similar to: Thoughts on building a Voicemail only Asterisk server?

hi all i'm running debian sid..and i tried to compile the las version of compiz because the old version of compiz did not supported compiz settings.. btw here is what i did: 1) followed the official compiling guide => http://go-compiz.org/index.php?title=Compiling 2) try to run by launching compiz...but the only results was to not view the windows' borders.. so i searched in the

On Fri, Dec 27, 2019 at 7:04 AM Eddie O'Connor <eoconnor25 at gmail.com> wrote: > > Wouldn't a "find grub / find grub*"....or "locate grub / locate grub*" > work?... > locate would require to install that package and run updatedb first. But, it would save time looking for files that have been already added to its db, specially compared to running

Hello, I have a graphics-related question: I was wondering if anyone knows of a way to create a bar chart that is colored with a three-part gradient that changes at fixed y-values. Each bar needs to fade green-to-yellow at Y=.10 and from yellow-to-red at Y=.20. Is there an option in a package somewhere that offers an easy way to do this? Attached is a chart I macgyvered together in Excel using a

-----BEGIN PGP SIGNED MESSAGE----- Friday, April 11, 1997 The Litterbox Sean B. Hamor <hamors@litterbox.org> LINUX IP MASQUERADE Synopsis: A problem exists in IP

Dear all, Hi. This is imacat from Taiwan. I'm new to this list. I was migrating from Qpopper to Dovecot. (Qpopper copies the mailbox when POP3 logged in, which causes quota problem.) Generally Dovecot is nice. However, I have a few questions: 1. I have a few virtual users, and I'm not using IMAP. Do I still have to give them a writable home directory? I mean, they

Hello, Long time subscriber/reader of this list - thank you for all the great ideas. Scenario: We currently provide a hosted ACD system using Mitel phones (speaking the Minet protocol) to an NCI based server solution. The logic behind this choice was the emulation of key system features etc... Many of our clients have asked for basic call queue functionality: - Agents having the ability to

Hey all...I'm trying to use gnophone to connect to my asterisk box behind my firewall..I thought I could just setup a tunnel with something like ssh host.com -L5036:asteriskserver:5036 and just change my gnophone to connect to localhost:5036 but I never see anything happen on the asterisk server. I'm even trying this on the same network just in case there is something funky with NAT.

Hello, We are working on implementing a solution for a medium service provider. They were previously using a Cisco AS5300 gateway with some PRI trunks to receive modem calls, then route them out the Internet. The Telco they were buying the trunks to discovered this configuration and restricted them due to legal conventions, and stated that in order to continue doing this, they would have to talk

Juan, I am running a Calling Card application on a Dell PowerEdge 2850 with Asterisk 1.0.7. Recording conversations I have seen on my server causes the processors to burn more than necessary so I would recommend what William from Signate recommended: " Consider saving recorded calls in a database on a separate server. It will be simpler to build a retrieval interface that does not

Hi I am setting up a phone system for a small office. The office will have 5-8 phones and a fax line. There are 4 hunt lines coming into the office. We have made no hardware purchase yet. Being an asterisk newbie, before I suscribed to this list I just assumed that I would buy voip phones and connect all the phones to a private ethernet network. However, I see many people inquiring about FXS

Hey all, I know you all saw the topic and let out a groan. However, I understand how to get an overhead paging system to work with respect *, however I am now looking for a small(?) paging amp, that I could hook 3 or 4 horns to. I would like to just have the * extention be routed to a soundcard and out an output, so I would like an amp that is voice signal activated. Has anyone found anything

I sent this last night but it never showed up in the thread list so I'm trying again. Please pardon me if it duplicates. So I've been banging my head against the rack on this one and am now turning to the group for help. I'm in the process of bringing five Asterisk servers (all originally built from source code by myself) from various versions (1.6.2.x,11.6-cert13, and 13.1-cert2) up

Hi all, first post. My company's office in the UK is soon going to get a Cisco VoIP solution system. What I am interested in, and couldn't find googling, is if it is possible to connect an Asterisk solution to the Cisco system and have all the nice advantages of it (mainly calling the extensions and directly reach the other office). Thanks, have a nice day Simone

Hi Mark, yes, you are right... these are different VLANs I configured the other phone to use the same IP (192.168.1.13)... and it worked flawlessly... on the SAME Networkcable in the same plug... so it must have something to do with the polycom phone config... remember... when I use tcp the phone tries to register, but does not even try with udp... thank you, yves Am 21.12.2016 um 13:34

Hi! I did manage to load phone with SIP image : SIP70.8-0-3S, made SEP-MAC.cnf.xml, but phone never read the configuration from it. On the screen it's written "Unprovisioned", and phone is not trying to register with asterisk. Please help!! MihaelaMJ -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all. What is the best hardware configuration to handle this following scenario? - 4 IVR menu with conference applications for each option; - Only SIP/g711 user access - 3500 simultaneous users(800 at the beginning) - No ZAP channels Where is the most important point of failure? CPU? Ethernet? RAM? Im planning to separate in three servers: Server01: 01 Xeon 3Ghz getting the 1st level of

Announcing a new Windows-based IAX/IAX2 client. Please download it and give it a try. Let me know about any bugs, and any missing features. I have yet to come up with a catchy name for it, so at this point it calls itself IAX Phone. (Suggestions? Non-derogatory suggestions, preferably). Download: http://www.sokol-associates.com/Downloads/IaxPhone.zip Reference & Support Page:

It is amazing to me at this point that there is not an official Digium list of supported servers (including 1u models!). Clearly the number 1 issue with the Digium PRI cards is the server that they are used in. The new cards even go as far as listing server that DO NOT work on the Digium site! The wiki references are old and do not have any testing parameters. C'mon guys! Certify a

Hi, I was wondering if anyone had any problems with the Dell 1800 series servers, with TDM400 cards? I saw that digium seems to recommend against a lot of the dells. Please let me know Thanks Sascha -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050802/d8219499/attachment.htm

sorry... typo.... the problematic phone has the 192.168.0.13 the asterisk has 192.168.1.211 when i connect a snom phone on the cable that was in the soundstation 6000 before and configure the phone to use 192.168.0.13 it does register on the asterisk via 5060/UDP... it would be helpful if someone, that has a running soundstation ip 6000 could send the configuration... :-/ regards, yves Am