similar to: Grandstream HT 503 Outoing 403 Forbidden

Question for all: Right now i have a Table in a mySQL DB that has a row called Ingredients. When the data is entered into the DB its enter like so from a text area: 1 1/2 lbs. beef top sirloin, thinly sliced 1/3 cup white sugar 1/3 cup rice wine vinegar 2 tablespoons frozen OJ concentrate 1 teaspoon salt 1 tablespoon soy sauce 1 cup long grain rice 2 cups water 1/4 cup cornstarch 2 teaspoons

My uncle recently bought a new computer and he wants me to help him install Guitar Pro. The thing is, he doesn't have Windows, and I'm thinking of installing Ubuntu with wine on the comp. Quiz time! Can you use wine, with GP, and if you can, deos it perform poorly in any ways?

Hi, I have a grandstream HT 503 ATA which seems to support TLS/SRTP. I generated the keys and certificates for both the server and the device as mentioned at https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial I then copied the grandstream.crt and grandstream.key over to the ATA. Unfortunately, the device is refusing to register and in asterisk cli i see SSL certificate ok ==

Hi Right now I have my url showing as the following: http://localhost/categories/italian/recipes/special-pasta-sauce I would like it to be: http://localhost/categories/italian/special-pasta-sauce (or even http://localhost/italian/special-pasta-sauce) my category model has_many :recipes and my recipes model belongs_to :category (for the purposes of my test project a recipe can only belong to one

>Asterisk and it works fine untill the following >situation: > >- one of the telco lines occasionally becomes mute after call is completed, would not provide dial tone, (not sure about ringing on that >line) - both via old and new PBX. >- zap show channel <n> would show that line as 'Offhook', though no telephone is off hook. > >If physical line would be

John Todd wrote.... > Cisco has an 802.11 phone called the 7920, which is apparently > shipping now. It is very expensive (>$550 USD) and only runs SCCP at > the moment, which is Cisco's proprietary VoIP protocol. However, if > it falls in line with some of Cisco's other high-end VoIP equipment, > that means it should have a trailing-edge SIP image running by

Dear R users, I would need some help in building R using minGW in windows 8.1. After using the command *configure *(./configure --enable-R-shlib --with-readline=no --with-x=no), I use the command *make. *This results in the following error: [...] make[2]: Leaving directory `/home/Edoardo/r-3.1.2/src/nmath' make[2]: Entering directory `/home/Edoardo/r-3.1.2/src/unix' make[3]: Entering

greetings like many, i get a daily logwatch email from "root" about 4AM last night, i rebooted a server from remote between midnight and 2AM and, uncle me, i received an extra logwatch from "anonymous" at the same machine can anyone explain or has anyone else else ever seen this behavior? just wondering and thanks in advance - rh -- Robert Hanson - Abba Communications

Hello I?m running a samba 3.0 server on slackware box as my PDC. I?m using a bunch of win2k client in Brazilian Portuguese. Everething run fine excepted that when win2k client try to overwrite files on all the server shares they get a message like "Unable to Copy file, make sure that the name is spelled correctly" ( Sorry for the translation, i?m using a Brazilian version of windows ).

Hi folks I am getting this error: root@collab:/home/admin# smbldap-passwd testuser Changing UNIX and samba passwords for testuser New password: Retype new password: I cannot generate the proper hash! uncle google was rather quiet on the subject :( what do you need config file wise? thanks Bernhard -- Graylion's Fetish & Fashion Store Goth and Kinky Boots, Clothing and Jewellery

Hi all, I have a strange situation on a Asterisk 1.2.17 with FreePBX 2.2.1 Doing a select in the CDR table I noticed there are some calls with billsec greater than duration, duration is always 0 in those calls. How can this happens ? Am I missing something ? Tnx in advance Regards Edoardo Serra WeBRainstorm S.r.l.

Outoing calls I make using Motif Google Voice Calls continue ringing even after the other end picks up. I have to restart Asterisk to resolve the issue. I don't see any errors. It's not recognizing that the other party picked up the phone and restarting Asterisk fixes it only for a day. -- Co-op Vacation Rentals www.coopvr.com 15218 Summit Ave Suite #300-354 Fontana, CA 92336

Hello everyone, I'm in the process of running an Icecast server and I would like to know some best pratices. 1. Should I place Icecast on port 8000 or should I change that to one more common (80, 443...)? 2. Should I place the server behind a webserver like ngingx or apache? 3.Can I disable the login interface? what can be disabled? My best guess is to run icecast behind a webserver,

Hi everyone, I have used used scriptalicious for the autocomplete. It works but with one problem. When scrolling down the list, the values not visible are not scrolled into focus. eg I use the keyboard down or up key. Here is my CSS <style type="text/css"> div.auto_complete { height: 100px; width: 350px; background:

hi all, "uncle"! i've thrown up my hands with cyrus, and am looking for a 'better alternative' ... i'm brand new to dovecot (haven't even built it yet ...), and have spent the last while reading through the LIST archives here. apart from the TECHNICAL aspects, which seem _more_ than adequate/competitive, i'm most attracted by a comment abt this

Hi Guys, we're needing a special implementation on Asterisk Our intention is to contribute the development and share back the code to Asterisk community Here is what we need: - An option to Asterisk Dial command which, if used, when calls is answered gives monodirectional audio (Caller should hear the called party but not vice-versa) - A DTMF sequence (maybe handled in features.conf) for

Hi all, I'm having a problem with some Asterisk servers interconnected with each other using IAX (I also tried with SIP without solving the problem) Sometimes, with apparently no reason, some peers become UNREACHABLE (I have qualify=yes in iax.conf) and REACHABLE again as soon as another qualify test is made. Our users are also complaining about audio loss during their calls, apparently

How can I get a list of all model classes in the domain of a Rails application (all models, both in "app/models" and in components/somedir/model.rb)? Thanx in advance for your precious help! Edoardo "Dado" Marcora

I'm in the process of nuking a bunch of dead mailboxes after they've been migrated to other servers - but the accounts have been kept. A simple shell script takes care of most of it, BUT.... (there's always a "but" isn't there?) One user has named all his mailboxes with leading hyphens. ie: -foo -bar -bazz and "doveadm mailbox delete" doesn't like it.

Just to tease people I'll be post a URL to my review of the HT-286 by thursday..