Sam,<div><br></div><div>Did you already pastebin a copy of your script and dialplan? I know we had talked about it. In any case, I'm hoping to see what you're doing so that we can offer you some alternative ideas. </div>
<div><br></div><div>-MC<br><br><div class="gmail_quote">On Wed, Aug 17, 2011 at 11:53 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>I don't see much difference in terms of "originate_disposition" when calling answer explicitly opposed to not calling it, so since it appears there is more issues not calling it I guess for now I should just call 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"><div class="im">
<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>
</div><b><span style="font-weight:bold">Sent:</span></b> Wednesday, August 17, 2011 11:00 AM<div><div></div><div class="h5"><br><b><span style="font-weight:bold">Subject:</span></b> Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide<br>
</div></div></font><div><div></div><div class="h5"><br><div>This is another problem related to the callflow of the provider that can be fixed.<div><br></div><div>In an ideal world, using the defaults, when the early media comes up on the b leg it will pass to the a leg which also will start sending early media and it will happily pass through.</div>
<div><br></div><div>My hunch is they have calls to you set on some kine of LCR hunt that is misconfigured and it's trying to get the answer to stop hunting which is not right.</div><div><br><br><div>
On Wed, Aug 17, 2011 at 12:32 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 have also found a side effect when I do not explicitly call answer on the inbound leg for b-leg calls that do not return "answer" when using another DID provider (VOIPInnovations). The side effect is that the a-leg can hear the telco network messages from the carrier like "I'm sorry the number you dialed is not a working number ..." or "The user is not accepting calls at the moment." <br>
</span></div><div><br><span></span></div><div><span>If I do explicitly call answer, then I cannot hear those telco messages, which would seem to be better fitting for my case.<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"><div><hr size="1"><b><span style="font-weight:bold">From:</span></b> Michael Collins <<a rel="nofollow" href="mailto:msc@freeswitch.org" target="_blank">msc@freeswitch.org</a>><br>
</div><div><b><span style="font-weight:bold">To:</span></b> FreeSWITCH Users Help <<a rel="nofollow" href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>><br>
</div>
<b><span style="font-weight:bold">Sent:</span></b> Wednesday, August 17, 2011 9:16 AM<div><div></div><div><br><b><span style="font-weight:bold">Subject:</span></b> Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide<br>
</div></div></font><div><div></div><div><br><div><br><br><div>On Tue, Aug 16, 2011 at 10:14 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>The DID provider I am using is from iCall, and I was searching through their website and noticed that they mentioned a quote with your name on it <a rel="nofollow" href="http://carriers.icall.com/open-source/" target="_blank">http://carriers.icall.com/open-source/</a></span></div>
<div><span>so it appears you have had experience with them.<br></span></div><div><br></div></div></div></blockquote><div>We have a lot of experience with iCall. I'm not familiar with any hard requirement to "answer" the inbound leg prior to bridging an outbound leg. What happens in your dialplan if you don't explicitly answer the inbound leg prior to calling the bridge app?</div>
<div>-MC</div><div> </div><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></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"><div><hr size="1">
<b><span style="font-weight:bold">From:</span></b> Anthony Minessale <<a rel="nofollow" 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 rel="nofollow" href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>><br>
</div><div><b><span style="font-weight:bold">Sent:</span></b> Tuesday, August 16, 2011 5:29 PM<br></div><div><div></div><div><b><span style="font-weight:bold">Subject:</span></b> Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide<br>
</div></div></font><div><div></div><div><br><div>You should never answer a call before bridging it anyway, it breaks all of the accounting.<div>It would make sense to find out why the provider is doing that and get it fixed.</div>
<div><br><br><div>On Mon, Aug 15, 2011 at 5:17 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>Anthony,<br>
<br>
My gripe was not about simply having a DIALSTATUS variable in Freeswitch which copies what is from "originate_disposition" what I wanted is to be able to get the status of the B-Leg because right now when early media is played (which i wanted) "originate_disposition" shows "ANSWER" which I think is caused by me explitly called the "answer" app in my dialplan before the bridge app, this is because my DID provider requires an answer/sip 200 or else it will keep re-sending the sip invite, therefore causing freeswitch to keep creating new channels. All I want is to be able to get the proper sip/hangup/dial statuses of the B-leg.<br>
</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"><div><hr size="1">
<b><span style="font-weight:bold">From:</span></b> Anthony Minessale <<a rel="nofollow" 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 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> Wednesday, August 10, 2011 8:52 AM<br></div><b><span style="font-weight:bold">Subject:</span></b> Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide<br></font><div>
<div></div><div><br><div>=D <div><br></div><div>ok, sure. If that's your only complaint.... see commit 9d98d49f0556fb79656c8403f285ae0a615439d3</div><br><div><br><br>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>
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<br></blockquote></div><br><br clear="all"><div><br></div>-- <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>
Twitter: <a rel="nofollow" href="http://twitter.com/FreeSWITCH_wire" target="_blank">http://twitter.com/FreeSWITCH_wire</a><br><br>AIM: anthm<br><a rel="nofollow" href="mailto:MSN%3Aanthony_minessale@hotmail.com" target="_blank">MSN:anthony_minessale@hotmail.com</a><br>
GTALK/JABBER/<a rel="nofollow" href="mailto:PAYPAL%3Aanthony.minessale@gmail.com" target="_blank">PAYPAL:anthony.minessale@gmail.com</a><br>
IRC: <a rel="nofollow" href="http://irc.freenode.net" target="_blank">irc.freenode.net</a> #freeswitch<br><br>FreeSWITCH Developer Conference<br><a rel="nofollow" href="mailto:sip%3A888@conference.freeswitch.org" target="_blank">sip:888@conference.freeswitch.org</a><br>
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<br></blockquote></div><br><br clear="all"><div><br></div>-- <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>
Twitter: <a rel="nofollow" href="http://twitter.com/FreeSWITCH_wire" target="_blank">http://twitter.com/FreeSWITCH_wire</a><br><br>AIM: anthm<br><a rel="nofollow" href="mailto:MSN%3Aanthony_minessale@hotmail.com" target="_blank">MSN:anthony_minessale@hotmail.com</a><br>
GTALK/JABBER/<a rel="nofollow" href="mailto:PAYPAL%3Aanthony.minessale@gmail.com" target="_blank">PAYPAL:anthony.minessale@gmail.com</a><br>
IRC: <a rel="nofollow" href="http://irc.freenode.net" target="_blank">irc.freenode.net</a> #freeswitch<br><br>FreeSWITCH Developer Conference<br><a rel="nofollow" href="mailto:sip%3A888@conference.freeswitch.org" target="_blank">sip:888@conference.freeswitch.org</a><br>
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pstn:+19193869900<br>
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