I have a linux server that is running Asterisk on a school campus. I have configured that server to make inside calls to different extensions and have set up various phones and mailboxes. I would now like to connect to the outside world.
Half the campus is already VOIP; therefore, there are VOIP and SIP servers on campus that are connected to the outside network. We are still in the testing stages with this machine though.
How would I go about connecting my Asterisk server to an already running SIP machine to connect to make outside calls that way? I have the IP address and the port number of the machine I am supposed to connect to. I would greatly appreciate any responses with ideas you may have. I am new to Asterisk but am trying to learn as much as I can.
Do I need to do anything with the Zap channels? I do not have any hardware cards installed on this machine itself, that is why I am trying to connect to another server to go through.
Hi
Ths is Reagrading connect Asterisk to Sip out side world to make call with voip
Asterisk server--------->Sip Server like ser or openser(softswtich)–>out
you need to set sip.conf this way
[general]
bindport=5060
bindaddr=2xx.xx.xxx.xx ur Asterisk server ip
maxexpiry=3600
checkmwi=10
vmexten=voicemail
videosupport=yes
rtcachefriends=yes
disallow=all
allow=ulaw
allow=alaw
allow=g729
rtptimeout=60
useragent=IPPBX
dtmfmode=auto
nat=yes
[SWITCH]
type=peer
host=ur sip server ip in campus or ur voip provider
trustrpid=yes
context=main
insecure=very
dtmfmode=auto
other are same as we mention for ur users
In extension.conf
[outcall]
exten => _1XXXXXXXXXX,1,Set(CALLERID(num)=64xxxxxxx)
this will route ur out world call from ur Asterisk to ur Sip server Then sip server route that call to out side as there confoig.
Also u have to Authenticate ur Asterisk ip address at Sip server to make & recivve calls
also may u have to use Register concept in sip.conf if ur sip server dealing with such option .
try this
Thank you very much for that info, I appreciate the help. I have in fact copied what you posted into the sip.conf and extensions.conf files and when I enter asterisk and type ‘sip show peers’, there is another device that is shown called “switch” because of what has been added in sip.conf.
Therefore, my asterisk server is able to communicate with the SIP server and acknowledges it’s host. I am still somewhat uncertain as to how I would go about calling an outside line though. Do I need to specify something like dial ‘9’ first? I think I need more to add to extensions.conf. just not certain what. Anyone have any ideas? Thank You again!