<div dir="ltr">Hi <span style="font-family:arial,sans-serif;font-size:13px">Moishe,</span><div><span style="font-family:arial,sans-serif;font-size:13px"><br></span></div><div style><span style="font-family:arial,sans-serif;font-size:13px">Thanks for your reply, My phone is behind two routers: first one is a ADSL modem (Draytek Vigor 2110F series) and the second is a Cisco router. I think it is not the NAT problem because the one-way audio is not always happened. I have tested many times in case A call to B. The results are: 2/3 of test calls worked wells, 1/3 of test calls have one-way audio problems and some of calls have one-way problem when the call is established but after waiting for 30 seconds or more the media is established automatically.</span></div>
<div style><span style="font-family:arial,sans-serif;font-size:13px"><br></span></div><div style><span style="font-family:arial,sans-serif;font-size:13px">I have spent a lot of time to identify the problem but i can't understand exactly what is it.</span></div>
<div style><span style="font-family:arial,sans-serif;font-size:13px"><br></span></div><div style><span style="font-family:arial,sans-serif;font-size:13px">Thanks for helping me.</span></div><div style><span style="font-family:arial,sans-serif;font-size:13px"><br>
</span></div><div class="gmail_extra"><br><br><div class="gmail_quote">2014-07-09 14:37 GMT+07:00 <span dir="ltr"><<a href="mailto:freeswitch-users-request@lists.freeswitch.org" target="_blank">freeswitch-users-request@lists.freeswitch.org</a>></span>:<br>
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<br>Today's Topics:<br>
<br>
1. Re: mod_cdr_mongodb and not correct start_uepoch (*_uepoch)<br>
values (Sergey Zhuravlov)<br>
2. Re: Cannot heard sound on FS (Moishe Grunstein)<br>
3. Re: Hangup B leg using featurecode (Samir Doshi)<br>
<br><br>---------- Thư được chuyển tiếp ----------<br>From: Sergey Zhuravlov <<a href="mailto:4orbit@gmail.com">4orbit@gmail.com</a>><br>To: <a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a><br>
Cc: <br>Date: Wed, 9 Jul 2014 09:29:35 +0400<br>Subject: Re: [Freeswitch-users] mod_cdr_mongodb and not correct start_uepoch (*_uepoch) values<br>Steven! Thank you!<br>
<br>
Will use 64-bit platform for mongodb!<br>
<br>
I'll tell you this in the google group CDR-Stats - they forgot to<br>
mention about it.<br>
<br>
> From: Steven Ayre <<a href="mailto:steveayre@gmail.com">steveayre@gmail.com</a>><br>
> To: FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a>><br>
> Cc:<br>
> Date: Tue, 8 Jul 2014 14:28:28 +0100<br>
> Subject: Re: [Freeswitch-users] mod_cdr_mongodb and not correct start_uepoch (*_uepoch) values<br>
> Reading the manual I'm guessing you're on a 32bit platform. Running on 64bit would be better as 32bit limits the size of the database.<br>
><br>
> Integers are 64bit on 64bit platforms, but you can use bigint to get 64bit numbers on a 32bit platform.<br>
> <a href="http://api.mongodb.org/perl/current/MongoDB/DataTypes.html" target="_blank">http://api.mongodb.org/perl/current/MongoDB/DataTypes.html</a><br>
><br>
><br>
> On 8 July 2014 14:23, Steven Ayre <<a href="mailto:steveayre@gmail.com">steveayre@gmail.com</a>> wrote:<br>
>><br>
>> I think the data type you're using in the database is the problem. 2147483647 is the maximum value you can store in a signed 32bit integer. You're trying to store a larger value so it's truncating it to the maximum possible value. Try a type that uses 64bits.<br>
>><br>
>><br>
>> On 8 July 2014 10:47, Sergey Zhuravlov <<a href="mailto:4orbit@gmail.com">4orbit@gmail.com</a>> wrote:<br>
