Headless sip client


I have Asterisk installed in a headless Ubuntu box and it’s working. I’d like to install a sip client in the same box, its purpose is to receive calls from other branches and I can look at the missed calls by other branches, possible? tips? Thanks,


Asterisk and SIP client (soft sip phone) on same computer is bad idea. Asterisk and sip phone will fight for same port (where they listen for SIP connections).

thanks for the reply, or is there a way, without a sip client, that asterisk can be configured to accept a call ? my need is simple, a client can call a particular number, and i just check the log for the time the call was made.

Of course. Asterisk can ring to some number, and if no answer to Answer the call, then you can play back, record or even use alsa channel - to speak with caller.
You dont need to answer to record who called, when and which extension.

ok, i did following:

exten=> 1000,1,Answer()
exten=> 1000,2,Hangup()

then in CLI, i did:

asterisk*CLI> console dial 1000
[Aug 18 09:40:42] WARNING[4172]: chan_oss.c:485 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory
[Aug 18 09:40:42] NOTICE[4172]: console_video.c:133 console_video_start: voice only, console video support not present
– Executing [1000@default:1] Answer(“Console/dsp”, “”) in new stack
<< Console call has been answered >>
– Executing [1000@default:2] Hangup(“Console/dsp”, “”) in new stack
== Spawn extension (default, 1000, 2) exited non-zero on ‘Console/dsp’
<< Hangup on console >>

looks fine, so I dial 1000 from sip phone, but got this:
== Using SIP RTP CoS mark 5
[Aug 18 09:41:29] NOTICE[797]: chan_sip.c:20039 handle_request_invite: Call from ‘05’ to extension ‘1000’ rejected because extension not found.

any idea why i can call from a console dial but not from a sip client?


If you have GSM - you cant dial to same number from same number.
If you have console - you can dial to console from SIP client, or from console to SIP client, but not from console to console.
And to use console you must have appropriate channel loaded - chan_alsa or something alike.