Does anyone know if Gigaset is for sale in the USA?Based on my assessment of phones and features, i would like to try the N300IP base along with C610H phones. I can only find the handsets on ebay, no retailers in USA.And I suspect they are using Europ..
When a call comes in and is parked using the Park() command it appears that the billing seconds do not include the time while the caller was parked. Is there a simple way to correct this with an option or setting in asterisk. This is not acceptable..
Weve deoplyed a number of pure VoIP wireless (wifi & proprietary) phones,but not dect. Is there a simple overview of integrating DECT phones with Asterisk somewhere?I assume the DECT basestation has a multi-account SIP VoIP interface, and the hands..
I work for a VoIP provider in Southern California. We are looking for someone very knowledgeable in Asterisk/VoIP to help work on the following: – Maintenance of current Asterisk servers, updating Asterisk, monitoring load, and other sysadmin task..
Has anyone seen Dahdi dropping incoming calls with Hangup cause 27?
It only drops whilst we are on the phone?
Its not every single call
In article <4FECCD0C.firstname.lastname@example.org>, James Sharpwrote: > On 6/28/2012 3:53 PM, Ernie Dunbar wrote: > > We have three Asterisk servers, Voip1, Voip2, and Voip3. Voip1 has a > > Wildcard TE405P (3rd Gen), Voip2 has a Wildcard TE410P/TE405P (..
I have been fighting all night with version 1.8 and have not found a way to do this with any command or Perl AGI->command. I need to play a file and wait until the customer presses at least $maxdigits to return, BUT, the file must continue playing un..
Does anyone have any experience of connecting SIP phones to an asterisk
server through the 2701HGV router that BT supply with their Infinity
Thanks in Adva..
I have a bunch of different customers on an Asterisk Box (the PBX). This Asterisk Box is behind another Asterisk box that provides a PSTN connection. Up to this point Ive been using IAX between the 2 Asterisk boxes, but I would like to use SIP inste..
Im currently running Asterisk 10.5.1, compiled from source, and just had someone call saying they couldnt get their voice mail. Looking into the users voice mail folder, I saw a .lock file. Removing this file, enabled them to get voice mail. Is anyb..