[asterisk-users] Using Local Asterisk Server with Siphon - Can't hear voice issue

[Tharindu Madushanka]

Hi,

It worked finally with GSM Codec only enabled at client side.. Initially
with G.711 (u-low) , G.711 (A-low) and GSM it didn't work. All enabled

by setting [CLI] sip set debug on
I saw asterisk having following logs..

    -- Remotely bridging SIP/macbook-00000041 and SIP/tharindu-00000042
set_destination: Parsing <sip:tharindu at 192.168.1.3<sip%3Atharindu at 192.168.1.3>
:
64540;ob> for address/port to send to
set_destination: set destination to 192.168.1.3:64540
Audio is at 5060
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP


-- 

So I enabled GSM only .. then everything got solved.. :)

---

I would like to register SIP users to the server using kind of web service..
instead of manually entering extensions and users using configuration
files..

TO achieve this could some body point some instructions. ??

One more thing..

Is it possible to automatically reload the servers after some small time ??
instead of manually typing the command on [CLI] console. ??

Thanks and Kind Regards,

Tharindu Madushanka,
tharindufit.wordpress.com
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