<html><head></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">Thanks, I tried both suggestions but no love. When I used&nbsp;<span style="color: rgb(148, 49, 192); font-family: Menlo; font-size: 11px; background-color: rgb(255, 255, 255); "><b>sip_h_In-Reply-To</b></span><span style="color: rgb(148, 49, 192); font-family: Menlo; font-size: 11px; background-color: rgb(255, 255, 255); "><b>=${sip_call_id} </b></span>used as below, the trace showed all normal but the cell phone does not ring at all. When I removed everything the cell rang but the original number was not passed. BTW, the sip_call_id was translated to (#s altered):<div><a href="mailto:sip_h_In-Reply-To=3912345-9123456295-612341@msw1.telengy.net">sip_h_In-Reply-To=3912345-9123456295-612341@msw1.telengy.net</a>, could that be an issue with ATT not liking it?</div><div><br></div><div>Mario G<br><div><br></div><div><br><div><div>On Jan 27, 2013, at 3:02 PM, Richard Brady wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite">Ok, nifty. They are letting you present a number you do not own as Caller ID on an outbound call if that outbound call is a forwarded leg of an inbound call.&nbsp;<div><br></div><div>They do this by looking the In-Reply-To header of the INVITE for the forwarded leg, which should contain the Call-ID of the orignal leg.&nbsp;</div>
<div><br></div><div>So you need to copy the <b>Call-ID</b> in order to authorize the <b>Caller ID</b>.</div><div><br></div><div>A couple things:</div><div><br></div><div>1. From the docs: <i>effective_caller_id_name Sets the effective callerid name. This is automatically exported to the B-leg; however,<b> it is not valid in an origination string</b>. In other words, set this before calling bridge, otherwise use <b>origination_caller_id_name</b></i></div>
<div><div><br></div><div>2. You shouldn't care about 1 above as it should be copied across from the A leg by default and you are not modifying it, so remove&nbsp;effective_caller_id_name and don't bother with&nbsp;origination_caller_id_name either.</div>
<div><br></div><div>3. You should use sip_h_ not sip_rh_ because you want the header in the new INVITE going out.</div><div><br></div><div>Perhaps try:</div><div><div><br></div><div><span style="font-family:Menlo;font-size:11px;background-color:rgb(255,255,255);color:rgb(183,40,0)">&lt;</span><span style="font-family:Menlo;font-size:11px;background-color:rgb(255,255,255)">action application=</span><span style="color:rgb(148,49,192);font-family:Menlo;font-size:11px;background-color:rgb(255,255,255)">"bridge"</span><span style="font-family:Menlo;font-size:11px;background-color:rgb(255,255,255)">&nbsp;data=</span><span style="color:rgb(148,49,192);font-family:Menlo;font-size:11px;background-color:rgb(255,255,255)">"{originate_timeout=45,</span><span style="color:rgb(148,49,192);font-family:Menlo;font-size:11px;background-color:rgb(255,255,255)">alert_info=n=${lua_ringtone}}$</span><span style="color:rgb(148,49,192);font-family:Menlo;font-size:11px;background-color:rgb(255,255,255)">{group_call(bria@${domain_</span><span style="color:rgb(148,49,192);font-family:Menlo;font-size:11px;background-color:rgb(255,255,255)">name}+A)},${group_call(</span><span style="color:rgb(148,49,192);font-family:Menlo;font-size:11px;background-color:rgb(255,255,255)">deskphone@${domain_name}+A)},[</span><span style="color:rgb(148,49,192);font-family:Menlo;font-size:11px;background-color:rgb(255,255,255)">leg_delay_start=20,leg_</span><span style="color:rgb(148,49,192);font-family:Menlo;font-size:11px;background-color:rgb(255,255,255)">timeout=23,<b>sip_h_In-Reply-To</b></span><span style="color:rgb(148,49,192);font-family:Menlo;font-size:11px;background-color:rgb(255,255,255)"><b>=${sip_call_id}</b>]</span><span style="color:rgb(148,49,192);font-family:Menlo;font-size:11px;background-color:rgb(255,255,255)">sofia/gateway/${dial_gateway}/</span><span style="color:rgb(148,49,192);font-family:Menlo;font-size:11px;background-color:rgb(255,255,255)">19161234567"</span><span style="font-family:Menlo;font-size:11px;background-color:rgb(255,255,255);color:rgb(183,40,0)">/&gt;</span></div>
