need some help in Asterisk configuration.
Installed asterisk-18.104.22.168 on Ubuntu 14.04 64 bit got an error missing app_hangup.so file while running the configure command. removed the file from modules.conf and rerun the configure.
Installed two zoiper sip clients and able to register to the server.
Need some help in configuring the extensions.conf to include the dial plan to make a call between the two sip clients installed in two ubuntu PCS in the same subnet. (One client in the same PC as the Asterisk server.
and also need some help in configuring the voicemail server.
The sip.conf file is having entries as below:
Please, never tail end an existing thread with your new thread.
allowguest=yes is normally bad practice.
type=peer is better than type=friend.
If you are sharing a machine, you will need to explicitly specify port numbers for at least one side, as Asterisk and the phone cannot both use 5060. I would advise against newbies using soft phones, as they tend to have bugs and restrictions that are not present in hard phones.
modules.conf doesn’t exist when configure is run, so can’t be used as an input to configure.
There is nothing special about the extensions.conf contents for calling between the phones, so please read the available documentation. You need to show us your existing attempts at using voicemail before we can go beyond what you should get from the documentation.