<font face="arial" size="2">Have you tryed phoning a fixed line instead of a cell phone?<br />is this giving the same result?<br /><br />I assume your outgoing call to a the cellphone goes through a Zap channel. Try another one (e.g. Zap channel 2), and let us know the result.<br /><br /><br />Alyed </font>
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Return-Path: <asterisk-users-bounces@lists.digium.com> Sat Apr 01 18:47:36 2006<br />Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by mail11.webcontrolcenter.com with SMTP;<br />Sat, 1 Apr 2006 18:47:36 -0700<br /></font>
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<br />I should've mentioned that before. I've tried doing that and it has no<br />effect. I've tried both upper and lower-case 'r's.<br /><br />I've also tried a workaround that I thought would work, but it doesn't:<br />Answering the call and then using the playtones(ringing) command before<br />connecting to my cellphone. <br /><br />-----Original Message-----<br /><br />Date: Sat, 1 Apr 2006 19:59:46 +0100<br />From: "Julian J. M." <julianjm@gmail.com><br />Subject: Re: [Asterisk-Users] Problem: ringtones stop unexpectedly<br />when multiple channels are dialed<br />To: "Asterisk Users Mailing List - Non-Commercial Discussion"<br /><asterisk-users@lists.digium.com><br />Message-ID:<br /><a60dba290604011059k242fd5c6lf9398b4228c624e3@mail.gmail.com><br />Content-Type: text/plain; charset=ISO-8859-1<br /><br />Try adding 'r' to the dial options. According to "show application dial":<br /><br />r - Indicate ringing to the calling party. Pass no audio to the<br />calling<br />party until the called channel has answered.<br /><br /><br />exten => 3058472194,1,Dial(SIP/1035&SIP/17864883123@richmedium,50, r)<br /><br />Julian.<br /><br />On 4/1/06, Carlos A. Alfaro <carlos@brightspeak.com>wrote:<br />><br />><br />><br />> Hello Everyone. I usually find my own solutions for problems but this<br />time,<br />> after several months, I've given up.<br />><br />><br />><br />> My asterisk is set up so that incoming calls from my voip provider ring on<br />> both my sip extension and my cellphone at the same time. When the system<br />> receives an incoming call, ringtones indicating that the call is being<br />> connected play normally for the first 5 seconds to the caller, but they<br />> suddenly stop as the call to my cellphone starts to make progress. This<br />> causes some people to hang up, despite the fact that the call is still<br />being<br />> connected. Callers who stay on the line are able to talk to me on either<br />> the sip extension or the cellphone once I pick up either one.<br />><br />><br />><br />> I have tried a lot of workarounds like including a priority to answer the<br />> incoming call, invoke the playtones command before the dial command, but<br />> this doesn't seem to work either. Can anyone replicate the problem? Have<br />I<br />> ran into a bug? I have pasted as much info as I deemed relevant; please<br />let<br />> me know if I'm missing something. Thanks.<br /><br /><br />------------------------------<br /><br />_______________________________________________<br />--Bandwidth and Colocation provided by Easynews.com --<br /><br />Asterisk-Users mailing list<br />To UNSUBSCRIBE or update options visit:<br />http://lists.digium.com/mailman/listinfo/asterisk-users<br /><br /><br />End of Asterisk-Users Digest, Vol 21, Issue 2<br />*********************************************<br /><br />_______________________________________________<br />--Bandwidth and Colocation provided by Easynews.com --<br /><br />Asterisk-Users mailing list<br />To UNSUBSCRIBE or update options visit:<br />http://lists.digium.com/mailman/listinfo/asterisk-users<br /><br /></carlos@brightspeak.com></a60dba290604011059k242fd5c6lf9398b4228c624e3@mail.gmail.com></asterisk-users@lists.digium.com></julianjm@gmail.com>