similar to: Setting up 1.6.2.9 on Debian 6.0 Squeeze

Hi. Is it possible on Solaris 10 5/09, to rollback to a ZFS snapshot, WITHOUT destroying later created clones or snapshots? Example: --($ ~)-- sudo zfs snapshot rpool/ROOT at 01 --($ ~)-- sudo zfs snapshot rpool/ROOT at 02 --($ ~)-- sudo zfs clone rpool/ROOT at 02 rpool/ROOT-02 --($ ~)-- LC_ALL=C sudo zfs rollback rpool/ROOT at 01 cannot rollback to ''rpool/ROOT at 01'': more

Hi, Using MeetMee(1234,M) on asterisk 1.6.2.9 (debian/testing) we have terrible sound: the MOH is unrecognizable and speakers can't be understood; it sounds "ghostly". However the prompts ("your are the only one in this conference, etc.") sound fine. Our server has a Digium T410P card with two E1 lines going in and the wct4xxp dahdi module. Any idea?

hi, list i want to know what is the best OS for install Asterisk 1.6.2.9, which should work properly on working system. i want to use CentOS5.2 or CentOS 5.4. Which is better and stable? Thanks for your help. -- Thanks for your supporting, have a nice day. Sucan

The Asterisk Development Team has announced the release of Asterisk 1.6.2.9. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.9 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by community

The Asterisk Development Team has announced the release of Asterisk 1.6.2.9. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.9 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by community

Hello I just set up Asterisk 1.6.2.9 through packages on a test host running Ubuntu 11.04, configured sip.conf/extensions.conf, and launched EyeBeam 1.5.20 to run the echo test. For some reason, even through I'm using a headset, there's a lot of echo and after a few seconds, it sounds like it enters a very fast loop before the echo stops somewhat. IOW, unusable sound. Here's a

Deutsche ?bersetzung folgt / German version following ===================================================== Hello, i have Asterisk running with 2 ISDN-Cards. One AVM Fritz for connection to german ISDN and one HFC-compatible-Card (NT mode) for connection to ISDN-Phone (later: ISDN-PBX). Here is my actual installation: ISDN -> Fritz - ASTERISK ? HFC-NT <- ISDN-Telephone If i pick up my

I'm attempting to get two Asterisk 0.4.0 PBXes to communicate with one another using IAX/IAX2 trunks. I've managed to get a semi-functional NAT Firewall working as a PBX (with Asterisk running directly on the firewall itself), but there are issues with bind()ing to various interfaces which is causing outbound SIP issues. To get around these issues, the idea is to do something like

Hi All, Got a strange (at least IMHO) issue that doesn't make much sense to me. Basic configuration is two sites with a site-to-site (aka router-to-router) VPN. Both sites have Cisco 7961G phones [with SIP firmware] on users' desks, and the only VoIP is internal - all of our outward telecom is on POTS or Centrex-enabled POTS lines. Site 1 has a Dell PowerEdge 1950 with Asterisk built

Hallo Herr Jahn, falls Sie eben Anrufe von uns bekommen haben und Sie nichts geh?rt haben - das liegt an Problemen, die wir momentan mit unserer Telefonanlage haben. Deswegen kurz per Mail: Ich habe die Pakete auf Datei-m2 installiert. Da wir mit dem Setup von den Samba-Servern ja zuletzt einige ?berraschungen erlebt hatten, w?rde ich vorschlagen, dass Sie den Server erstmal mit einem anderen

Hello good ppl, A couple of questions for multiple pbxes 1. Is it possible to support multiple pbxes in one Asterisk box(using contexts, etc.)? 2. Can we use the "domain" field in sip.conf to specify the different domains for sip users, having one domain for each pbx? I just tried registering two xlites, with different domain names (with the same specified in sip.conf). But, Asterisk

Hi at all! I'm actually fighting to make sieve in dovecot work and made quite a success by now. However, I still fail at the following constellation: Background: I'm a Mail Admin of a small IT department and we are already using Dovecot as LDA with a filtering server. Any user can easily create filter rules that apply to him (to make it easier for my colleagues we use the Roundcube plugin

As far as I can understand the mentioned improvement is targeted on a polling mode for a MSSQL, there is no wonder for observing relatively high hypervisor CPU consumption without any sign of same consumption in a guest itself. Certain kinds of applications (PBXes, Chrome/Chromium, seemingly MSSQL) may poll some resource very frequently, causing wakeups in the host for guest process and increasing

hi everybody and thanks for replying to my previous posts! i have some questions about the new cards announced on digium's website: these cards are 1-4 fxs interfaces, meaning i can take 4 analog phone handsets w/standard rj11's and plug them in to dial via *, right? not until fxo is implemented (is it going to be? when?) can i plug analog phone lines in? if the ports are made fxo/fxs

Does anyone have a working example of how to use the "switch" directive to peer two Asterisk PBXes? -- - Ian C. Blenke <icblenke@nks.net> (This message bound by the following: http://www.nks.net/email_disclaimer.html)

In May, I posted an inquiry to the list concerning my desire to configure my own CLASS codes in extensions.conf rather than having them hard coded into the channel drivers. I have a number of old rotary dial telephones that (obviously) can't dial *. Traditionally in the US, "11" can be dialed in place of "*" as the first digit dialed. Many people mentioned that this

I'm trying to use Asterisk app_parkandannouce to build a global parking pool from within a couple of Norstar PBXes. Right now I can blind transfer calls into the parking lot, but the slot announcement relies on calling back the 'transferee' after the call is parked and I can't pass enough callerid data out from within the PBX to be able to route the call back in (ie. no PRI

Hi all I noticed that most caller are quite confused by the standard voicemail announcement text. Especialy as the number read is the 'internal' number. Callers often hang up because they think having called the wrong number when they hear the announcement. Is there a way (like in many other PBXes) that the VoiceMail user could record his own announcement? (like, hello, this is the

Hi Well it is well some time that my last DUNDI peer has become unreachable. I guess too many issues with spoofed numbers etc. But I am wondering, do people, especially larger entities like telcos, still use DUNDI? I know that in some Hamradio communities, DUNDI is used to interconnect PBXes, but that is with private phone number ranges, not connected to the public. Want some DUNDI peering?

I need to deploy some quasi-virtual-PBXes, and I'd like to avoid having to be hands on for each new phone number deployed... so I would like to set up some administrative extensions that can record greetings... lets say: [admin] exten => 8(NXXNXXXXXX),1,Record($1|-greeting.gsm) [incoming] exten => _(NXXNXXXXXX),1,Playback($1|-greeting) exten => _(NXXNXXXXXX),2,Goto($1,1000) exten