>>><br>
>>> hi! I tune cdr-stats now.<br>
>>><br>
>>> Found that in the database mongodb, which stores data from freeswitch with mod_cdr_mongodb strange value<br>
>>><br>
>>> start_epoch ": 1399643826," start_uepoch ": 2147483647 .... "start_stamp": "2014-05-09 13:57:06"<br>
>>><br>
>>> The problem with the value "start_uepoch": 2147483647<br>
>>><br>
>>> This is not true value should be:<br>
>>><br>
>>> <start_epoch> 1399643826 </ start_epoch><br>
>>> <start_uepoch> 1399643826442976 </ start_uepoch><br>
>>><br>
>>> This is a file from cdr_xml<br>
>>><br>
>>> Someone faced with such behavior mod_cdr_mongodb?<br>
>>> HOWTO correct it?<br>
>>><br>
>>> FreeSWITCH Version 1.5.12b+git~20140419T062924Z~f408dae2af~32bit (git f408dae 2014-04-19 06:29:24Z 32bit)<br>
>>><br>
>>> Ubuntu 12.04.4 LTS<br>
>>><br>
>>><br>
>>><br>
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>>><br>
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>>><br>
<br>
--<br>
WBR, Sergey<br>
<br>
GTALK/<a href="mailto:JABBER%3A4orbit@gmail.com">JABBER:4orbit@gmail.com</a><br>
<br>
<br>
<br><br>---------- Thư được chuyển tiếp ----------<br>From: Moishe Grunstein <<a href="mailto:max@nysolutions.com">max@nysolutions.com</a>><br>To: FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a>><br>
Cc: <br>Date: Wed, 9 Jul 2014 06:25:52 +0000<br>Subject: Re: [Freeswitch-users] Cannot heard sound on FS<br>
<div lang="EN-US" link="blue" vlink="purple">
<div>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">You are having a NAT related issue, what router is doing the NAT, is UPNP enabled in the router? What ports are forwarded to the Freeswitch?<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p>
<p class="MsoNormal"><span style="font-size:7.5pt;font-family:"Verdana","sans-serif";color:#333399">Thanks,<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:7.5pt;font-family:"Verdana","sans-serif";color:#333399"><u></u> <u></u></span></p>
<p class="MsoNormal"><span style="font-size:7.5pt;font-family:"Verdana","sans-serif";color:#333399">Moishe Grunstein<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:7.5pt;font-family:"Verdana","sans-serif";color:#333399">Tornado Computer Systems, Inc.<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:7.5pt;font-family:"Verdana","sans-serif";color:#333399">212.400.7650 <a href="http://888.IPPBX.US" target="_blank">888.IPPBX.US</a><br>
</span><b><span style="font-size:7.5pt;font-family:"Verdana","sans-serif";color:#76923c">Service Request Email:
<a href="mailto:support@nysolutions.com" target="_blank">support@nysolutions.com</a> </span></b><span style="font-size:10.0pt;font-family:"Arial","sans-serif";color:#333399"><u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:7.5pt;font-family:"Verdana","sans-serif";color:#333399">Polycom Certified VAR<br>
Microsoft Small Business Specialist, Cisco SMB Select Certified</span><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#333399"><u></u><u></u></span></p>
<p class="MsoNormal"><a href="http://www.nysolutions.com/" target="_blank"><span style="font-size:10.0pt;font-family:"Arial","sans-serif";color:#1f497d;text-decoration:none"><img border="0" width="147" height="68" src="cid:image001.jpg@01CF9B1D.1D86BCC0" alt="cid:image001.jpg@01C72F94.9EE45D60"></span></a><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:8.0pt;font-family:"Calibri","sans-serif";color:#c0504d">Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions
* Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS</span><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p>
