Hello,<br><br><br>I'm partner manager at snom. The SI-90 and SI-120 ARE NOT snom phones. <br><br>Our Indian joint venture is selling these! <br>These phones may not be called snom, that's why their name is SI-90 and SI-120.
<br><br>The phones are not enginieered and developed in Germany as the normal snom300 series. <br>Neither the software nor the hardware was not developed by snom!<br><br>I'm not happy with the situation, but I may ask you to directly contact snom india for support. For me this is a clear bug and should be fixed by them.
<br><br>I will in parallel contact snom India, so that they take care about the issue.<br><br>In future if you want to get a reliable and working phone, buy a snom from the actual 300 series. I developed Asterisk solutions myself in the past.
<br><br>Transfer, Music on Hold, Busy lamp field, etc. all works smooth and hassle free.<br><br><br>Regards<br><br>Tim Koehler<br><br><br><br><br><div><span class="gmail_quote">On 7/20/07, <b class="gmail_sendername">Keshav K.
</b> <<a href="mailto:kesh.keshav@yahoo.com">kesh.keshav@yahoo.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><font style="font-family: times new roman;" size="3">
<span style="color: rgb(0, 0, 127);">There is one thing,</span><br style="color: rgb(0, 0, 127);"><span style="color: rgb(0, 0, 127);">just forget that your phone is a snom phone or whatever...</span><br style="color: rgb(0, 0, 127);">
<br style="color: rgb(0, 0, 127);"><span style="color: rgb(0, 0, 127);">simply to make a blind call transfer press #8, according to the my feature.conf, default it is #, or you can assign it any, then after pressing that you will listen a IVR "transfer" and dial the desired number followed by the # sign, then you will connect to the new number, now hangup your phone, and the other two will be connected.
</span><br style="color: rgb(0, 0, 127);"><br style="color: rgb(0, 0, 127);"><span style="color: rgb(0, 0, 127);">But make sure, that in your extensions.conf you should have the entry for "t", as I have showed in the entry..
</span><br style="color: rgb(0, 0, 127);"><br style="color: rgb(0, 0, 127);"><span style="color: rgb(0, 0, 127);">Regards,</span><br style="color: rgb(0, 0, 127);"><span style="color: rgb(0, 0, 127);">Keshav</span></font>
<div><span class="e" id="q_113e2c5c76958a74_1"><br><br><br><br><b><i>satish patel <<a href="mailto:satish_patel_2000_2000@yahoo.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">satish_patel_2000_2000@yahoo.com
</a>></i></b> wrote:</span></div><blockquote style="border-left: 2px solid rgb(16, 16, 255); margin-left: 5px; padding-left: 5px;"><div><span class="e" id="q_113e2c5c76958a74_3"> but what buttons i will use for call transfer ??? I have SNOM SI 120 phon with transfer button so how to do it ?
<br><br><b><i>"Keshav K." <<a href="mailto:kesh.keshav@yahoo.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">kesh.keshav@yahoo.com</a>></i></b> wrote: <blockquote style="border-left: 2px solid rgb(16, 16, 255); padding-left: 5px; margin-left: 5px;">
<span style="color: rgb(0, 0, 127);"><span style="font-family: comic sans ms;"><font size="3">Hi,<br>To use call tranfer you have to make entry in extension.conf...<br><br>exten => _7.,1,Dial(SIP/${EXTEN},20,Ttr)<br><br>
then in feature.conf----<br><br>[featuremap]<br>blindxfer =>
#8 ; Blind transfer (default is #)<br>;disconnect => *0 ; Disconnect (default is *)<br>;automon => *1 ; One Touch Record a.k.a. Touch Monitor<br>atxfer => #9 ; Attended transfer
<br>parkcall => #72 ; Park call (one step parking)<br><br>I'm using this...end its working wonderfully.<br><br>--Keshav<br><br></font></span></span><br><b><i>satish patel <<a href="mailto:satish_patel_2000_2000@yahoo.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
satish_patel_2000_2000@yahoo.com</a>></i></b> wrote: <blockquote style="border-left: 2px solid rgb(16, 16, 255); padding-left: 5px; margin-left: 5px;">Dear
all<br><br> I have beginer in Voip and i have configured Asterisk server with 100 IP SIP phone ( SNOM ) everything is fine but problem is how to transfer call from one user to other means i call to some one and then someone want to transfer call to another person how it is possible i have also try with
feartue.conf but it is now working i have also read document on voip-info website but now clear yet can anyone explain me how to asterisk transfer call from one user to other and what extention.conf look like is there any change in
sip.conf or extention.conf <br><br><br>Rgd<br><br>Satish patel<br> <div></div> <hr size="1"> <a href="http://us.rd.yahoo.com/evt=49938/*http://tools.search.yahoo.com/toolbar/features/mail/" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
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http://lists.digium.com/mailman/listinfo/asterisk-users</a></blockquote><br><br><br><font style="color: rgb(0, 0, 127); font-family: comic sans ms;" size="3">Regards,<br>Kesh<br><span style="font-style: italic;">" Lets change the future...lets change the world."
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http://lists.digium.com/mailman/listinfo/asterisk-users</a><br></blockquote></div><br><br clear="all"><br>-- <br>---------------------------------------------------<br>snom technology AG<br><br>Tim Koehler <br>Partner Manager
<br><a href="mailto:tim.koehler@snom.de">tim.koehler@snom.de</a>