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[trixbox - The Open Platform For Business Telephony] Voicemail Broken!! | trixbox: If I use putty and run the dir command on the device folder these folders all show up, but if I browse that folder with WinSCP they do not show up. Also none of these folders show up using the Dir command on the var/spool/asterisk/voicemail/default/ directory.

[Comments for Nerd Vittles] Nerd Vittles » It's PBX in a Flash 1.7.5.5: The Lean, Mean ...: You can only do this if you have set up dynamic DNS or if you have pointed a hostname at your fixed IP address. Once you have done this, and assuming your ISP is cooperative, then you will receive your voicemails via email if you wish (this is set within FreePBX),and your PBX will email you when FreePBX needs an update.

[trixbox - The Open Platform For Business Telephony] MWI with trixbox 2.2 and Audiocodes MP-124 fxs gateway | trixbox: If you alter sip_additional.conf (which clearly states at the top not to change) and then run a reload from FreePBX your changes in sip_additional.conf will be overwritten based on the dial plan in the database. There is no cron job and FreePBX does not update any phone or endpoint configuration files.

[WebWot.com] It's PBX in a Flash 1.7.5.5: The Lean, Mean Asterisk Machine ...: All the firewalls in the world won’t protect you from malicious phone calls on your nickel if you use your extension number or something like 1234 for your extension password because the SIP and IAX ports typically are exposed to allow connections to your providers. In addition to making up secure passwords, the latest version of FreePBX also lets you define the IP address or subnet that can access each of your extensions.

[VOIP-info.org Wiki Changes] Asterisk gui vmail.cgi - voip-info.org: "Attached is a patch I made to app_voicemail.c that will set the owner, group and perms of all files and directories that are created so that vmail.cgi can run as user/group apache without setuid on vmail.cgi. The owner/group/perms are configurable in some static variables at the top so if you need them to change, you only need to change them there.

[trixbox - The Open Platform For Business Telephony] New Aastra XML scripts 2.3.0 ready for beta (Beta6) | trixbox: We have identified some MAJOR problems with the deviceanduser mode which has never been updated since DEVSTATE has been introduced so all the user hints are wrong, we are looking for a fix in beta3 we already did some fixes in Beta2 but I would advise not to login/logout with the keys but with *11 and *12 until we fix everything. The fix may even be to actually call *11 and *12 as the AGI behind it is very complex and changes everytime freePBX updates it.

[All About Telephony] Asterisk 1.6: Different modules can subscribe to MWI and Asterisk will inform them about changes in golosvyh boxes. This also means that if change the contents of the voice mail from outside, such as web intferfeysom to voice mail, Asterisk still have to be Periodically check the contents of voice mailboxes.

[VOIP-info.org Wiki Changes] Asterisk at large - voip-info.org: OpenSER is pure sip server so and it is light wieght binary so it can handle 10,000 call per second so i am useing OpenSER front of Asterisk so openser handle my call SIP registration request and my PSTN functionality done by Asterisk caz OpenSER not support telephpny hardware. Asterisk provide media application services like VoiceMail, Confrance, IVR, PSTN function.

[It's enough to be on your way...] It's enough to be on your way...: Asterisk/Exchange Voicemail: When you call OVA (Outlook Voice Access), you can say voicemail, email, calendar, address book. Deleting a voicemail in OVA clears from Outlook and vice versa.

[VOIP-info.org Wiki Changes] Asterisk phone snom - voip-info.org: I wanted to keep the DND status and display on my snom phones in line with my DND records in my Asterisk database. I wanted this for two reasons a) it makes sense to the user to keep the DND indication on the phone in line with the behavior of the dialplan and b) when routing an incoming PSTN call to a number of Sip Phones having the actual phone in DND makes life easier for my dialplan logic (I can just do Dial(SIP/112&SIP/113&SIP/114 without having to worry about handling such a group call in stdexten).

[Nerd Vittles] Nerd Vittles » It's Orgasmatron 5.1: The Ultimate Asterisk Kitchen ...: You'll need four pieces of information: your 10-digit Google Voice phone number, your Google Voice account name (which is the email address you used to set up your GV account), your GV password (no spaces!), and your 11-digit SIPgate RingBack DID (beginning with a 1).

[Skype Mobile] Skype - Skype Mobile blog - Skype for iPhone: Update for iOS 4 ...: I don't know why this happens but all the skype chat alternatives like nimbuzz, im+, truphone etc all sign you out of skype after so long BUT other im platforms like msn and google talk don't - even when run through third party clients - why is this, is there a technical reason or is it because skype are a bunch of ******??

[The iPhone Blog] How To: Disable 3G on the iPhone 3G For More Talk but Less Speed ...: I just got my fido iPhone 3G which i placed an order for on the 11th July on the phone, the agent that served me that day told me that i could disable the data connection on their end and they have done that for me, but she told me that some apps would still require 3G network so if i disable it, i would not be able to access some apps, like visual voicemail, which is fine for me since i have caller ID and hate using voicemail. So i plug in it my computer, sync everything, went in to Settings >

[iPhone Download Blog] How to Use an iPhone Without the Data Plan: I tried the steps, it works just fine”¦disabled edge and got everything working WiFi onlyn no edge, e-mail, texting, voice mail still working fine. Got an e-mail saying that I needed to call customer service because I did not have a qualified .

[Comments for Nerd Vittles] Nerd Vittles » Newbie's Guide to Asterisk@Home 2.4: Unabridged ...: So I’m trying to get switched over to Asterisk@home using 2 quad span Digium cards for our old Proprietary PBX, and when the behavior switched from the ringall after hours behavior to the hunt group this morning (just put in in kind of hybrid style for testing last night) and noticed that we were not recieving any calls on the KSU (FXS’s were being used so that the asterisk box had control, but the KSU users didn’t notice too much). I loooked in the log file and noticed a syntax error durring the incomming calls that didn’t ever ring.

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