I'd suggest using different user names and getting asterisk to handle the cleverness... And, well, doing it this way is pretty simple, straight forward, basic asterisk functionality...<br><br>Trying to get two different instances registered as the same user, is, as you've found out, not going to be trivial to implement... It's just not how it works...<br>
<br>If you implemented what I suggested below, using two different usernames for the two ports on the SPA, it would just work...<br><br>d<br><br><div class="gmail_quote">2009/8/5 Faheem <span dir="ltr"><<a href="mailto:faheem_imt@yahoo.com">faheem_imt@yahoo.com</a>></span><br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><table border="0" cellpadding="0" cellspacing="0"><tbody><tr><td style="font-family: inherit; font-style: inherit; font-variant: inherit; font-weight: inherit; font-size: inherit; line-height: inherit; font-size-adjust: inherit; font-stretch: inherit;" valign="top">
<br>By placing OPENSIP in front of Asterisk, we can register multiple
accounts, and we can successfully make call for Outgoing only. But in
case of incoming it fails. <br>
<br>
If two users are registered with asterisk or OpenSIP then the user that
is registered latest is considered to be valid, and he is able to make
calls, other user with earlier registration can not make call.<br>
My point here is in chain_sip.c what are variables or structure that
need to maintain so that we can consider all registered users as active
users.<br>
<br>Thanks!<br>Faheem<br><br>--- On <b>Wed, 8/5/09, D Tucny <i><<a href="mailto:d@tucny.com" target="_blank">d@tucny.com</a>></i></b> wrote:<br><blockquote style="border-left: 2px solid rgb(16, 16, 255); margin-left: 5px; padding-left: 5px;">
<br>From: D Tucny <<a href="mailto:d@tucny.com" target="_blank">d@tucny.com</a>><br>Subject: Re: [asterisk-users] how to implement CLONED LINE Feature in asterisk?<br>To: "Asterisk Users Mailing List - Non-Commercial Discussion" <<a href="mailto:asterisk-users@lists.digium.com" target="_blank">asterisk-users@lists.digium.com</a>><br>
Date: Wednesday, August 5, 2009, 11:06 AM<div><div></div><div class="h5"><br><br><div><div class="gmail_quote">2009/8/4 Faheem <span dir="ltr"><<a rel="nofollow" href="http://mc/compose?to=faheem_imt@yahoo.com" target="_blank">faheem_imt@yahoo.com</a>></span><br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<table border="0" cellpadding="0" cellspacing="0"><tbody><tr><td style="font-family: inherit; font-style: inherit; font-variant: inherit; font-weight: inherit; font-size: inherit; line-height: inherit; font-size-adjust: inherit; font-stretch: inherit;" valign="top">
how to implement CLONED LINE Feature in asterisk<br><br>Hey, I want to implement Clone Line feature in asterisk. I am using SPA-2100.<br>The feature should work in this way.<br><br>There are two ports in the SPA-2100 both are registered with asterisk with same username/password, and have the same (phone number)<br>
<br><br>
<center>
<table border="1" cellpadding="1" cellspacing="1" width="561">
<col width="81">
<col width="91">
<col width="216">
<col width="158">
<tbody><tr>
<td width="81">
<p align="center"><br>
</p>
</td>
<td width="91">
<p align="center"><font face="Times New Roman, serif"><font size="3"><b>No
one on the phone </b></font></font>
</p>
</td>
<td width="216">
<p align="center"><font face="Times New Roman, serif"><font size="3"><b>One
phone in use </b></font></font>
</p>
</td>
<td width="158">
<p align="center"><font face="Times New Roman, serif"><font size="3"><b>Both
phones in use </b></font></font>
</p>
</td>
</tr>
<tr>
<td width="81">
<p align="center"><font face="Times New Roman, serif"><font size="3"><b>Incoming
Calls </b></font></font>
</p>
</td>
<td width="91">
<p align="center"><font face="Times New Roman, serif"><font size="3">Both
phones ring </font></font>
</p>
</td>
<td width="216">
<p align="center"><font face="Times New Roman, serif"><font size="3">Phone
in use receives call waiting notification, unused phone rings </font></font>
</p>
</td>
<td width="158">
<p align="center"><font face="Times New Roman, serif"><font size="3">Both
phones receive call waiting notification </font></font>
</p>
</td>
</tr>
<tr>
<td width="81">
<p align="center"><font face="Times New Roman, serif"><font size="3"><b>Outgoing
Calls </b></font></font>
</p>
</td>
<td width="91">
<p align="center"><font face="Times New Roman, serif"><font size="3">Both
phones can call out </font></font>
</p>
</td>
<td width="216">
<p align="center"><font face="Times New Roman, serif"><font size="3">The
unused phone can call out </font></font>
</p>
</td>
<td width="158">
<p align="center"><font face="Times New Roman, serif"><font size="3">Neither
phone can call out </font></font>
</p>
</td>
</tr>
</tbody></table>
</center>
<br><br> * Inbound:<br> - Both ports will ring. Whichever port is picked up first, will field the call.<br> - Any additional calls that come in would give call waiting notification to the first line, and ring the second line.<br>
- Once the second line is being utilized, all incoming calls will be notifications in the form of call waiting beeps.<br><br> * Outbound:<br> - You will have the ability to dial out from port one.<br> - You will be able to dial a different party on port two.<br>
<br>*** Note ***<br> - If you have an active call on port one, and pick up port two, you will NOT have the same call that is currently active on port one. The Cloned Line will share the same voice mail and will have the same telephone number as the original
telephone line.<br><br> - The Cloned Line is NOT a second telephone number. The telephone number that is assigned to the second phone port on the device is the same telephone number as the number assigned to phone port one. <br>
</td></tr></tbody></table></blockquote></div><br>In sip.conf<br>[line1]<br>username=line1<br>secret=line1password<br>type=friend<br>host=dynamic<br>context=outboundcalls<br>mailbox=1234@default<br><br>[line2]<br>username=line2<br>
secret=line2password<br>
type=friend<br>
host=dynamic<br>
context=outboundcalls<br>
mailbox=1234@default<br>
<br>In extensions.conf<br>[default]<br>exten => 1234,1,NoOp(About to dial both phones)<br>exten => 1234,n,Macro(stdexten,${EXTEN},SIP/line1&SIP/line2)<br>exten => 1234,n,Hangup()<br><br>or for trunk<br>[default]<br>
exten => 1234,1,NoOp(About to dial both phones)<br>
exten => 1234,n,Gosub(stdexten(${EXTEN},SIP/line1&SIP/line2))<br>
exten => 1234,n,Hangup()<br>
<br>then stdexten would be default as comes in the sample configs...<br><br>That should be everything you want if you configure the SPA-2100 to register line 1 with username line1 and line 2 with username line2...<br><br>
d<br>
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