<div>hi sam,</div><div><br></div><div>i found this post</div><a href="http://comments.gmane.org/gmane.comp.telephony.freeswitch.user/8477">http://comments.gmane.org/gmane.comp.telephony.freeswitch.user/8477</a><div><br clear="all">
modify the script to suit your need.</div><div><br></div><div>hope it helps. just dig on w/ FS :-)</div><div>-nandy<br><br><div class="gmail_quote">On Thu, Aug 11, 2011 at 12:28 AM, Sam <span dir="ltr"><<a href="mailto:lakersman2006@yahoo.com">lakersman2006@yahoo.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;"><div><div style="color:#000;background-color:#fff;font-family:times new roman, new york, times, serif;font-size:12pt">
<div><span>Thanks for being so accommodating. I was a bit frustrated in trying to port over an asterisk agi script to freeswitch. I have spent many hours trying to learn how to configure freeswitch, I was about to give up, but I will play with the new changes you made and see if that works for me. <br>
</span></div><div><span><br></span></div><div><span>One other question, when the bridged call hangs up I do not see any value for the hangup time when using getVariable("hangup_time"), so how can I get it?<br></span></div>
<div><br></div><div style="font-family:times new roman, new york, times, serif;font-size:12pt"><div style="font-family:times new roman, new york, times, serif;font-size:12pt"><font face="Arial" size="2"><hr size="1"><b><span style="font-weight:bold">From:</span></b> Anthony Minessale
<<a href="mailto:anthony.minessale@gmail.com" target="_blank">anthony.minessale@gmail.com</a>><br><b><span style="font-weight:bold">To:</span></b> FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>><br>
<b><span style="font-weight:bold">Sent:</span></b> Wednesday, August 10, 2011 8:52 AM<br><b><span style="font-weight:bold">Subject:</span></b> Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide<br></font><div><div>
</div><div class="h5"><br><div>=D <div><br></div><div>ok, sure. If that's your only complaint.... see commit 9d98d49f0556fb79656c8403f285ae0a615439d3</div><div><br></div><div>Some caveats</div><div><br></div><div>1) There is actually less specific, more generalized data in this DIALSTATUS variable than what we already report, when you're ready to move on see the originate_disposition variable: It's kind of like going from reporting the precise geo-location of a cafe in Paris to generalizing it to "EUROPE" </div>
<div><br></div><div>We follow the Q.850 standard for call cause codes and follow the SIP RFC to map sip response codes to/from the Q.850 equivalent. Also each module has its own version "sip_hangup_disposition" for sip so you have both the real sip response code AND the official Q.850 equiv variables set on each call.</div>
<div><br></div><div><br></div><div>2) We don't have a torture feature so we never return that code.</div><div><br></div><div><br></div><div>3) Since our originate can return before a call is answered I added "EARLY" which means the originate succeeded but its still not answered.</div>
<div><br></div><div>4) For any others that do not map directly to FreeSWITCH, I just installed a copy of originate_disposition for good measure.</div><div><br>P.S </div><div><br></div><div>This email took longer to compose than the patch did while sitting in the middle of a crowded room so you probably could have simply parsed the originate originate_disposition yourself but if it helps people get over being stuck in a paradigm, it's worth it for me to write........</div>
<div> </div><div><br><div>On Tue, Aug 9, 2011 at 7:54 PM, Sam <span dir="ltr"><<a rel="nofollow" href="mailto:lakersman2006@yahoo.com" target="_blank">lakersman2006@yahoo.com</a>></span> wrote:<br><blockquote style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div><div style="color:#000;background-color:#fff;font-family:times new roman, new york, times, serif;font-size:12pt"><div><span>I find that Asterisk's AGI is much easier to use, they allow you to retrieve the dial status much easier than freeswitch's api's. Come on freeswitch, if you want to be better than asterisk, you should make it easier to get the dialstatus, etc. At this point asterisk is still defacto.<br>
</span></div><div><br></div><div style="font-family:times new roman, new york, times, serif;font-size:12pt"><div style="font-family:times new roman, new york, times, serif;font-size:12pt"><font face="Arial" size="2"><hr size="1">
