You can also hardcode these values in call.htm find below lines:<br><br> i_port = 4062 + (((new Date().getTime()) % 5) * 1000);^M<br> s_proxy = "<a href="http://sipml5.org">sipml5.org</a>";^M<br>
<br>and change them to<br><br> i_port = "<port>/ws";^M<br> s_proxy = "ws://<* server IP>:";^M<br><br>Change <port> and <* server IP> with required values.<br>
<br>Regards,<br>Qasim<br><div class="gmail_extra"><br><br><div class="gmail_quote">On Wed, Nov 7, 2012 at 7:52 PM, Joshua Colp <span dir="ltr"><<a href="mailto:jcolp@digium.com" target="_blank">jcolp@digium.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Lionel BEAUDOIN wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Hello,<br>
</blockquote>
<br>
Hola,<br>
<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
I saw your email in a forum message, can you help me, I try to use<br>
SIPML5 with an Asterisk 11 server ?<br>
<br>
My Asterisk server is installed on a Debian server.<br>
I have download all the sources from <a href="http://sipml5.org" target="_blank">sipml5.org</a><br>
</blockquote>
<br>
Please ensure you have followed the instructions at <a href="https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support" target="_blank">https://wiki.asterisk.org/<u></u>wiki/display/AST/Asterisk+<u></u>WebRTC+Support</a> to set up the Asterisk side of things for WebSocket.<br>
<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
I have modifiied call.htm to target the requests on my server.<br>
<br>
- If I use the port 5060, I can register but I cant emet calls<br>
- If I use the port 8088, I can't register.<br>
<br>
I think it's because I don't use the WS protocol but when I watch the<br>
request on the 8088 port with tcpdump, I see that transport is UDP.<br>
<br>
How can I define a registring session with WS transport in the call.htm<br>
file ?<br>
</blockquote>
<br>
You don't need to use your own copy of sipml5. Point a suitable browser to the following URL:<br>
<br>
<a href="http://sipml5.org/call.htm?svn=9" target="_blank">http://sipml5.org/call.htm?<u></u>svn=9</a><br>
<br>
Go into "Expert Mode" and disable Video support. Use the WebSocket Server URL for your server, like below:<br>
<br>
ws://<hostname or IP address of Asterisk>:8088/ws<br>
<br>
Fill out the rest of the registration details as you normally would.<br>
<br>
Display Name: <account name in sip.conf><br>
Private Identity: <account name in sip.conf><br>
Public Identity: sip:<account name in sip.conf>@<hostname or IP address of Asterisk><br>
Password: <password configured in sip.conf><br>
Realm: <hostname or IP address of Asterisk><br>
<br>
In the future please send emails of this type to the asterisk-users mailing list so that everyone can see the conversation and learn. I've copied my reply to it.<br>
<br>
Cheers,<br>
<br>
-- <br>
Joshua Colp<br>
Digium, Inc. | Senior Software Developer<br>
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA<br>
Check us out at: <a href="http://www.digium.com" target="_blank">www.digium.com</a> & <a href="http://www.asterisk.org" target="_blank">www.asterisk.org</a><br>
<br>
--<br>
______________________________<u></u>______________________________<u></u>_________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
<a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/<u></u>mailman/listinfo/asterisk-<u></u>users</a><br>
</blockquote></div><br></div>