What is my SIP URI?

Asterisk Asterisk Support
1

What is the SIP URI for thufir101?

on the client pc:

thufir@doge:~$ 
thufir@doge:~$ sudo sipsak -vv -s sip:thufir101@192.168.1.2

message received:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 127.0.1.1:53035;branch=z9hG4bK.6571652e;alias;received=192.168.1.3;rport=53035
From: sip:sipsak@127.0.1.1:53035;tag=5dbb979d
To: sip:thufir101@192.168.1.2;tag=as3ddfdd14
Call-ID: 1572575133@127.0.1.1
CSeq: 1 OPTIONS
Server: Asterisk PBX 1.8.32.1-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0



** reply received after 0.674 ms **
   SIP/2.0 404 Not Found
   final received
thufir@doge:~$
2

  1. Why?

  2. Asterisk doesn’t need to know the URI to accept incoming calls. Any relevant information will be stored against the peer, not the user, entry.

3

Hi thufir!

Is it this you are after?

http://www.3cx.com/pbx/sip-uri/

You might need some port-forwarding on your firewall/gateway so it know where to find * on your network!

4

[quote=“nypon”]Hi thufir!

Is it this you are after?

http://www.3cx.com/pbx/sip-uri/

You might need some port-forwarding on your firewall/gateway so it know where to find * on your network![/quote]

Yes, I managed to get:

thufir@doge:~$ 
thufir@doge:~$ sudo sipsak -vv -s sip:345@tleilax -m "hi"
No SRV record: _sip._tcp.tleilax
No SRV record: _sip._udp.tleilax
using A record: tleilax
Max-Forwards set to 0

message received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.1.1:53641;branch=z9hG4bK.1742dbb3;alias;received=192.168.1.3;rport=53641
From: sip:sipsak@127.0.1.1:53641;tag=60c31be9
To: sip:345@tleilax;tag=as76dc02d6
Call-ID: 1623399401@127.0.1.1
CSeq: 1 OPTIONS
Server: Asterisk PBX 1.8.29.0-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:192.168.1.2:5060>
Accept: application/sdp
Content-Length: 0



** reply received after 0.827 ms **
   SIP/2.0 200 OK
   final received
thufir@doge:~$

However, if that’s my SIP URI above, how do I configure a client softphone? The asterisk server is on tleilax, the client pc is doge.

If I’m getting “200 OK” that means, I take it, that the connection is ok. So, if the connection is ok, how do I actually use that SIP account?

5

to confirm the connection parameters for a soft phone.

I’ll read more about peers in the O’Reilly, definitive Asterisk.