<b><span style="font-weight:bold">From:</span></b> Nestor A Diaz <<a rel="nofollow" href="mailto:nestor@tiendalinux.com" target="_blank">nestor@tiendalinux.com</a>><br><b><span style="font-weight:bold">To:</span></b> <a rel="nofollow" href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a><br>
<b><span style="font-weight:bold">Sent:</span></b> Tuesday, August 9, 2011 9:48 AM<br><b><span style="font-weight:bold">Subject:</span></b> [Freeswitch-users] Asterisk to FreeSWITCH migration guide<br></font><div><div></div>
<div><br><div>
<tt><big><big>Hi Guys.</big><br>
<br>
<big>I am starting to use FreeSWITCH, i am an asterisk user since the
1.0.7
release appears as a package on the debian distribution, at the
beginning i was amazed by the fact i can build a PBX for my own
business
and i did, later i began to install this system for my customers and
sooner i meet the problems, however being the software open source i
always find a way to fix things using patchs from others, sometimes i
felt how my life was at risk when the system stops working and that
usually happens when i have to use queues and dealing with digium
hardware.</big><br>
<br>
<big>Fixing those problems either by applying patches or by changing
the
hardware where the digium cards were supposed to be installed helps me,
but that was to much stress for me and seeking for a balance that will
let me invest more time on services, configuration and hoping to have
better hardware
options brings me to freeswitch.</big><br>
<br>
<big>I agree with freeswitch philosophy that instead of having
thousands of
modules
that don't work fine i prefer just a few that works the way it should
be, a rock solid system for a corporate pbx and a call center is what i
want.</big><br>
<br>
<big>So here i am trying to begin the conversion, and i hope the
information
we can transcript in this list will help others that want to try
another
alternative to asterisk.</big><br>
<br>
<big>First of all i think the saner for a migration is to have the two
systems
running either on the same machine or different and use the stable
features of each one.</big><br>
<br>
<big>So could you please freeswitch users help me with this rosetta
stone migration guide in order to post it to <a rel="nofollow" href="http://voip-info.org" target="_blank">voip-info.org</a> or
freeswitch wiki (i list only the ones i currently use ):<br>
<br>
</big></big></tt><br>
<table border="0" cellspacing="0">
<colgroup><col width="199"><col width="199"><col width="213"></colgroup>
<tbody>
<tr>
<td style="border:1px solid rgb(0, 0, 0)" align="LEFT" height="17" width="199"><b><font face="DejaVu Sans Mono">Technology</font></b></td>
<td style="border:1px solid rgb(0, 0, 0)" align="LEFT" width="199"><b><font face="DejaVu Sans Mono">Asterisk</font></b></td>
<td style="border:1px solid rgb(0, 0, 0)" align="LEFT" width="213"><b><font face="DejaVu Sans Mono">Freeswitch</font></b></td>
</tr>
<tr>
<td style="border:1px solid rgb(0, 0, 0)" align="LEFT" height="32"><font face="DejaVu Sans Mono">PSTN Connectivity (Digium /
Sangoma)</font></td>
<td style="border:1px solid rgb(0, 0, 0)" align="LEFT"><font face="DejaVu Sans Mono">dahdi</font></td>
<td style="border:1px solid rgb(0, 0, 0)" align="LEFT"><font face="DejaVu Sans Mono">freetdm</font></td>
</tr>
<tr>
<td style="border:1px solid rgb(0, 0, 0)" align="LEFT" height="106"><font face="DejaVu Sans Mono">IAX2</font></td>
<td style="border:1px solid rgb(0, 0, 0)" align="LEFT"><font face="DejaVu Sans Mono">mod_iax</font></td>
<td style="border:1px solid rgb(0, 0, 0)" align="LEFT"><font face="DejaVu Sans Mono">?? none stable yet.<br>
Use Asterisk to forward traffic via SIP.<br>
Enable Hardware HPET for IAX2 trunk if card not available for Asterisk </font></td>
</tr>
<tr>
<td style="border:1px solid rgb(0, 0, 0)" align="LEFT" height="32"><font face="DejaVu Sans Mono">Bluetooth Channel</font></td>
<td style="border:1px solid rgb(0, 0, 0)" align="LEFT"><font face="DejaVu Sans Mono">chan_mobile</font></td>
<td style="border:1px solid rgb(0, 0, 0)" align="LEFT"><font face="DejaVu Sans Mono">??<br>
Use asterisk via SIP<br>
</font></td>
</tr>
<tr>
<td style="border:1px solid rgb(0, 0, 0)" align="LEFT" height="32"><font face="DejaVu Sans Mono">Skype</font></td>
