NOTIFY messages are a SIP feature outside of an INVITE call flow. Ending the call using 200+BYE, 4xx, 5xx or 6xx are your only options I'm afraid.<div><br></div><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">
I'm writing a CTI application so terminating the call via the api prior to answer without errors is really important.</blockquote><div><br></div>Any call that's unanswered is a failed call, but that's not necessarily an error (eg user busy).<div>
<br></div><div>I understand your concern that the caller might not know if their call succeeded or not... Is there any possibility perhaps to answer the call, send it to a playback (eg 'your request was successful') and then hangup?</div>
<div><div><br></div><div>-Steve</div><div><br></div><div><br></div><div><br><br>
<div class="gmail_quote">On 11 February 2013 18:37, Gary Foreman <span dir="ltr"><<a href="mailto:gaz.foreman@gmail.com" target="_blank">gaz.foreman@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr">I was concerned that might be the case, I have managed to get different errors by send alternative hangup codes "user_busy" for example is recognised but as you say all are still errors.<div><br>
</div>
<div>I'm planning on using Polycom sip phones, does anyone know if it is possible to end an unanswered call using a notify message or something? Like the talk sip notify message used to answer.<br></div><div><br></div>
<div>I'm writing a CTI application so terminating the call via the api prior to answer without errors is really important. <br></div><div><br></div><div>Thanks again<br><div class="gmail_extra"><br><br><div class="gmail_quote">
<div>
On Mon, Feb 11, 2013 at 6:07 PM, <span dir="ltr"><<a href="mailto:freeswitch-users-request@lists.freeswitch.org" target="_blank">freeswitch-users-request@lists.freeswitch.org</a>></span> wrote:<br></div><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>Today's Topics:<br>
<br></div>
1. Re: Throttle Error (Avi Marcus)<br>
2. Re: Terminating calls prior to answer using api (Steven Ayre)<br>
3. Re: Terminating calls prior to answer using api (Steven Ayre)<br>
4. High Availability Cluster Module for FreeSWITCH - Major News<br>
Update (Eliot Gable)<br>
5. Re: High Availability Cluster Module for FreeSWITCH<br>
(Luis Daniel Lucio Quiroz)<div><br>
<br><br>---------- Forwarded message ----------<br>From: Avi Marcus <<a href="mailto:avi@avimarcus.net" target="_blank">avi@avimarcus.net</a>><br>To: FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>><br>
Cc: <br></div>Date: Mon, 11 Feb 2013 19:09:33 +0200<br>Subject: Re: [Freeswitch-users] Throttle Error<br><p>As Ken's original message said, try "fsctl sps 40" or the like. <br>
He said that otherwise only a restart, not reload xml would work. </p>
<p>-Avi<br>
(This message was painstakingly thumbed out on my mobile, so apologies for brevity and errors.)</p>
<p></p><blockquote type="cite">On Feb 11, 2013 6:57 PM, "Archana Venugopan" <<a href="mailto:a.venugopan@mundio.com" target="_blank">a.venugopan@mundio.com</a>> wrote:<br><br>
<div lang="EN-GB" link="blue" vlink="purple">
<div>
<p><span style="font-size:11pt;font-family:Calibri,sans-serif;color:rgb(31,73,125)">Hi,<u></u><u></u></span></p>
<p><span style="font-size:11pt;font-family:Calibri,sans-serif;color:rgb(31,73,125)"><u></u> <u></u></span></p>
<p><span style="font-size:11pt;font-family:Calibri,sans-serif;color:rgb(31,73,125)">When I gave fsctl sps it shows still 10. Can you please tell me apart from reload_xml do we have to anyother thing after changing switch.conf.xml?<u></u><u></u></span></p>
<p><span style="font-size:11pt;font-family:Calibri,sans-serif;color:rgb(31,73,125)"><u></u> <u></u></span></p>
<div>
<p><span style="font-size:11pt;font-family:Calibri,sans-serif;color:rgb(31,73,125)">Regards,<u></u><u></u></span></p>
<p><span style="font-size:11pt;font-family:Calibri,sans-serif;color:rgb(31,73,125)">Archana<u></u><u></u></span></p>
</div>
<p><span style="font-size:11pt;font-family:Calibri,sans-serif;color:rgb(31,73,125)"><u></u> <u></u></span></p>
<div>
<div style="border-style:solid none none;border-top-color:rgb(181,196,223);border-top-width:1pt;padding:3pt 0cm 0cm">
<p><b><span lang="EN-US" style="font-size:10pt;font-family:Tahoma,sans-serif">From:</span></b><span lang="EN-US" style="font-size:10pt;font-family:Tahoma,sans-serif"> <a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a> [mailto:<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a>]
<b>On Behalf Of </b>Archana Venugopan<br>
<b>Sent:</b> 11 February 2013 16:37</span></p><p><font color="#500050"><br>To: FreeSWITCH Users Help<br>Subject: Re: [Freeswitch-users] Throttle Error</font></p><p></p>
</div>
</div><p><font color="#500050"><br><br> <br><br>Ya i did re-loaded the xml. And this is what I see in status<br><br> <br><br>UP 0 years, 6 days, 9 hours, 3...</font></p></div>
</div><div>