<p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">From:</span></b><span style="font-size:10.0pt;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>Nguy?n Van Nghia Em<br>
<b>Sent:</b> Wednesday, July 09, 2014 12:38 AM<br>
<b>To:</b> <a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a><br>
<b>Subject:</b> [Freeswitch-users] Cannot heard sound on FS<u></u><u></u></span></p>
<p class="MsoNormal"><u></u> <u></u></p>
<div>
<p class="MsoNormal">Dear All,<u></u><u></u></p>
<div>
<p class="MsoNormal"><u></u> <u></u></p>
</div>
<div>
<p class="MsoNormal">I have a FS production deployment:<u></u><u></u></p>
</div>
<div>
<p class="MsoNormal">- A FS server behind a NAT network<u></u><u></u></p>
</div>
<div>
<p class="MsoNormal">- User can register and make calls via internet.<u></u><u></u></p>
</div>
<div>
<p class="MsoNormal"><u></u> <u></u></p>
</div>
<div>
<p class="MsoNormal">It have worked well more than 2 years, but it took problem now:<u></u><u></u></p>
</div>
<div>
<p class="MsoNormal"><u></u> <u></u></p>
</div>
<div>
<p class="MsoNormal">Assume I have following cases:<u></u><u></u></p>
</div>
<div>
<p class="MsoNormal">1. Phone number in PSTN call to Extension in my branch<u></u><u></u></p>
</div>
<div>
<p class="MsoNormal">2. Extensions from another branch call to Extension in my branch (via sip trunk)<u></u><u></u></p>
</div>
<div>
<p class="MsoNormal">3. An extension in my branch call to another extension in my branch<u></u><u></u></p>
</div>
<div>
<p class="MsoNormal">4. An extension in my branch call to a phone number in PSTN <u></u><u></u></p>
</div>
<div>
<p class="MsoNormal">5. An extension in my branch call to a an extension in another branch (via sip trunk)<u></u><u></u></p>
</div>
<div>
<p class="MsoNormal"><u></u> <u></u></p>
</div>
<div>
<p class="MsoNormal">These are results:<u></u><u></u></p>
</div>
<div>
<p class="MsoNormal">Cases 1. 2. and 3.: Caller heard well but sometime Calleed can't heard from caller. If i wait for a longer time (may be 1 minutes, 2 minutes...) Calleed might heard from Caller.<u></u><u></u></p>
</div>
<div>
<p class="MsoNormal">Cases 4.5: Both Caller and Calleed heard well.<u></u><u></u></p>
</div>
<div>
<p class="MsoNormal"><u></u> <u></u></p>
</div>
<div>
<p class="MsoNormal">I have captured PCAP file on FS server, I saw that there were no different between cases 1,2,3 and case 4,5. I have seen many packets that FS server send to calleed IP.<u></u><u></u></p>
</div>
<div>
<p class="MsoNormal"><u></u> <u></u></p>
</div>
<div>
<p class="MsoNormal">There are only one change to my branch network. Network administrator make a NAT (80 and 8000) for camera service at public interface to a private IP in my branch network.<u></u><u></u></p>
</div>
<div>
<p class="MsoNormal"><u></u> <u></u></p>
</div>
<div>
<p class="MsoNormal">My question is: why Calleed in case 1, 2, 3 can not heard from caller ?<u></u><u></u></p>
</div>
<div>
<p class="MsoNormal"><u></u> <u></u></p>
</div>
<div>
<p class="MsoNormal">Another question: FS use RTP for carrying out voice/video but I can only see about 11 RTP packets (for one direction), but there are too many UDP packets in call session. I have attached the picture for that.<u></u><u></u></p>
</div>
<div>
<p class="MsoNormal"><u></u> <u></u></p>
</div>
<div>
<p class="MsoNormal">In the picture:<u></u><u></u></p>
</div>
<div>
<p class="MsoNormal"><a href="http://113.161.90.103" target="_blank">113.161.90.103</a>: public ip address in my branch<u></u><u></u></p>
</div>
<div>
