similar to: OT: Free DID/SIP accounts

I have a rather simple setup running under Asterisk 1.4. I'd like to move it to a new install of 1.6. Before I bring it online are there any gotchas I should look for? A Gotcha README would be better but searching with Google and the forums, for me, gets hits that deal with hardware issues -- cards etc. Nothing about depreciated/changed commands. TIA, Rod --

FYI/Heads up, I /just/ received what looks like a phishing attempt for information about Open Source PBX usage. It says it comes from Digium but all the links (including the one for digium.com) point elsewhere. Rod --

Anyone tried installing Asterisk in a AWS server? \\||/ Rod --

I was thinking about all the old email I have and don't want to get rid of and how much space it is taking on my mail server. The idea came to me to install Dovecot on a server at home then I could drop mail into I want to save. It doesn't need to receive mail except what that is moved to it. Anyone see problems with doing this? Thanks, Rod --

I'm still searching but hoping someone can offer a clue-stick. Long story short! I had a server crash suddenly and all I can get at are the files. Built a new host and copied the data and config files over, correcting ownership and permissions (hopefully) as I went. But now I can't get logged in. Messages in /var/log/dovecot/dovecot-info.log, without saslauthd running, are like

And if that ain't confusing I don't know what would be. I bought a TDM400 with two modules (FXO, FXS) a couple or so years ago and ended up never using it. Passed it along to a friend who is having some problems with it. (He isn't on this list.) We've both tried searches using Google but haven't been able to find anything that helps. So this is more a question of

> From: "Roderick A. Anderson" <raanders at cyber-office.net> > Subject: [Dovecot] RHEL5/CentOS5 YUM repo, rpm, or spec file for 2.0? > > I don't remember sing any mention come across the list reference the > Subject line and nothing shows up within the first three pages of a > Google search. > > Anyone know of a YUM repo. RPM or spec file for

I've got the text files created -- thanks to Russell Bryant -- for re-building the core and extra sounds using another voice but I'm not sure which formats to actually build. This will be a small/personal system using Vitelity.net so will only have SIP connections. The /var/lib/asterisk/sounds/ directory contains .alaw, .g722, .g729, .gsm, .ulaw, and .wav. What are the minimal

As I said in a previous reply the server is going great. In fact I can even send mail via it. (On the really old server I'm moving from I couldn't get authentication for outbound to work.) I now have a couple of small issues to deal with before moving completely off the old system. Virtual domains aliases? My reading seems to indicate that Postfix only handles aliases in one

More about IMAP, less about Dovecot. This is more an IMAP question than specifically about Dovecot. Sorry to ask here but I've been researching, reading and lurking with no answer so far. What I'm doing is running reports as a user from a remote system to provide a list of messages (Sender/From, Subject, Date, size) in a specific folder(?). In one case it isn't even a Dovecot

I'm working at the next part of the virtual domains mail server. I'm moving this account (raanders at acm.org is a forwarder) which has a bunch of procmail rules to file into folders. My question is if it is more efficient is use? if { ... } elsif { ... } elsif { ... } else This seems to be the way many of the example scripts do it but I found at least one that used if

Folks, Please forgive my ignorance. I think that what I am asking must be so obvious that no one bothers to write it down. But I don't know the answer so... I want to set up * with one incoming VOIP phone number. If someone calls me and is talking to me on that phone number, how does a second caller get to * voicemail instead of a busy signal? Thanks, -- Art Z. -- Art Zemon,

Hi, I'm struggling to separate inbound calls fro a SIP provider that does not send DID. I have tried .......sip.com/12345678 on register string different context=from-no-did Port not possible as only support 5060 Any suggestions? Thank you! HB

Hi all, I have two sipgate accounts (numbers), if I have both accounts register only one will work for incoming calls (which is all i'm interested in). However if I disable either account the other account will work perfectly. Am I missing something obvious? Thanks in advance, Ray. Excerpts from sip.conf - [general] 8<---- SNIP! ---->8 Register => 1212121:aaaaaaaa at

Anyone aware of a Web User Interface or GUI to allow users to create simple/common-type sieve scripts? I found the sieve testing and have used it as I play; plus mangesieve (and Net:ManageSieve.pm) but want to allow users to build so some clicking and maybe a little typing and then end up with some filters. So before I go invent this wheel I'd like to know what others are doing? TIA,

Dear all, I have an asterisk sip issue which I don't believe is unique. I use a registrar (sipgate.co.uk) where I have 3 different accounts. These accounts provide me with three seperate local phone numbers which allow me to allocate them to seperate users. By using just one of these accounts I can set asterisk up to send and receive calls no problem. However, when I start to introduce an

Yah, jimc again. I still cannot figure out what is wrong with my dc. The problem began with being unable to log on from Windows boxes after a power hit. The initial symptoms were: 1) When I log in on a Windows machine, with an account that should have administrator permissions, I don't have said admin rights. 2) When I attempt to launch the Windows remote server tools, I can select the

Dear all Suppose I have teeth like data similar like x <- 1:200 y <- 0.03*x[1:100]+rnorm(100, mean=.001, sd=.03) z <- 3-rep(seq(1,100,10),each=10)*.03+rnorm(100,mean=.001, sd=.03) plot(x,c(y,z)) and I want to have a gradient estimations for some values from increasing part of data like y.agg <- aggregate(diff(c(y,z)), list(rep(seq(1,200,10),each=10)[1:199]), mean) y.agg[1:10,]

Hi, when i include a sip configuration from another file in my sip.conf using #include /etc/asterisk/sip-sipgate.conf everything seems to be working. The peer is listed when i execute "sip show peers" and Status is "OK". But the peer is not listed using "sip show registry". I need to place the "register => ..." in the sip.conf to make it work. Is this

> Hi folks, > My SMB server is a Helios Ethershare 3.1 on a Tru64 Alpha machine. > We are now integrating more and more gentoo-Linux Servers. These are the > SMB Clients. > > On Obelix (one of our gentoo-Servers) I can mount the SMB Volume from the > Ethershare Server. > obelix nc4smb # whoami > root > obelix # mount -t smbfs -o username=m.heckmann