Thanks MC. You got it right.<div><br></div><div>Appreciate your reply.</div><div><br></div><div>Cheers ! </div><div>Dip Mehta<br><br><div class="gmail_quote">On Sat, Oct 29, 2011 at 12:07 AM, <span dir="ltr"><<a href="mailto:freeswitch-users-request@lists.freeswitch.org">freeswitch-users-request@lists.freeswitch.org</a>></span> wrote:<br>
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<br>Today's Topics:<br>
<br>
1. Re: Hangup Causes (Michael Collins)<br>
2. Re: Dialing issues (Michael Collins)<br>
3. Re: High load on database server (<a href="mailto:john@mobex.biz">john@mobex.biz</a>)<br>
<br><br>---------- Forwarded message ----------<br>From: Michael Collins <<a href="mailto:msc@freeswitch.org">msc@freeswitch.org</a>><br>To: FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a>><br>
Date: Fri, 28 Oct 2011 11:10:21 -0700<br>Subject: Re: [Freeswitch-users] Hangup Causes<br>Although not easy on a production system, your best bet is to catch the behavior in the act. Usually this means that you have to rely upon humans to report the symptoms to you in a timely and accurate manner. <div>
<br></div>
<div>One way you can help track these down is to use something like pcapsipdump and do a ring buffer that covers at least several days. Then, when you hear about a symptom you can go back and look at the corresponding pcap file which will have both signaling and audio. From there you can hopefully trace the root of the problem.</div>
<div><br></div><div>-MC</div><div><br></div><div><a href="http://wiki.freeswitch.org/wiki/Packet_Capture#pcapsipdump" target="_blank">http://wiki.freeswitch.org/wiki/Packet_Capture#pcapsipdump</a><br><div><br><div class="gmail_quote">
On Fri, Oct 28, 2011 at 9:19 AM, Jason Moran <span dir="ltr"><<a href="mailto:jmoran@secureachsystems.com" target="_blank">jmoran@secureachsystems.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div lang="EN-US" link="blue" vlink="purple">
<div>
<p class="MsoNormal">I have some questions about the types of statuses we are
getting (Yes, I’m already familiar with <a href="http://wiki.freeswitch.org/wiki/Hangup_causes" target="_blank">http://wiki.freeswitch.org/wiki/Hangup_causes</a>
)<u></u><u></u></p>
<p class="MsoNormal"><u></u> <u></u></p>
<p class="MsoNormal">We just deployed FS to production a couple of days ago and
here is a sample of our Hangup Causes:<u></u><u></u></p>
<p class="MsoNormal">CALL_REJECTED 1<u></u><u></u></p>
<p class="MsoNormal">NO_ANSWER 738<u></u><u></u></p>
<p class="MsoNormal">NO_USER_RESPONSE 175<u></u><u></u></p>
<p class="MsoNormal">NORMAL_CLEARING 1667 (Expected
that this is only Early Hangups)<u></u><u></u></p>
<p class="MsoNormal">NORMAL_TEMPORARY_FAILURE 4<u></u><u></u></p>
<p class="MsoNormal">NORMAL_TEMPORARY_FAILURE (Rang) 37
(Rang means this status happened more than 5 seconds after the session was
started)<u></u><u></u></p>
<p class="MsoNormal">Complete 224
(Complete is our own status instead of NORMAL_CLEARING, meaning WE hung up
because we finished the entire message).<u></u><u></u></p>
<p class="MsoNormal">ORIGINATOR_CANCEL (Rang) 3<u></u><u></u></p>
<p class="MsoNormal">RECOVERY_ON_TIMER_EXPIRE (Rang) 9 <u></u><u></u></p>
<p class="MsoNormal">UNALLOCATED_NUMBER 2<u></u><u></u></p>
<p class="MsoNormal">USER_BUSY 35<u></u><u></u></p>
<p class="MsoNormal"><u></u> <u></u></p>
<p class="MsoNormal">My questions:<u></u><u></u></p>
<p><u></u><span>1.<span style="font:7.0pt "Times New Roman"">
</span></span><u></u>NORMAL_TEMPORARY_FAILURE happens a lot, and often
returns 30 or more seconds after starting. Could this be a “silent call”
(etc)? We have gotten 3-4 complaints of silent calls and I’m hoping to
find the cause.<u></u><u></u></p>
<p><u></u><span>2.<span style="font:7.0pt "Times New Roman"">
</span></span><u></u>Is there anything here (or ideas elsewhere?) to explain
why a call will be silent once in a while?<u></u><u></u></p>
