We felt that it would be good to let you know about some minor changes happening with our community services.For quite some time, weve had a consolidated authentication server for most of our community services. This means that you use the same usern..
Sorry for a possible retransmit: the first was sent from an incorrect email address.Im trying to use the Polycom SoundStation IP 7000 with Confbridge.But the transcoding from siren14 to slin32 is via slin.First, it seems odd that theres no transco..
We found this URL: http://sourceforge.net/projects/asteriskvideo/But these applications seem too old for Asterisk 11.Are there any video applications for Asterisk 11?We need these applications to implement IVVR.Or any other solution is to be appreciated.Tha..
I have a question about management tool for asterisk. What tool are you used to use to manage asterisk? I have a system with freePBX and I just want to know if is there any tool that better ..
I have the following scenario. A small network has DHCP but does not publish option 66. An Asterisk server is on the network, but the Asterisk version does not support DPMA and it is hard to switch the version. However, there is a possibility to h..
all,Im getting compilation error as trying to install latest version of dahdi on CentOS box 5.9 which I now updated from 5.6. I also installed the dependencies but still not getting the clue to get install the driver. Listing down the errors below..
,I have a query ,basically i use three server for own call center. The server A and B i have configure the 60-60 channel each server. Server A and B(or call transfers into server X) calls hitting into server X.Both the server have contain same CLI m..
Im connecting a Polycom SoundStation IP 7000 and trying to use siren14. I downloaded the codecs and now it will properly transcode to connect to other phones and play any files that are in .wav format.But when it tries to play any files with .sire..
everyone,Im having a hard time figuring this issue out, we just switched from a T1 PRI to a SIP trunk provider and thats when the issue started. Now when someone forwards all calls on their phone to a cellphone, when a customer calls in, Asterisk correc..