Headless sip client

Asterisk Asterisk General
1

Hi,

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,

Angelo

2

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).

3

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.

4

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.

5

ok, i did following:

[default]
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?

Thanks

6

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.