<p><u></u><span>3.<span style="font:7.0pt "Times New Roman"">
</span></span><u></u>Is this spread of causes normal or does it seem strange<u></u><u></u></p>
<p><u></u><span>4.<span style="font:7.0pt "Times New Roman"">
</span></span><u></u>We’ve had a few people note that the message on
their voicemail is over before they can hear it - even though we play the
message twice (to make short messages longer). Does anybody know of a solution
for this?<u></u><u></u></p>
<p class="MsoNormal"><u></u> <u></u></p>
<p class="MsoNormal">Thanks! -Jason<u></u><u></u></p>
<p class="MsoNormal"><u></u> <u></u></p>
</div>
</div>
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<br></blockquote></div><br></div></div>
<br><br>---------- Forwarded message ----------<br>From: Michael Collins <<a href="mailto:msc@freeswitch.org">msc@freeswitch.org</a>><br>To: FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a>><br>
Date: Fri, 28 Oct 2011 11:15:13 -0700<br>Subject: Re: [Freeswitch-users] Dialing issues<br>This is a newfies configuration issue. I've only looked briefly at the newfies dialer (I don't want to get anywhere near it because it's AGPL - yuck) and I do know that it requires you to specify your gateway in some manner. It does not look like you have done that. I'd review this information from the <a href="http://newfies-dialer.org" target="_blank">newfies-dialer.org</a> documentation site:<div>
<br></div><div><h3 style="margin-top:0px;margin-right:0px;margin-bottom:0.75em;margin-left:0px;padding-top:0px;padding-right:0px;padding-bottom:0px;padding-left:0px;border-top-width:0px;border-right-width:0px;border-bottom-width:0px;border-left-width:0px;border-style:initial;border-color:initial;outline-width:0px;outline-style:initial;outline-color:initial;font-size:2.4em;vertical-align:baseline;background-color:rgb(236, 236, 236);line-height:1em;font-weight:lighter;color:rgb(102, 102, 102);font-family:Arial, Helvetica, 'Liberation Sans', FreeSans, sans-serif">
Newfies-Dialer’s Gateway</h3><p style="margin-top:0px;margin-right:0px;margin-bottom:1.5em;margin-left:0px;padding-top:0px;padding-right:0px;padding-bottom:0px;padding-left:0px;border-top-width:0px;border-right-width:0px;border-bottom-width:0px;border-left-width:0px;border-style:initial;border-color:initial;outline-width:0px;outline-style:initial;outline-color:initial;font-size:1.2em;vertical-align:baseline;background-color:rgb(236, 236, 236);color:rgb(102, 102, 102);font-family:Arial, Helvetica, 'Liberation Sans', FreeSans, sans-serif;line-height:18px">
Log into the Newfies-Dialer interface using your root username and password previously created in these instructions.</p><p style="margin-top:0px;margin-right:0px;margin-bottom:1.5em;margin-left:0px;padding-top:0px;padding-right:0px;padding-bottom:0px;padding-left:0px;border-top-width:0px;border-right-width:0px;border-bottom-width:0px;border-left-width:0px;border-style:initial;border-color:initial;outline-width:0px;outline-style:initial;outline-color:initial;font-size:1.2em;vertical-align:baseline;background-color:rgb(236, 236, 236);color:rgb(102, 102, 102);font-family:Arial, Helvetica, 'Liberation Sans', FreeSans, sans-serif;line-height:18px">
Click Admin, and locate and enter the “Dialer Gateways” panel on the dashboard, then add a dialer gateway. Only the fields in bold need to be filled in. That is to say, the name of gateway, for identification, and the Gateway’s field, e.g.</p>
<pre style="margin-top:0px;margin-bottom:18px;padding-top:0px;padding-right:0px;padding-bottom:0px;padding-left:0px;border-top-width:0px;border-right-width:0px;border-bottom-width:0px;border-left-width:0px;border-style:initial;border-color:initial;outline-width:0px;outline-style:initial;outline-color:initial;font-size:10px;vertical-align:baseline;background-color:rgb(236, 236, 236);color:rgb(102, 102, 102);line-height:18px">