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<br></div></blockquote><div><p></p>
<br><br>---------- Forwarded message ----------<br>From: Steven Ayre <<a href="mailto:steveayre@gmail.com" target="_blank">steveayre@gmail.com</a>><br>To: FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>><br>
Cc: <br></div><div>Date: Mon, 11 Feb 2013 17:40:57 +0000<br>Subject: Re: [Freeswitch-users] Terminating calls prior to answer using api<br></div><div><div><div>You either answer a call with 200, redirect with 3xx, or give an error 4xx 5xx or 6xx... basically without answering your only option is to return an error.</div>
<div><br></div><div>The specific error code can show a varying number of levels of 'error' though. Some might be a perfectly valid result (eg 486 User Busy) while other's imply a more serious error (eg 500 Server Internal Error). But there's not really one for a successful unanswered call.</div>
<div><br></div><div>How your phone behaves on the various errors is really up to the SIP client though. A generic unable to connect message doesn't sound very useful, I'd expect different error tones depending on the response code.</div>
<div><br></div><div>-Steve</div><div><br></div><div><br><br><div class="gmail_quote">On 11 February 2013 15:30, Gary Foreman <span dir="ltr"><<a href="mailto:gaz.foreman@gmail.com" target="_blank">gaz.foreman@gmail.com</a>></span> wrote:<br>
<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"><div dir="ltr">Hi,<div><br></div><div>I'm having an issue gracefully terminating calls prior to them being answered.</div>
<div><br></div><div>When I issue the uuid_kill command prior to the call being answered my sip client gives an error "unable to connect" with an error tone.</div>
<div><br></div><div>Is it possible to cancel the client dial without causing an error? Maybe using a sip notify message or something?</div><div><br></div><div>I can't seem to find a list of valid sip notify messages anywhere.</div>
<div><br></div><div>Thanks</div></div>
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<br></blockquote></div><br></div>
<br><br></div></div><div>---------- Forwarded message ----------<br>From: Steven Ayre <<a href="mailto:steveayre@gmail.com" target="_blank">steveayre@gmail.com</a>><br>To: FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>><br>
Cc: <br></div><div>Date: Mon, 11 Feb 2013 17:42:51 +0000<br>Subject: Re: [Freeswitch-users] Terminating calls prior to answer using api<br></div><div><div><div>The NORMAL_CLEARING ISDN clearing cause (16) has no mapping to any SIP code for example, it's a code you normally would only see a) when the caller hangs up at any time (regardless of answered or not) or b) when call has been answered <i>then</i> callee hangs up.</div>
<div><br></div><div>-Steve</div><div><br></div><div><br></div><br><br><div class="gmail_quote">On 11 February 2013 17:40, Steven Ayre <span dir="ltr"><<a href="mailto:steveayre@gmail.com" target="_blank">steveayre@gmail.com</a>></span> wrote:<br>
<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"><div>You either answer a call with 200, redirect with 3xx, or give an error 4xx 5xx or 6xx... basically without answering your only option is to return an error.</div>
<div><br></div><div>The specific error code can show a varying number of levels of 'error' though. Some might be a perfectly valid result (eg 486 User Busy) while other's imply a more serious error (eg 500 Server Internal Error). But there's not really one for a successful unanswered call.</div>
<div><br></div><div>How your phone behaves on the various errors is really up to the SIP client though. A generic unable to connect message doesn't sound very useful, I'd expect different error tones depending on the response code.</div>
<div><br></div><div>-Steve</div><div><br></div><div><br><br><div class="gmail_quote"><div>On 11 February 2013 15:30, Gary Foreman <span dir="ltr"><<a href="mailto:gaz.foreman@gmail.com" target="_blank">gaz.foreman@gmail.com</a>></span> wrote:<br>
</div><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"><div><div><div dir="ltr">Hi,<div><br></div><div>I'm having an issue gracefully terminating calls prior to them being answered.</div>
<div><br></div><div>When I issue the uuid_kill command prior to the call being answered my sip client gives an error "unable to connect" with an error tone.</div>
<div><br></div><div>Is it possible to cancel the client dial without causing an error? Maybe using a sip notify message or something?</div><div><br></div><div>I can't seem to find a list of valid sip notify messages anywhere.</div>
<div><br></div><div>Thanks</div></div>
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</blockquote></div><br>
<br><br></div></div><div>---------- Forwarded message ----------<br>From: Eliot Gable <<a href="mailto:egable%2Bfreeswitch@gmail.com" target="_blank">egable+freeswitch@gmail.com</a>><br>To: FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>><br>
