Using Asterix as a SIP client for IVR/call forwarding


#1

Hi All,

I am a * n00b trying to set it up as a SIP client so it can be used as an IVR system, and to forward calls/take messages. So far am having no luck. All i can get it to do is register with the SIP server and thats it. Have created (i think) the correct extensions etc but when i try to call in it goes to my SIP providers mailbox, however if i delete the extensions in extensions.cfg, * complains it cant find it??

====================================
SIP.CONF

[general]
port = 5060 ; Port to bind to
bindaddr = myipaddress ; Address to bind to
context = default ; Default for incoming calls
srvlookup=yes ; Enable SRV lookups on outbound calls
dtmfmode=inband
allow=all ; Allow all codecs

register => myusername:mypassword@voipfone.co.uk/1234

[voipfone.co.uk]
type=user
username=myusername
secret=mypassword
host=voipfone.co.uk
nat=yes
canreinvite=no

[voipfone.co.uk]
type=peer
host=voipfone.co.uk
context=default
nat=yes
canreinvite=no

====================================
EXTENSIONS.CONF

[default]
exten => s,1,Answer()
exten => s,2,Playback(goodbye)
exten => s,3,Hangup()

Thanks for any help.


#2

First off, you have two sip definitions both called voipfone.co.uk You should only have ONE. Since sip definitions inherit the attributes that were previoulsy set, the second one will override the previous definition. In your case, it’s just re-writing the definition altogether, which wipes out the first one completely.

Second. Why just an s extension? Give it a real dial-able exension, (even if it’s just a single digit, like 3) and use your sip client to call extension 3.


#3

Hi,
Thanks for your reply!

I dont have any clients connecting to the * server (yet anyway) I just want the server to answer any incoming external SIP calls that it recieves and show me some sign that its answering them. Once I know that I would like it to play the IVR options to the caller.

Thanks


#4

That’s somewhat odd.

I understand that you just want all incoming calls just answered by the system, but sip setup messages usually require a destination. Sip clients can’t usually dial “null”.

Try giving it a destination extension, and then setup a client (a softwphone would be a good choice) to call the extension.


#5

Hi,

Have already tried that, with no luck =( Ive just deleted all the config files in /asterisk/etc except sip.conf extensions.conf and voicemail.conf and now im getting completely different results.

To be honest i think im in way over my head. Ive read through the manual, and on every website i cant find and nothing seems to do what its supposed to.