sofia/gateway/myprovider/</pre><p style="margin-top:0px;margin-right:0px;margin-bottom:1.5em;margin-left:0px;padding-top:0px;padding-right:0px;padding-bottom:0px;padding-left:0px;border-top-width:0px;border-right-width:0px;border-bottom-width:0px;border-left-width:0px;border-style:initial;border-color:initial;outline-width:0px;outline-style:initial;outline-color:initial;font-size:1.2em;vertical-align:baseline;background-color:rgb(236, 236, 236);color:rgb(102, 102, 102);font-family:Arial, Helvetica, 'Liberation Sans', FreeSans, sans-serif;line-height:18px">
“myprovider” is the name of the gateway which you can take from the gateway name when you typed “sofia status” when configuring the Freeswitch Gateway. Finally, check the status is active, and save. The trunk should now appear in the gateway list.</p>
<div>-MC</div><br><div class="gmail_quote">On Fri, Oct 28, 2011 at 1:28 AM, Dip Mehta <span dir="ltr"><<a href="mailto:dipkumar.mehta@gmail.com" target="_blank">dipkumar.mehta@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Hi,<div><br></div><div>I am trying to integrate Newfies Dialer with Freeswitch. </div><div><br></div><div>This is my sofia status </div><div><br></div><div><div>freeswitch@127.0.0.1@internal> sofia status</div><div> Name Type Data State</div>
<div>=================================================================================================</div><div> <b><u>:XX.XX.XX.XX</u></b> alias internal ALIASED</div>
<div> external profile sip:mod_sofia@<b><u>:XX.XX.XX.XX</u></b>:5080 RUNNING (0)</div><div> external::<a href="http://example.com" target="_blank">example.com</a> gateway <a href="mailto:sip%3Ajoeuser@example.com" target="_blank">sip:joeuser@example.com</a> NOREG</div>
<div> <b><u> external::XX.XX.XX.XX gateway sip:password@XX.XX.XX.XX REGED</u></b></div><div> internal profile <a href="http://sip:mod_sofia@199.167.145.102:5060" target="_blank">sip:mod_sofia@199.167.145.102:5060</a> RUNNING (0)</div>
<div> internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0)</div><div>=================================================================================================</div><div>
3 profiles 1 alias</div></div><div><br></div><div><br></div><div>Looks like the Service provider is now registered with the Freeswitch. I am asuming as it says REGED in the status. </div><div><br></div><div>However, when i start my campaign through the dialer, I am unable to make the outgoing calls which results in these errors. </div>
<div><br></div><div><div>2011-10-28 04:24:41.992674 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables</div><div>2011-10-28 04:24:41.992674 [DEBUG] switch_event.c:1521 Parsing variable [plivo_request_uuid]=[48df3dbe-013e-11e1-8b92-bc305bec2901]</div>
<div>2011-10-28 04:24:41.992674 [DEBUG] switch_event.c:1521 Parsing variable [plivo_answer_url]=[http://<b><u>XX.XX.XX.XX</u></b>:9080/api/dialer_cdr/answercall/]</div><div>2011-10-28 04:24:41.992674 [DEBUG] switch_event.c:1521 Parsing variable [plivo_hangup_url]=[http://<b><u>XX.XX.XX.XX</u></b>:9080/api/dialer_cdr/hangupcall/]</div>
<div>2011-10-28 04:24:41.992674 [DEBUG] switch_event.c:1521 Parsing variable [origination_caller_id_number]=<a href="tel:%5B8888888888" value="+18888888888" target="_blank">[8888888888</a>]</div><div>2011-10-28 04:24:41.992674 [DEBUG] switch_event.c:1521 Parsing variable [bridge_early_media]=[true]</div>
<div>2011-10-28 04:24:41.992674 [DEBUG] switch_event.c:1521 Parsing variable [hangup_after_bridge]=[true]</div><div>2011-10-28 04:24:41.992674 [DEBUG] switch_event.c:1521 Parsing variable [plivo_app]=[true]</div><div>2011-10-28 04:24:41.992674 [DEBUG] switch_event.c:1521 Parsing variable [absolute_codec_string]=[PCMA]</div>
<div>2011-10-28 04:24:41.992674 [DEBUG] switch_event.c:1521 Parsing variable [originate_timeout]=[1800]</div><div>2011-10-28 04:24:41.992674 [DEBUG] switch_event.c:1521 Parsing variable [ignore_early_media]=[true]</div><div>
