<br><br><div class="gmail_quote">On Tue, Jul 26, 2011 at 12:37 AM, Nikhil <span dir="ltr"><<a href="mailto:d.nikhil@cem-solutions.net">d.nikhil@cem-solutions.net</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<br>
I am using asterisk as a client not as a server. For client I need features like transfer ,call forward ,multiple lines as in normal IP Phones like CISOC,polycom.<br>
<br>
In asterisk ,we have chan_alsa driver that will communicate to the local soundcard. If I installed asterisk in my ubuntu system,and using CLI command I can make calls outside and once call connected I can hear and talk from my Headphone.<br>
<br>
I planing to enhance chan_alsa module to get the features same as in SIP client.<br>
<br>
Thanks<br><font color="#888888">
Nikhil</font><div><div></div><div class="h5"><br>
<br>
On 07/26/2011 12:57 AM, Duncan Turnbull wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Asterisk can run operator phones with no problem, there are multiple phones out there with addon buttons for automating shared line appearances forwards and other functions<br>
<br>
For example yealink have the t38 with 6 lines and 16 buttons and the ex 38 with 38 additional programmable buttons to add to that if you need<br>
<br>
Are you talking about a phone that is not sip based?<br>
<br>
I am not sure why you need to use chan_alsa?<br>
<br>
Cheers Duncan<br>
<br>
Sent from my iPhone please excuse the typos<br>
<br>
On 25/07/2011, at 12:30 AM, Nikhil<<a href="mailto:d.nikhil@cem-solutions.net" target="_blank">d.nikhil@cem-solutions.<u></u>net</a>> wrote:<br>
<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Any reply on this......<br>
<br>
On 07/22/2011 12:56 PM, Nikhil wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Hi<br>
Does anyone used asterisk as a operator phone,with multiple lines and features like transfer forward and etc.I used chan_alsa driver to make asterisk as SIP Phone,but it has limitation,we cant make or receive multiple calls,and will not able to do any features like transfer forward etc. Is any other application available in asterisk to do this .<br>
<br>
Thanks<br>
Nikhil<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>
<br>
<br>
</blockquote>
<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>
--<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>
<br>
<br>
</blockquote>
<br>
<br></div></div>
--<div><div></div><div class="h5"><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>
</div></div></blockquote></div><br><div>Using asterisk as a client sounds interesting. I guess all the existing sip clients suck?</div><div><br></div><div>-Kyle</div>