Cc: <br></div>Date: Mon, 11 Feb 2013 12:44:55 -0500<br>Subject: [Freeswitch-users] High Availability Cluster Module for FreeSWITCH - Major News Update<br>After discussing this project in further detail with Tony, I have decided to cancel the Kickstarter project. Funding will now go through FreeSWITCH Solutions. If you would like to contribute towards making mod_ha_cluster a reality, please send contributions to FreeSWITCH Solutions.<div>
<br></div><div>Thanks!<br clear="all"><div><br></div>-- <br>Eliot Gable<br><br></div><div>
<br><br>---------- Forwarded message ----------<br>From: Luis Daniel Lucio Quiroz <<a href="mailto:luis.daniel.lucio@gmail.com" target="_blank">luis.daniel.lucio@gmail.com</a>><br>To: FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>><br>
Cc: <br></div>Date: Mon, 11 Feb 2013 13:06:16 -0500<br>Subject: Re: [Freeswitch-users] High Availability Cluster Module for FreeSWITCH<br>Is the latest snapshot ofyour module at<br>
git://<a href="http://git.freeswitch.org/freeswitch-contrib.git" target="_blank">git.freeswitch.org/freeswitch-contrib.git</a> ?<br>
<br>
2013/2/11 Eliot Gable <<a href="mailto:egable%2Bfreeswitch@gmail.com" target="_blank">egable+freeswitch@gmail.com</a>>:<br>
> On Mon, Feb 11, 2013 at 7:36 AM, Marcin Gozdalik <<a href="mailto:gozdal@gmail.com" target="_blank">gozdal@gmail.com</a>> wrote:<br>
>><br>
>> +1<br>
>><br>
>> I do not doubt mod_ha is necessary inside of FS and it may be<br>
>> better/simpler than writing Pacemaker resource agent, but writing<br>
>> yet-another-cluster-communication-engine is IMHO the wrong way to go<br>
>> and using Corosync for communication will give a lot of value from<br>
>> mature codebase.<br>
>><br>
><br>
> I understand what you are saying, but what I am trying to get across is that<br>
> I am not writing yet-another-cluster-communication-engine. All I am really<br>
> doing is combining a multicast messaging API written by Tony and the event<br>
> API in FS to broadcast existing state information between multiple FS nodes,<br>
> as well as adding a tiny amount of logic on top of that to coordinate call<br>
> fail over and recovery. That's probably a little over-simplified, but it<br>
> gets the point across. The network communication code is already in FS and<br>
> well tested. The event system is already in FS and well tested. I have<br>
> already written the code to the point that it parses the configuration files<br>
> and starts sending heartbeats out all of the interfaces configured. I have<br>
> also already written a lot of the code that deals with the state<br>
> transitions. All I am talking about doing is implementing a tiny little<br>
> finite state machine. It's a pretty trivial programming task. In fact, I<br>
> think it was covered in my first year at Carnegie Mellon University. Of<br>
> course, I had already figured out how to write such things in high school, I<br>
> just did not know what it was called at that point. My point is, that this<br>
> is not yet-another-cluster-communication-engine. It is a very specific and<br>
> small finite state machine designed solely with the goal in mind of making<br>
> FS have just enough information to coordinate call fail over internally. If<br>
> I recall correctly, a lot of people also said writing<br>
> yet-another-VoIP-server was a waste of time, but now we have FreeSWITCH, and<br>
> it was obviously worth the effort. And I am not even trying to do something<br>
> as complex as that. If you think this is<br>
> yet-another-cluster-communication-engine, you are missing the point. It is<br>
> not. It never will be.<br>
><br>
> Look at Sonus, Genband, Broadsoft, Veraz, etc. All the big-name<br>
> carrier-grade telecom providers have a built-in solution for automatic call<br>
> fail over. The only way FreeSWITCH will ever compete with such solutions is<br>
> if it also has that feature. Pacemaker and Corosync are overkill just to get<br>
> FS to handle single node failures and provide call recovery. It took me a<br>
> full 3 months of working with them every day to really understand how to<br>
> deploy them properly in conjunction with FreeSWITCH and Postgres to provide<br>
> a carrier-grade hot-standby solution which was robust enough to handle 99%<br>
> of the failures I could throw at it. Granted, this was back when the<br>
> configuration still needed to be written by hand in XML and prior the<br>
> existence of any resource agent for FreeSWITCH. But, even with those<br>
> changes, deploying Pacemaker and Corosync is not a simple task. If that is<br>
> the requirement for FS to have HA, it will never truly stand a chance<br>
> against commercial offerings.<div><br>
><br>
><br>
><br>
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