<p class="MsoNormal"><a href="http://192.168.45.4" target="_blank">192.168.45.4</a>: FS server private ip behind NAT<u></u><u></u></p>
</div>
<div>
<p class="MsoNormal"><a href="http://10.12.10.140" target="_blank">10.12.10.140</a>: My AudioCodes Phone's private IP in my branch <u></u><u></u></p>
</div>
<div>
<p class="MsoNormal">Thanks in advance for your help.<br clear="all">
<u></u><u></u></p>
<div>
<p class="MsoNormal"><u></u> <u></u></p>
</div>
<p class="MsoNormal">-- <br>
Best regards,<br>
-------------------------------------------------------------------------------------------------------------<br>
EM NGUYEN<u></u><u></u></p>
</div>
<div>
<p class="MsoNormal">Website: <a href="http://freeswitch.vn" target="_blank">http://freeswitch.vn</a> <br>
Mobile: +84 949669075<br>
Yahoo & Skype: nghiaembt<br>
--------------------------------------------------------------------------------------------------------------
<u></u><u></u></p>
</div>
</div>
</div>
</div>
<br><br>---------- Thư được chuyển tiếp ----------<br>From: Samir Doshi <<a href="mailto:smrdoshi@gmail.com">smrdoshi@gmail.com</a>><br>To: FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a>><br>
Cc: <br>Date: Wed, 9 Jul 2014 13:06:15 +0530<br>Subject: Re: [Freeswitch-users] Hangup B leg using featurecode<br><div dir="ltr"><div>Hi, </div><div><br></div><div>Thanks for the reply. </div><div><br></div><div>Just tried and got below result,</div>
<div><br></div><div>2014-07-09 03:22:17.928601 [DEBUG] switch_ivr_async.c:3555 sofia/default/<a href="mailto:1212@192.168.1.2" target="_blank">1212@192.168.1.2</a> Processing meta digit '0' [uuid_kill::ada0bc56-0739-11e4-bc9e-2fdf7c2f35e5]</div>
<div>2014-07-09 03:22:17.928601 [ERR] switch_core_session.c:2535 Invalid Application uuid_kill</div><div><br></div><div><br></div><div><br></div><div><br></div><div>On Wed, Jul 9, 2014 at 2:51 AM, Bernhard Wendel <span dir="ltr"><<a href="mailto:bernhard@zeppelindesk.com" target="_blank">bernhard@zeppelindesk.com</a>></span> wrote:<br>
</div><div class="gmail_extra"><div class="gmail_quote"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div text="#000000" bgcolor="#FFFFFF">
Hi,<br>
<br>
to me it looks like FS is processing the hangup on the whole call
(i.e. legs a <b>and</b> b). What you want is to only close leg b
and keep leg a. You might try using the uuid_kill command.<br>
Then the last line should look like<br>
<action application="bind_meta_app" data="# a b
uuid_kill::$${uuid_of_b_leg}"/>
<br>
<br>
You still need to replace <i>uuid_of_b_leg</i> with the correct
channel variable. I know there is one, but haven't found it.<br>
<br>
<br>
<div>Am 08.07.2014 10:06, schrieb Sam:<br>
</div><div><div>
<blockquote type="cite">
<pre>Hi Guys,
I need to hangup B leg when A leg user press #0 key in bridge. For that I
have configured below lines in dialplan,
<action application="set" data="continue_on_fail=true"/>
<action application="set" data="hangup_after_bridge=false"/>
<action application="set" data="bind_meta_key=#"/>
<action application="bind_meta_app" data="# a b hangup::NORMAL_CLEARING"/>
However when A leg user pressing #, its dropping A leg as well in place of B
leg only.
Can anybody please point out the issue?
--
View this message in context: <a href="http://freeswitch-users.2379917.n2.nabble.com/Hangup-B-leg-using-featurecode-tp7596115.html" target="_blank">http://freeswitch-users.2379917.n2.nabble.com/Hangup-B-leg-using-featurecode-tp7596115.html</a>
Sent from the freeswitch-users mailing list archive at Nabble.com.
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