similar to: 503 "Unavailable"

[I originally sent this using the wrong email address - apologies if it appears twice] Hello Everyone, Has anyone tried installing CentOS 4 x86_64 on a machine with an Adaptec 2230SLP card? I've just tried it, and Anaconda could not detect the card. I've manually loaded the aacraid driver, which then starts the install process. But, when the install gets to disk partitioning, the

Hi all, Has anyone seen Evolution "forget" which emails have been read in CentOS 4? I have a client that's reported he's reading his email, closes Evolution, and when he starts it again, the messages are marked as unread. I found bug reports about this here: http://bugzilla.gnome.org/show_bug.cgi?id=273787 https://bugzilla.ubuntu.com/show_bug.cgi?id=4373 It's supposedly

Dear all - Some wisom please. We have a need to customise the Q.931 clearing causes being sent back to the network based on decisions made in some scripts or extension logic. i.e. I want to be able to decide to clear a call (either answered or during the alerting phase) with any clearing cause needed. E.g. 'All circuits busy' or 'network fault' or 'vacant number' etc.

Hello Everyone, I've been running Asterisk as our PBX for several months now, and recently I've been thinking about using one of the VoIP providers to lower our phone bill. I know that VoIP providers can supply their customers with a local number and/or virtual numbers, and then that number/account can be used with Asterisk (well, it depends on the provider and whether or not their

I am looking for a supplier of Digium hardware in Canada. Any suggetions? Thanks, Adi

Hi all, I'm experiencing something rather strange with our Asterisk setup. I'll go through the steps. Scenario 1: 1. I call someone on the outside (from my analog phone extension). 2. Everything appears honkey dorey. 3. Transfer called party to conference or another extension - fails. Scenario 2: 1. Receive a call at my extension from the outside. 2. Everything appears honkey dorey.

Hello all, This is my first post to this mailing list. I've been following the development (not as a developer, but as an interested user) of Asterisk for over a year now. It's only in the last month that we've decided to give it a try (hey, might as well...everything else we use is open source). We ordered a TDM11B (1 FXS, and 1 FXO). The card and Asterisk install was uneventful.

Hey all, Wondering if this is possible.. Incoming call is answered through X100P, then an extension is dialed using the same X100P card. Basically I want to dial in, enter 9 + <phone#> and have it do a flash then have it dial *08 <the same phone number> + # on the same PSTN line to have it transfer my call to another phone number. I realize this isn't very safe, but I would

Hello: I want to have a local telephone number that, when the people calls this number (via mobile or normal PSTN), the voip provider stablishes a SIP session to my asterisk box. It is possible? If yes... What providers have this service in Europe? It is difficult to configure and get things working ok? Will my asterisk box see the mobile or normal PSTN phone# that is calling the number

My question pertains to the use of IAE.. I would like to setup 2 Asterisk boxes. One would be located in our office behind the firewall and hooked up to our analog lines. The other would be located in a remote datacenter and used for our remote employees to connect to. I would like to be able to accept calls on the Office Asterisk server and route them to the Datacenter Asterisk server. Is

I'm having an oddball issue with a Polycom SoundPoint IP 500. As you can see below Asterisk thinks there are 2 SIP channels active, but show channels tells me there are no calls active. Anyone have any idea why this is happening? The Polycom occasionally stops accepting calls and requires a power cycle. fs-1*CLI> sip show channels Peer User/ANR Call ID Seq

Anyone in the Toronto area interested in getting together to share notes and swap war stories? -- Jim Van Meggelen jim@vanmeggelen.ca -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.6.9 - Release Date: 06/01/2005

Folks, I'm curious to know how the volume of Asterisk-Users rates as far mailing lists go. This list sees over 200 messages per day, which has GOT to put it in the top 5%, doesn't it? I'd love to know if anyone has knowledge of any organization that might maintain such stats. Regards, Jim. -- Jim Van Meggelen jim@vanmeggelen.ca -- No virus found in this outgoing message.

I am getting occasional seek error on my boot drive. The messages are as follows: pshda: dma_intr: status = 0x51 { Drive ReadySeekComplete Error } hda: dma_intr: error= 0x84 { DriveStatusError BadCRC } ide: failed opcode was: unknown Everything seems to be working fine but these errors do have me concerned. The last time they occurred I was created a fairly large cpio backup file in the

Hello, If one would like to build a redundant Asterisk setup, would it be possible to exchange the locationdb for the SIP users between then? IE, the following setup: SIP Phones -------------- Asterisk ------------------------ SIP carrier | | ------- Asterisk (standby) ------ Asterisk is used as a PABX in this setup, so the

I have an asterisk box setup with a T1 card (hooked to a pri from the telco). Sometimes when people call a number that rings down that pri, the first ring is really really long, like 3 normal rings put together. My indications.conf is the default that comes out after you do a make install. Is this (first long ring) something I can control?

Hello, does anybody know how to achieve the following with SSH... a) accept RSA authentication for all but root from any IP b) accept RSA authentication for root from a couple IPs/Netmasks c) accept password authentication for all but root from a dozen Netmasks d) accept password authentication for root from 3 local netmasks only ie. make authentication depend on the

hi folks. i have the following puzzle: when i call certain cell phone# using a regular phone & POTS. the called cell phone co. usually return a message such as phone travel out of range or phone is busy etc. if the phone is unreachable. now when i have the following setup: sip phone -> asterisk -> PRI -> phone co. i call the same cell# and if it's unavailable. the PRI return

I have an app that generates callfiles in the outgoing queue, which connect a channel to an AGI (Perl script) at an extension. The AGI calls the Dial command over a SIP channel. Sometimes I need to stop the outgoing calls after the requests have been made. I delete the callfiles from the outgoing directory, but there are still some calls "in the pipeline". Especially if Dials failed at

Hi, I am thinking of settting up a thin client server setup. I would like to set it up to work with linux and windows. I was going to use Symbiont but it says it needs a different version of linux to install. Has anyone used it with Centos before? Would it be easier to just install LTSP and set it up. I would like to b have a small box that does the dhcp and then allows users to logon to