiri Kuthan <jiri at iptel.org> >Subject: Re:

This is almost certainly not an Asterisk-specific posting, but due to my inability to find a VoIP-focused Cisco list, I'll post here in the hopes of finding a more diverse user community. I am using a Cisco 7960 (version 6.0 SIP firmware) with Asterisk, and have been experiencing situations where the phone locks up. "Locks up" means that the bottom part of the screen

John, I spoke with Level(3) last week regarding SIP termination. They quoted $0.01/minute, with an 11 Million Minute / Month minimum. Ugh! -dg -------------------------------------------------------------- Darnell Gadberry President binaryMedia darnell AT binmedia DOT com ------------ Date: Fri, 19 Dec 2003 21:12:22 -0500 To: asterisk-users@lists.digium.com From: John Todd

All, I've been trying to troubleshoot a difficult-to-pin-down domain join issue, where a fully-updated CentOS 6 host cannot join a domain hosted by a fully-patched Windows 2008 R2 domain controller. I am running CentOS 6's samba build version 3.6.9-167.el6_5. I have checked all of the usual suspects (time, kerberos, DNS lookup), and all seem well: # kinit administrator Password for

Interesting notes on the 79xx series. The 7920 is the wireless phone; not mentioned here. For a more complete guide to Cisco's phones, see: http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheets_list.html The 7902 is the "very inexpensive" Cisco phone, and it looks like it will be SCCP (Skinny) only. Twiddling my thumbs here waiting for the chan_sccp to

hi folks, As a response to " Dynamic lookup of $var at is deprecated. Support will be removed in Puppet 2.8. Use a fully-qualified variable name (e.g., $classname::variable) or parameterized classes." warning i''m trying to apply fully-qualified variable name solution following this Doc http://docs.puppetlabs.com/guides/scope_and_puppet.html, but i have a problem with variable

Hi there Just a couple of questions to help me understand. When I upgrade from 3.0.22 to 3.0.23c the access restrictions on my shares on the member server no longer work. I read in the Changes file "Since Samba 3.0.8, it has been recommended that all domain accounts listed in smb.conf on a member server be fully qualified with the...." So far so good I know what I have to do still I

Hi, In the foreman apache module, I get an error when trying to apply the "::config" part of it. The section that causes problem is: file{ "${apache::params::configdir}/${apache::params::conffile}": mode => ''0644'', notify => Exec[''reload-apache''], require => Class[''apache::install''], } in

I really should be doing something better on this beautiful weekend, but I'm trying to save myself some time for later projects by documenting some things that have been particularly troublesome in the past. That being said... I've written up a configuration guide for the Cisco ATA-186, which describes some of the features that are possible to set in the ATA and specifically what

Trying to cleanup warning messages like the following: Dynamic lookup of $domain at /etc/puppetlabs/puppet/modules/centrifydc/templates/centrifydc.conf.erb:222 is deprecated. Support will be removed in Puppet 2.8. Use a fully-qualified variable name (e.g., $classname::variable) or parameterized classes. Do I simply replace <%= domain %> with <%= ::domain %> ? Same goes with

Environment: Samba server: ?Solaris 10 5/08 ?Samba version 3.0.24 Windows client: ?Windows Server 2003 Service Pack 2 extract from smb.conf: [global] ? ? ? ?workgroup = BBB.CCC ? ? ? ?wins server = xxx.xxx.xxx.xxx ? ? ? ?smb ports = 139 [WINDCHILL] ? ? ? ?path = /ptc/eaiXMLFiles ? ? ? ?force group = plm ? ? ? ?read only = No ? ? ? ?force create mode = 0775 Up till May 12, you could create a

Greetings! I've ran into a couple difficulties for a particular rsync use-case I'm involved in implementing - hoping someone here might be able to clue me in! I've got an existing linux box, with a particular directory that needs to be mirrored onto another server, into a different base path on that server - i.e.: server1:/var/lib/bar/ to server2:/backupdir/var/lib/bar/ Now,

Sipura does include STUN as an option. It has for quite some time. We are using it with all of our Sipuras behind NAT'd gateways and it works great! Try upgrading to the latest Sipura firmware rev. Darren Nay > -----Original Message----- > From: John Todd [mailto:jtodd@loligo.com] > Sent: Saturday, May 22, 2004 1:57 PM > To: asterisk-users@lists.digium.com > Subject:

>Date: Sat, 11 Oct 2003 22:07:49 -0700 >To: asterisk-users@lists.digium.com >From: John Todd <jtodd@loligo.com> >Subject: RE: [Asterisk-Users] SIP / IAX over satellite > >[post re-ordered chronologically] > >>-----Original Message----- >>From: asterisk-users-admin@lists.digium.com >>[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Tilghman

I was using Puppet 0.24.8 and now upgrade the puppet server and client both to 0.25.4 (on 1 machine i.e. server and client), but i am facing some problems. While testing, some of the configs which were/are running fine on older version generates error here; for instance, i have defined the FileSource path in ''/etc/puppet/fileserver.conf'' [myFileSrc] path

-----Original Message----- From: Adams, Gavin Sent: Monday, August 04, 2003 6:10 PM To: 'asterisk-users@lists.digium.com' Subject: Cisco 7960, SIP, NAT, Voicemal Hey all, I've got a couple 79xx phones working peer-to-peer and am now trying to work on the voice mail. In extensions.conf: [ATL] exten => 4001,1,Dial(SIP/gadams)|10 exten => 4001,2,Voicemail,u4001 exten =>

I know I heard people looking for inexpensive ADSI phones a while back. Can't vouch for these guys, but this looks like a reasonable deal for what perhaps are new phones. No experience with ADSI, myself, but thought I'd pass it along. These folks also carry the Nortel analog units that do intercom and 2-line, as has been discussed in prior threads. JT ---------------------

Mark has fixed the REGISTER issues to be more RFC compliant. I've created a new thread so that those of you who got bored with the old thread might read this new one. The feature that has just been added was added a while ago, but now it actually seems to _work_. :-) If you have a SIP server to which you are trying to REGISTER, and they demand valid domain (the part after the

Hello all - Second-to-last notice to collect RSVP's for interested parties for the Asterisk Birds of a Feather dinner/beer meeting. There are five people signed up for this at the moment. Looks like it's going to be the 23rd of September, since that fits well with the VON conference schedule. Please email me directly for details. JT >Date: Thu, 21 Aug 2003 01:52:52 -0700

I am trying to do a "factory reset" of an ATA186 using the widely-available instructions (basically dialing "FACTRESET#" on the keypad while at the menu prompt). I have done this a number of times before with success, but on this unit the lady spells out "P A S S W D" when I finish up the entry. Does anyone know what to do next? Hitting the star key (which is