<html><body><div style="color:#000; background-color:#fff; font-family:HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif;font-size:16px"><div id="yui_3_16_0_1_1418136944274_7902"><span></span></div> <div id="yui_3_16_0_1_1418136944274_7932" class="qtdSeparateBR"><br><br></div><div id="yui_3_16_0_1_1418136944274_7916" style="display: block;" class="yahoo_quoted"> <div id="yui_3_16_0_1_1418136944274_7915" style="font-family: HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif; font-size: 16px;"> <div id="yui_3_16_0_1_1418136944274_7914" style="font-family: HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif; font-size: 16px;"> <div id="yui_3_16_0_1_1418136944274_7931" dir="ltr"> <font id="yui_3_16_0_1_1418136944274_7930" face="Arial" size="2"> El Lunes, 8 de diciembre, 2014 12:51:42, Matthew Jordan <mjordan@digium.com> escribió:<br> </font> </div> <br><br> <div id="yui_3_16_0_1_1418136944274_7913" class="y_msg_container">On Fri, Dec 5, 2014 at 3:23 PM, Rodrigo Montiel<div class="yqt9150597047" id="yqtfd22598"><br clear="none"><<a shape="rect" ymailto="mailto:guevara2309@yahoo.com.ar" href="mailto:guevara2309@yahoo.com.ar">guevara2309@yahoo.com.ar</a>> wrote:<br clear="none">> Hi masters,<br clear="none">><br clear="none">> I’m not an expert on this my friends, but I’m trying to understand which the<br clear="none">> expected behaviour is from Asterisk side when you deal with the following<br clear="none">> scenario:<br clear="none">><br clear="none">> Caller —> GSM Gateway with SIM card A —> Asterisk queue —> extension 1000<br clear="none">><br clear="none">> GSM gateway with call waiting activated on SIM A<br clear="none">> Queue with “skip busy agent” disabled and ringall strategy.<br clear="none">> SIP extension 1000 with call waiting activated, and member of Asterisk<br clear="none">> queue.<br clear="none">><br clear="none">> a) One Caller calls to SIM card A of GSM Gateway, call is forwarded to<br clear="none">> Asterisk queue where SIP extension 1000 answers.<br clear="none">> b) New Caller calls the same SIM card A of GSM gateway (it has call waiting<br clear="none">> activated on the sim card), call is forwarded to Asterisk queue to the same<br clear="none">> extension 1000 and a pop-up appears with the second call.<br clear="none">> c) extension 1000 accepts it so put on hold first call, then try to pickup<br clear="none">> the new one.<br clear="none">><br clear="none">> The thing is that the SIP re-invite with sendonly attribute can be seen from<br clear="none">> extension 1000 to Asterisk queue, but this SIP invite is not being forwarded<br clear="none">> to GSM gateway. So the GSM gateway keeps waiting for it and because it never<br clear="none">> appears the 1st call is dropped.<br clear="none">><br clear="none">> Maybe you have had this issue in the past. I know that Im not an expert, but<br clear="none">> I have been researching a lot and trying to vary configurations without<br clear="none">> clues.<br clear="none">><br clear="none">> The question is: Is it expected for the Asterisk queue to redirect this<br clear="none">> on-hold message (SIP re-invite with sendonly media attribute) to the GSM<br clear="none">> gateway so it can manage it call waiting feature on the same SIM card?<br clear="none">><br clear="none">> If we repeat the same scenario without queue intervention (i.e. call goes<br clear="none">> directly to the extension) the SIP re-invite floods normally between<br clear="none">> Asterisk and GSM gateway, so GSM gateway can decide what to do with the<br clear="none">> call.<br clear="none">><br clear="none">> I have no specific queue configuration, seems that queues.conf does not have<br clear="none">> any parameter to allow this behaviour of re-sending re-invite/on-hold<br clear="none">> messages.<br clear="none">><br clear="none">> Vendor from GSM gateway side is pointing that “Asterisk js not resending<br clear="none">> on-hold message”.</div><br clear="none">><br clear="none"><br clear="none">Asterisk is a back to back user agent. As such, it does not "forward"<br clear="none">or proxy any SIP messages. The re-INVITE sent from the SIP device<br clear="none">represented by extension 1000 in your scenario is handled by Asterisk,<br clear="none">and causes the channel on the other side of the bridge with that SIP<br clear="none">channel to be put on Hold.<br clear="none"><br clear="none">There is no mechanism in Asterisk today to pass through a re-INVITE to<br clear="none"><div id="yui_3_16_0_1_1418136944274_7923">initiate a remote hold.</div><div id="yui_3_16_0_1_1418136944274_7924"><br></div><br clear="none">-- <br clear="none">Matthew Jordan<br clear="none">Digium, Inc. | Engineering Manager<br clear="none">445 Jan Davis Drive NW - Huntsville, AL 35806 - USA<br clear="none"><div id="yui_3_16_0_1_1418136944274_7926">Check us out at: <a shape="rect" href="http://digium.com/" target="_blank">http://digium.com </a>& <a id="yui_3_16_0_1_1418136944274_7917" shape="rect" href="http://asterisk.org/" target="_blank">http://asterisk.org</a></div><div id="yui_3_16_0_1_1418136944274_7927"><br></div><div id="yui_3_16_0_1_1418136944274_8249" dir="ltr">Thank you master!!<br><a id="yui_3_16_0_1_1418136944274_7917" shape="rect" href="http://asterisk.org/" target="_blank"></a></div><br></div> </div> </div> </div> </div></body></html>