<div dir="ltr"><div><div>On 29 July 2013 16:55, Kevin Larsen <span dir="ltr"><<a href="mailto:kevin.larsen@pioneerballoon.com" target="_blank">kevin.larsen@pioneerballoon.com</a>></span> wrote:<br></div></div><div class="gmail_extra">
<div class="gmail_quote"><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
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<br><font size="1" color="#5f5f5f" face="sans-serif">From:
</font><font size="1" face="sans-serif">Steve Davies <<a href="mailto:davies147@gmail.com" target="_blank">davies147@gmail.com</a>></font>
<br><font size="1" color="#5f5f5f" face="sans-serif">To:
</font><font size="1" face="sans-serif">Asterisk Users Mailing
List - Non-Commercial Discussion <<a href="mailto:asterisk-users@lists.digium.com" target="_blank">asterisk-users@lists.digium.com</a>>,
</font>
<br><font size="1" color="#5f5f5f" face="sans-serif">Date:
</font><font size="1" face="sans-serif">07/29/2013 10:53 AM</font>
<br><font size="1" color="#5f5f5f" face="sans-serif">Subject:
</font><font size="1" face="sans-serif">[asterisk-users]
Connected Line presentation in 1.8.x upwards</font>
<br><font size="1" color="#5f5f5f" face="sans-serif">Sent by:
</font><font size="1" face="sans-serif"><a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a></font>
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<br><font size="3">Hi,</font>
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<br><font size="3">I've searched the </font><a href="http://asterisk.org/" target="_blank"><font size="3" color="blue"><u>asterisk.org</u></font></a><font size="3">
and voip-info wiki sites, but not found an answer that seems to match.</font>
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<br><font size="3">Hopefully this is a simple question. COLP is working very
well on our system - Unfortunately it is working a bit TOO well in some
circumstances. We have some "untrusted" trunks. On these trunks,
an initial CallerID can be used, but any redirected caller numbers, COLP
updates etc are not safe to accept. Sadly I cannot find how to cause COLP
updates to be ignored for a trunk.</font>
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<br><font size="3">I need solutions for SIP, IAX and DAHDI, what options
do I have? This applies to both in- and out-bound calls.</font>
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<br><font size="3">Are there some variables that I can set just before dialling
an outbound call, and immediately on receiving an inbound call to determine
what the callerID values will be for the entire duration of the call? (ie.
old-style pre-COLP behaviour for specific trunks)</font>
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<br><font size="3">Thanks for any pointers.</font>
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<br><font size="3">Regards,</font>
<br><font size="3">Steve</font>
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<br></div></div><font face="sans-serif">I believe what you are looking for in
Dial is the 'I' option.</font>
<br><br><br></blockquote><div class="gmail_quote"><br></div>Ah. Many thanks.<div><br></div><div>It appears that the normally reliable voip-info wiki is out of date and does not include that option. I should probably have just used Asterisk's built-in documentation anyway :)<div>
<br></div><div>I guess on an inbound call I will have to conditionally set 'I' on the Dial command based on the originating channel?</div><div><br></div><div>I will also have to go and check what affect this has when a call is SIP REFER'ed as that might result in an asymmetric requirement. The internal SIP handset will want updating, but the external SIP trunk call will not.</div>
<div><br></div></div><div>Regards,</div><div>Steve</div><div><br></div></div></div></div>