Hi,<div><br><div>yes, please see MusicOnHold() Application. You can call this app in your dialplan. This however will use the default music class and the corresponding music files placed in the asterisk server. If you don't want to stream music from Asterisk server side, try creating a new MusiconHold Class without any proper directory. That way Asterisk would only complain that there is no file to be streamed.</div>
<div><br></div><div>Regards,</div><div>Sammy<br><br><div class="gmail_quote">On Sat, Jan 14, 2012 at 6:25 AM, Johannes Zweng <span dir="ltr"><<a href="mailto:john999888@zweng.at">john999888@zweng.at</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><span>Hi!</span><div><br></div><div>Maybe I am missing something or am a little blind at the moment, but I didn't find out how asterisk can place a call on hold when acting as user agent client to another SIP server.</div>
<div><br></div><div>Scenario:</div><div>----------</div><div>Asterisk registers to another SIP server (provider) as user agent.</div><div>An inbound call from this other SIP server comes in and arrives at asterisk.</div>
<div>Asterisk performs some actions in the dialplan and should place the call on hold after some time, so that the caller only hears the on hold music from my provider (not streamed by my Asterisk).</div><div>
<br></div><div>Technically speaking I want asterisk to send a re-INVITE message containing an updated SDP body with the attribute "a=sendonly" or "a=inactive" added so that the SIP server of my provider (where Asterisk is registered to as user) will recognize that the call should be placed on hold.</div>
<div><br></div><div><br></div><div>A good example of what I want to achieve is presented in Section 2.1 of RFC 5359 (Session Initiation Protocol Service Examples) (<a href="http://tools.ietf.org/html/rfc5359#section-2.1" style="color:rgb(17,85,204)" target="_blank">http://tools.ietf.org/html/rfc5359#section-2.1</a>) where "Bob" would be my Asterisk (as UAC), "Alice" is the external caller and "Proxy" is the provider's SIP server.</div>
<div><br></div><div><br></div><div>Question:</div><div>----------</div><div>Is there any way to perform this from the dialplan or by means of the manager API? Is there an application like "Hold"?</div>
<div><br></div><div><br></div><div>Kind regards and greetings from Austria,</div><div>John :-)</div>
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