similar to: Release Cycles [was RE: special tricks for developersboxon centos 5]

Hello, all. After reading the README, UPGRADE.txt, and a quick tour through google, is it safe to assume to upgrade from 1.6.1.1 to 1.6.1.2, one simply compiles and installs over the old installation being careful to NOT install the sample files? Thanks - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsullivan at opensourcedevel.com

Hello, all. I see there is an "o" option for the Dial() command which reverts to the previous behavior of using the original callerid throughout the call - I suppose more specifically, using the callerid from leg 1 for leg 2 in B2BUA if I understand it correctly. That seems to be highly desirable behavior; I know we are seeing some problems with call history and call forwarding because

Hello, all. After many pages of googling and testing in the lab, I'm still a bit perplexed about how to implement tls protection for the asterisk manager. manager.conf allows one to specify the cert file but one normally must also specify the private key file. If I simply enter the cert file: sslenable=yes sslbindport=5038 sslbindaddr=172.x.x.8 sslcert=/etc/pki/tls/certs/pbxc.pem ; path

Hello, all. I'm afraid I've been dropped into the deep end even though I am an Asterisk novice. I've set up a few tiny, tiny systems in the past and have now been asked to pull together Asterisk, FreePBX, Kamailio, RTPProxy, and Fedora Directory Server into a VoIP service. After googling and reading for most of the last 24 hours, I finally have my head around the components and how

Hello, all. Since implementing an iptables firewall between the Asterisk PBX and several SIP phones, the Asterisk PBX ability to "reinvite" has been broken even when the phones are on the same network (i.e., no firewall between the phones). We've been beating our heads against the wall thinking it was the complex rule set but it appears the issue is ip_conntrack_sip. Before I drop

Hello, all. In our multi-tenant environment, we would like to be able to use the reinvite media redirection within Asterisk for calls within a tenant but not between tenants. We would like inter-tenant calls to be fully proxied by the Asterisk server. I think the answer is, "we can't," but I thought I'd ask anyway. I'd dearly like to remove the substantial traffic

Hello, all. My apologies up front but I must be brain cramping on something very simple. I've tried to pare down my configuration to the absolute minimum for SIP traffic just to understand how it works. My incoming calls are not finding the "s" extension in my dial-plan. I am assuming SIP calls can do this. I am using Asterisk 1.6.1.1 sip.conf has nothing but: [general]

Hello, everyone. No need to read this message. I'm posting for documentation for other poor, ignorant slobs like me who are struggling to pull together the many technologies to make converged networks happen. Hopefully, this will help save someone else the time I spent. I started the below email until I realized I had solved multiple parts of a compound problem but not all at the same time.

Hello, all. I'm having a deeply frustrating time getting multiple parking lots to work and am wondering what I am doing wrong. I am using Asterisk 1.6.1.1. I defined two separate parking lots in features.conf as follows: [parkinglot_a100] ; SSI context => a100-parking parkpos => 900-920 findslot => next [parkinglot_a10] ; EBC context => a10-parking parkpos => 800-820

I am following a document teaching how to use regression and right at the onset I get an R error. I understand that variables "conc" and "signal" should have the same length but I am using the what R manual suggests to drop the error and it is not cooperating. What gives? The manual says that the default is na.ombit by the way, not that it's addition does anything How

mailboxes). Are you certain that removing either 612 or 610 mailbox would keep Asterisk from complaining ? > > However, the MWI does not indicate voice mails for 610 and I keep seeing > this error message: > > ERROR[2549]: app_voicemail.c:1630 messagecount: Couldn't find mailbox > 610 in context a10 > > However, mailbox 610 is clearly defined in voicemail.conf: >

Hi, I just ran across the annoying issue, that with lda_mailbox_autocreate = yes lda_mailbox_autosubscribe = yes set, parent folders of newly created subfolders get subscribed without being existent. Let me give an example: In sieve I use fileinto "foo/bar"; but "foo/" does not exist as folder. So the Mailbox "foo/bar" gets created, but without "foo/"

Hello, all. I'm running Asterisk 1.6.1.6 on CentOS 5.3 in a multi-tenant environment with IMAP voice mail storage on Zimbra. One of our clients is having a problem when transferring voice mails from one mailbox to another (option 8 in the standard voice application menu) using their Snom 320 and 360 phones. The end results is the final recipient cannot listen to the voicemail. We also email

This is what I get when I do ldd on the icecast binary. [root@gala bin]# ldd icecast libcrypt.so.1 => /lib/libcrypt.so.1 (0x0ffb3000) libm.so.6 => /lib/libm.so.6 (0x0ff65000) libpthread.so.0 => /lib/libpthread.so.0 (0x0ff2e000) libc.so.6 => /lib/libc.so.6 (0x0fde9000) /lib/ld.so.1 => /lib/ld.so.1 (0x30000000) [root@gala bin]# It looks like

On 02 Apr 2020, at 03:48, Aki Tuomi <aki.tuomi at open-xchange.com> wrote: > On 2.4.2020 11.07, @lbutlr wrote: >> Is there any reason that a .active_sieve file would not load because it is a symbolic link? The target of the link is readable by the mail user, but instead of getting the trace for the .active_sieve file, I only get the trace for the default sieve file. > > Does

Emperors1000@aol.com wrote: > > Dear reader, > The emperorsclothes list does not seem to be working. Please forgive us if > you receive this post twice. - Jared Israel, www.tenc.net > ************************** > URL for this article: http://emperors-clothes.com/articles/rozoff/abyss.htm > > To join the Emperor's Clothes email list, go to >

Hi all, I hope this is the correct list to post such a question. I was trying to install the R-project on Debian and encountered significant problems with the same. The main problem is the installation of the libc6 package. I need this package in order to install the R-core package. However, the libc6 is dependent on the libdb1-compat package, which just refuses to install on my server.

Just a thought as we explore the brave new world of converged voice and emails. Voice mail boxes typically hold a very small number of messages while email folders contain thousands. Do we need to rethink the traditionally small limits on voice mail boxes when storing in IMAP or are the messages counted separately? Thanks - John -- John A. Sullivan III Open Source Development Corporation +1

Hello, all. The little bit of reading I've done on lua makes me eager to give it a try. However, when I try to install it (Asterisk 1.6.1.1 on CentOS 5.3), it is not available in menuselect. I have installed lua and lua-devel. I've seen very little about it in my Internet searches. What else must I do so that it installs? Thanks - John Oh, by the way, I'm having a similar problem

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