Asterisk WS Error

Asterisk Asterisk Support
1

Hello guys, I’m new to the forum. I’m trying to create an application that through the browser make a conference call. I’ve configured asterisk to accept the call and create a conference, and so far so good. But when I go to test it with the SIPml5 webapp, I get an error on websocket.

my client debug is below:

<--- SIP read from WS:172.18.41.10:62440 --->
REGISTER sip:10.64.1.52 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKiF0L8CBnUlUBdiPGl9QWsBIOiiTXHUvP;rport
From: <sip:>;tag=qcComVt9WT9Tp9RgI9zq
To: <sip:>
Contact: "6002"<sip:null@df7jal23ls0d.invalid;transport=ws>;expires=200;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: e222137d-d16f-df63-30c0-72a1996f9294
CSeq: 42862 REGISTER
Content-Length: 0
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5/v0.0.0000.0
Organization: Doubango Telecom
Supported: path

<------------->
--- (12 headers 0 lines) ---

<--- Transmitting (no NAT) to 172.18.41.10:5060 --->
SIP/2.0 404 Not found
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKiF0L8CBnUlUBdiPGl9QWsBIOiiTXHUvP;rport;received=172.18.41.10
From: <sip:>;tag=qcComVt9WT9Tp9RgI9zq
To: <sip:>;tag=as6087df99
Call-ID: e222137d-d16f-df63-30c0-72a1996f9294
CSeq: 42862 REGISTER
Server: Asterisk PBX 13.9.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
[Jul 14 14:37:12] NOTICE[15774]: chan_sip.c:28446 handle_request_register: Registration from '<sip:>' failed for '172.18.41.10:62440' - Not a local domain

Premise that both the asterisk server that the webapp sipml5 are behind NAT. Asterisk I think I configured well as with ZOIPER I can safely carry the call. With websocket not.

Someone is kind enough to help me?

Regards,

2

The SIP REGISTER appears to be incomplete as the URI in the From header and To header is invalid. They do not contain any address information. This is likely what is causing your problem.

3

Thank you jcolp for the reply.

Have you any suggestion for me? I didn’t :frowning: :frowning: :frowning:

My sip.conf is below

[general]
bindaddr=0.0.0.0
bindport=5060
transport=udp,ws,tls,wss
nat=on,force_rport
externip=10.64.1.52
localnet=192.168.246.13/255.255.255.0

[6002]
host=dynamic
secret=6002
type=friend
callerid="6002" <6002>
context=cmt
transport=udp,ws
directmedia=no
disallow=all
allow=all
dtmfmode=rfc2833
avpf=yes
icesupport=yes
encryption=yes

Please, help me…

4

This would be on the SIPml5 side, not Asterisk. I have no experience with it.

5

Ok, no problem… Another question… In console asterisk i can see

WebSocket connection from '172.18.41.10:51029' for protocol 'sip' accepted using version '13'
[Jul 14 16:52:10] NOTICE[29265]: chan_sip.c:28446 handle_request_register: Registration from '<sip:>' failed for '172.18.41.10:51029' - Not a local domain

How can i resolve the error “not a local domain” ?

6

The issue I mentioned with SIPml5 is likely causing that, and why it’s not working.

7

Ok, perfect!

Anyone can help me? :slight_smile:

8

Hi jcolp, one question… Do you have a basic html application that make call to asterisk? very basic, one button to call, one button to close.

9

No, I don’t have anything like that.