2011-10-28 04:24:41.992674 [ERR] switch_core_session.c:426 <b>Could not locate channel type</b> <b><u>XX.XX.XX.XX Customers contact number</u></b></div><div>2011-10-28 04:24:41.992674 [NOTICE] switch_ivr_originate.c:2459 Cannot create outgoing channel of type [<b><u>XX.XX.XX.XX Customers contact number</u></b>] cause: [CHAN_NOT_IMPLEMENTED]</div>
<div>2011-10-28 04:24:41.992674 [DEBUG] switch_ivr_originate.c:3348 Originate Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED]</div><div>2011-10-28 04:25:00.052673 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables</div>
<div>2011-10-28 04:25:00.052673 [DEBUG] switch_event.c:1521 Parsing variable [plivo_request_uuid]=[53a1c460-013e-11e1-8b92-bc305bec2901]</div><div>2011-10-28 04:25:00.052673 [DEBUG] switch_event.c:1521 Parsing variable [plivo_answer_url]=[http://<b><u>XX.XX.XX.XX</u></b>:9080/api/dialer_cdr/answercall/]</div>
<div>2011-10-28 04:25:00.052673 [DEBUG] switch_event.c:1521 Parsing variable [plivo_hangup_url]=[http://<b><u>XX.XX.XX.XX</u></b>2:9080/api/dialer_cdr/hangupcall/]</div><div>2011-10-28 04:25:00.052673 [DEBUG] switch_event.c:1521 Parsing variable [origination_caller_id_number]=<a href="tel:%5B8888888888" value="+18888888888" target="_blank">[8888888888</a>]</div>
<div>2011-10-28 04:25:00.052673 [DEBUG] switch_event.c:1521 Parsing variable [bridge_early_media]=[true]</div><div>2011-10-28 04:25:00.052673 [DEBUG] switch_event.c:1521 Parsing variable [hangup_after_bridge]=[true]</div>
<div>2011-10-28 04:25:00.052673 [DEBUG] switch_event.c:1521 Parsing variable [plivo_app]=[true]</div><div>2011-10-28 04:25:00.052673 [DEBUG] switch_event.c:1521 Parsing variable [absolute_codec_string]=[PCMA]</div><div>2011-10-28 04:25:00.052673 [DEBUG] switch_event.c:1521 Parsing variable [originate_timeout]=[1800]</div>
<div>2011-10-28 04:25:00.052673 [DEBUG] switch_event.c:1521 Parsing variable [ignore_early_media]=[true]</div><div>2011-10-28 04:25:00.052673 [ERR] switch_core_session.c:426 Could not locate channel type <b><u>XX.XX.XX.XX Customers contact number</u></b></div>
<div>2011-10-28 04:25:00.052673 [NOTICE] switch_ivr_originate.c:2459 Cannot create outgoing channel of type [<b><u>XX.XX.XX.XX Customers contact number</u></b>] cause: [CHAN_NOT_IMPLEMENTED]</div><div>2011-10-28 04:25:00.052673 [DEBUG] switch_ivr_originate.c:3348 Originate Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED]</div>
<div><br></div></div><div><br></div><div>Please advice.</div><div><br></div><div><br></div>
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<br><br>---------- Forwarded message ----------<br>From: <<a href="mailto:john@mobex.biz">john@mobex.biz</a>><br>To: "FreeSWITCH Users Help" <<a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a>><br>
Date: Fri, 28 Oct 2011 11:36:45 -0700<br>Subject: Re: [Freeswitch-users] High load on database server<br><div><span style="font-family:Verdana;color:#000000;font-size:10pt"><div>I'm probably in over my head, but i could use help.</div>
<div><br></div><div>it appears my Freeswitch went down and came back up (possible from volume) when it did a large volume accoutn is getting bolcked 407 messages:</div><div><br></div><div>Im getting the following message log:</div>
<div><br></div><div>2011-10-28 14:24:54.778973 [DEBUG] sofia.c:7099 IP 66.42.61.24 Rejected by acl "domains". Falling back to Digest auth.<br>2011-10-28 14:24:54.778973 [WARNING] sofia_reg.c:1400 SIP auth challenge (INVITE) on sofia profile 'internal' for [<a href="mailto:18125362643@216.134.222.93" target="_blank">18125362643@216.134.222.93</a>] from ip 66.42.61.24</div>
<div><br></div><div>That IP address is set up as <a href="http://66.42.61.24/32" target="_blank">66.42.61.24/32</a> in the cidr.</div><div><br></div><div>What do i do to get them back up?</div><div><br></div><div>Thanks </div>
<div><br></div><div><br></div><div><br></div><div><br></div></span></div>
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