asterisk users - Nov 2012 - Can you help me to use SIPML5 with Asterisk ?

Lionel BEAUDOIN wrote:
> Hello,
Hola,
> I saw your email in a forum message, can you help me, I try to use
> SIPML5 with an Asterisk 11 server ?
>
> My Asterisk server is installed on a Debian server.
> I have download all the sources from sipml5.org
Please ensure you have followed the instructions at 
https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support to 
set up the Asterisk side of things for WebSocket.
> I have modifiied call.htm to target the requests on my server.
>
> - If I use the port 5060, I can register but I cant emet calls
> - If I use the port 8088, I can't register.
>
> I think it's because I don't use the WS protocol but when I watch
the
> request on the 8088 port with tcpdump, I see that transport is UDP.
>
> How can I define a registring session with WS transport in the call.htm
> file ?
You don't need to use your own copy of sipml5. Point a suitable browser 
to the following URL:

http://sipml5.org/call.htm?svn=9

Go into "Expert Mode" and disable Video support. Use the WebSocket 
Server URL for your server, like below:

ws://<hostname or IP address of Asterisk>:8088/ws

Fill out the rest of the registration details as you normally would.

Display Name: <account name in sip.conf>
Private Identity: <account name in sip.conf>
Public Identity: sip:<account name in sip.conf>@<hostname or IP address
of Asterisk>
Password: <password configured in sip.conf>
Realm: <hostname or IP address of Asterisk>

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.

Cheers,

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at:  www.digium.com  & www.asterisk.org

You can also hardcode these values in call.htm find below lines:

                i_port = 4062 + (((new Date().getTime()) % 5) * 1000);^M
                s_proxy = "sipml5.org";^M

and change them to

                i_port = "<port>/ws";^M
                s_proxy = "ws://<* server IP>:";^M

Change <port> and <* server IP> with required values.

Regards,
Qasim


On Wed, Nov 7, 2012 at 7:52 PM, Joshua Colp <jcolp at digium.com> wrote:
> Lionel BEAUDOIN wrote:
>
>> Hello,
>>
>
> Hola,
>
>  I saw your email in a forum message, can you help me, I try to use
>> SIPML5 with an Asterisk 11 server ?
>>
>> My Asterisk server is installed on a Debian server.
>> I have download all the sources from sipml5.org
>>
>
> Please ensure you have followed the instructions at
>
https://wiki.asterisk.org/**wiki/display/AST/Asterisk+**WebRTC+Support<https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support>to
set up the Asterisk side of things for WebSocket.
>
>  I have modifiied call.htm to target the requests on my server.
>>
>> - If I use the port 5060, I can register but I cant emet calls
>> - If I use the port 8088, I can't register.
>>
>> I think it's because I don't use the WS protocol but when I
watch the
>> request on the 8088 port with tcpdump, I see that transport is UDP.
>>
>> How can I define a registring session with WS transport in the call.htm
>> file ?
>>
>
> You don't need to use your own copy of sipml5. Point a suitable browser
to
> the following URL:
>
> http://sipml5.org/call.htm?**svn=9 <http://sipml5.org/call.htm?svn=9>
>
> Go into "Expert Mode" and disable Video support. Use the
WebSocket Server
> URL for your server, like below:
>
> ws://<hostname or IP address of Asterisk>:8088/ws
>
> Fill out the rest of the registration details as you normally would.
>
> Display Name: <account name in sip.conf>
> Private Identity: <account name in sip.conf>
> Public Identity: sip:<account name in sip.conf>@<hostname or IP
address of
> Asterisk>
> Password: <password configured in sip.conf>
> Realm: <hostname or IP address of Asterisk>
>
> 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.
>
> Cheers,
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at:  www.digium.com  & www.asterisk.org
>
> --
> ______________________________**______________________________**_________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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