<td style="border:1px solid rgb(0, 0, 0)" align="LEFT"><font face="DejaVu Sans Mono">Skypeforasterisk (no longer for sale)</font></td>
<td style="border:1px solid rgb(0, 0, 0)" align="LEFT"><font face="DejaVu Sans Mono">mod_skypeopen</font></td>
</tr>
<tr>
<td style="border:1px solid rgb(0, 0, 0)" align="LEFT" height="17"><font face="DejaVu Sans Mono">CDR Stadistics</font></td>
<td style="border:1px solid rgb(0, 0, 0)" align="LEFT"><font face="DejaVu Sans Mono"> Arternic cdr-stats </font></td>
<td style="border:1px solid rgb(0, 0, 0)" align="LEFT"><font face="DejaVu Sans Mono">??</font></td>
</tr>
<tr>
<td style="border:1px solid rgb(0, 0, 0)" align="LEFT" height="32"><font face="DejaVu Sans Mono">Queue Statistics </font></td>
<td style="border:1px solid rgb(0, 0, 0)" align="LEFT"><font face="DejaVu Sans Mono"> Asteriskguru queue-stats</font></td>
<td style="border:1px solid rgb(0, 0, 0)" align="LEFT"><font face="DejaVu Sans Mono">??</font></td>
</tr>
<tr>
<td style="border:1px solid rgb(0, 0, 0)" align="LEFT" height="17"><font face="DejaVu Sans Mono">Web Management</font></td>
<td style="border:1px solid rgb(0, 0, 0)" align="LEFT"><font face="DejaVu Sans Mono">Freepbx</font></td>
<td style="border:1px solid rgb(0, 0, 0)" align="LEFT"><font face="DejaVu Sans Mono">??</font></td>
</tr>
<tr>
<td style="border:1px solid rgb(0, 0, 0)" align="LEFT" height="17"><font face="DejaVu Sans Mono">IVR</font></td>
<td style="border:1px solid rgb(0, 0, 0)" align="LEFT"><font face="DejaVu Sans Mono">AGI / AMI</font></td>
<td style="border:1px solid rgb(0, 0, 0)" align="LEFT"><font face="DejaVu Sans Mono">Event Socket</font></td>
</tr>
<tr>
<td style="border:1px solid rgb(0, 0, 0)" align="LEFT" height="62"><font face="DejaVu Sans Mono">Codec G.729</font></td>
<td style="border:1px solid rgb(0, 0, 0)" align="LEFT"><font face="DejaVu Sans Mono">Transcodind Cards<br>
G.729 licenses<br>
Free G.729 (Intel IPP)</font></td>
<td style="border:1px solid rgb(0, 0, 0)" align="LEFT"><font face="DejaVu Sans Mono">Transcodind Cards<br>
G.729 licenses<br>
fsg729 Intel IPP(any experience with it ? )</font></td>
</tr>
<tr>
<td style="border:1px solid rgb(0, 0, 0)" align="LEFT" height="32"><font face="DejaVu Sans Mono">Fax Handling</font></td>
<td style="border:1px solid rgb(0, 0, 0)" align="LEFT"><font face="DejaVu Sans Mono"> Iaxmodem with Hylafax</font></td>
<td style="border:1px solid rgb(0, 0, 0)" align="LEFT"><font face="DejaVu Sans Mono">??<br>
Iaxmodem via asterisk to FS via SIP ?<br>
</font></td>
</tr>
<tr>
<td style="border:1px solid rgb(0, 0, 0)" align="LEFT" height="17"><font face="DejaVu Sans Mono">SIP</font></td>
<td style="border:1px solid rgb(0, 0, 0)" align="LEFT"><font face="DejaVu Sans Mono">chan_sip</font></td>
<td style="border:1px solid rgb(0, 0, 0)" align="LEFT"><font face="DejaVu Sans Mono">sofia</font></td>
</tr>
<tr>
<td style="border:1px solid rgb(0, 0, 0)" align="LEFT" height="17"><font face="DejaVu Sans Mono">ACD</font></td>
<td style="border:1px solid rgb(0, 0, 0)" align="LEFT"><font face="DejaVu Sans Mono">app_queue</font></td>
<td style="border:1px solid rgb(0, 0, 0)" align="LEFT"><font face="DejaVu Sans Mono">mod_callcenter</font></td>
</tr>
</tbody>
</table>
<br>
<big><big><tt><br>
<big>Thank you all</big><small><br>
</small><br>
<br>
-- <br>
Nestor A. Diaz<br>
Ingeniero de Sistemas<br>
Tel. <a rel="nofollow">+57 1-485-3020 x 211</a><br>
Cel. <a rel="nofollow">+57 316-227-3593</a><br>
Tel. SIP: <a rel="nofollow">sip:211@tiendalinux.com</a><br>
Email/MSN: <a rel="nofollow" href="mailto:nestor@tiendalinux.com" target="_blank">nestor@tiendalinux.com</a><br>
<a rel="nofollow" href="http://www.tiendalinux.com/" target="_blank">http://www.tiendalinux.com/</a><br>
Bogota, Colombia <br>
</tt></big></big>
<br>
</div><br></div></div><div>_______________________________________________<br>Join us at ClueCon 2011, Aug 9-11, Chicago<br><a rel="nofollow" href="http://www.cluecon.com" target="_blank">http://www.cluecon.com</a> 877-7-4ACLUE<br>
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<br></blockquote></div><br><br clear="all"><br>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a rel="nofollow" href="http://www.freeswitch.org/" target="_blank">http://www.freeswitch.org/</a><br>ClueCon <a rel="nofollow" href="http://www.cluecon.com/" target="_blank">http://www.cluecon.com/</a><br>
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