<div><br></div><div>Hope this helps.&nbsp;</div><div><br></div><div>Richard</div><div><br></div><div><br><div class="gmail_quote">On 21 January 2013 19:54, Mario G <span dir="ltr">&lt;<a href="mailto:mario_fs@mgtech.com" target="_blank">mario_fs@mgtech.com</a>&gt;</span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div style="word-wrap:break-word">Thanks, apparently I had it wrong, the doc below states that the PBX must support it incoming, they pointed me to using effective_caller_id which I added to the bridge but it still does not work. &nbsp;Would love to fix this since the cell phones currently have no idea who is calling.<div>
Mario G<br><div><br></div><div><div style="margin-top:0px;margin-right:0px;margin-bottom:0px;margin-left:0px;font:normal normal normal 11px/normal Menlo;color:rgb(148,49,192)"><span style="color:#b72800">&lt;</span><span style="color:#000000">action application=</span>"bridge"<span style="color:#000000"> data=</span>"{originate_timeout=45,alert_info=n=${lua_ringtone}}${group_call(bria@${domain_name}+A)},${group_call(deskphone@${domain_name}+A)},[leg_delay_start=20,leg_timeout=23,effective_caller_id_number=${caller_id_number}]sofia/gateway/${dial_gateway}/19161234567"<span style="color:#b72800">/&gt;</span></div>
<div><span style="color:#b72800"><br></span></div><div><ul style="text-indent:0px;letter-spacing:normal;font-variant:normal;text-align:-webkit-auto;font-style:normal;font-weight:normal;padding:0px;line-height:normal;text-transform:none;font-size:13px;white-space:normal;margin:0px 1px 0px 16px;font-family:Verdana,Arial,Tahoma,sans-serif;word-spacing:0px">
<li style="font-size:12pt;font-weight:bold;padding-bottom:28px"><p style="font-size:10pt;color:rgb(95,95,95);padding:6px 0px 0px;font-weight:normal">Please note that this feature is ONLY AVAILABLE for customers using a SIP PBX that either supports (or allows the configuring of) the "in-reply-to" header (defined by RFC 3261) for incoming calls which are forwarded to an outbound trunk. In these instances Callcentric will "Pass-Through" the CallerID from the original call which was received to the outbound bridged/forwarded call.</p>
</li></ul></div><div><div><div><div class="h5"><div>On Jan 19, 2013, at 5:13 PM, Richard Brady wrote:</div><br></div></div><blockquote type="cite"><div><div class="h5"><div>On 20 January 2013 00:06, Mario G <span dir="ltr">&lt;<a href="mailto:mario_fs@mgtech.com" target="_blank">mario_fs@mgtech.com</a>&gt;</span> wrote:</div>
<div><div class="gmail_quote"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div style="word-wrap:break-word"><div style="margin-top:0px;margin-right:0px;margin-bottom:0px;margin-left:0px;font:normal normal normal 11px/normal Menlo;color:rgb(42,126,49)">I never did this so I must be missing something, I tried both below but the bridge then fails. Can anyone shed some light on what I am doing wrong. My ITSP now supports in-reply-to so I can pass the caller ID to a forwarded call from FS.</div>

</div></blockquote><div><br></div><div>In-Reply-To should contain a Call-ID not a caller ID. They are very different.<div><br></div><div>The following would make a bit more sense, but still not a lot:</div><div><br></div>

<div>&lt;action application="set" data="sip_rh_In-Reply-To=${sip_call_id}"/&gt;<br><br>Using In-Reply-To in a response doesn't seem right to me. I would expect it to appear in an INVITE. For for example, you get a missed call and you call the person back, then the INVITE for the callback would have a new Call-ID but the original Call-ID in the In-Reply-To header. That said, I have no idea what your ITSPs intended use for the header is.&nbsp;</div>

</div><div><br></div><div>Richard</div></div></div></div></div>
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