search for: callcontroller

Has anyone figured this out yet? Lots of places say to add the following to sip.conf of an Asterisk 1.2 system (current production machine/Asterisk as root): tos=0xB8 (Hex B8 = Decimal 184 = Binary 10111000) or if you are running Asterisk v1.4 or newer: tos_sip=cs3 ; Sets TOS for SIP packets. tos_audio=ef ; Sets TOS for RTP audio packets. tos_video=af41 ;

Can anyone comment on the DSCP quality of service settings on your Asterisk server? The network we're setting up has data on the default VLAN, Asterisk server and phones on VLAN 4, and we're using Polycom phones with a PC hooked up to the phone's pass-thru port. What iptables settings are you using on the Asterisk server for DSCP? What are your Polycom DSCP settings? We're using

Hello, I have the following structure: - app - controllers - helpdesk - call_controller.rb (Helpdesk::CallController < ApplicationController) - models - employee.rb - department.rb (Department < ActiveRecord::Base) - helpdesk - call.rb - call_history.rb (Helpdesk::CallHistory < ActiveRecord::Base) Is it possible to have a relationship between call_history and department? The...

Hi every all, since a few weeks I came back to asterisk and tried to install version 1.6. The installation went fine so I decided to buy new dids on Voxbone. I have added the sip peers of Voxbone Belgium1 like this in the sip.conf [81.201.82.39] host=dynamic type=friend insecure=very context=your_context canreinvite=no qualify=no deny=0.0.0.0/0.0.0.0 permit=81.201.82.39/255.255.255.255 but

Hi All, I bought a DID number from VOxbone...this number could be dialed from any PSTN line and could be forwarded to any SIP server like asterisk server...Now I need to forward this number to my asterisk server so when a customer dial this number from his GSM or Land line PSTN number the call will be forwarde to my asterisk server and I need to play a wav file for example.. Can you please give me