<html xmlns:v="urn:schemas-microsoft-com:vml" xmlns:o="urn:schemas-microsoft-com:office:office" xmlns:w="urn:schemas-microsoft-com:office:word" xmlns:x="urn:schemas-microsoft-com:office:excel" xmlns:m="http://schemas.microsoft.com/office/2004/12/omml" xmlns="http://www.w3.org/TR/REC-html40">
<head>
<meta http-equiv=Content-Type content="text/html; charset=us-ascii">
<meta name=Generator content="Microsoft Word 12 (filtered medium)">
<style>
<!--
/* Font Definitions */
@font-face
{font-family:"Cambria Math";
panose-1:2 4 5 3 5 4 6 3 2 4;}
@font-face
{font-family:Calibri;
panose-1:2 15 5 2 2 2 4 3 2 4;}
@font-face
{font-family:Tahoma;
panose-1:2 11 6 4 3 5 4 4 2 4;}
@font-face
{font-family:Consolas;
panose-1:2 11 6 9 2 2 4 3 2 4;}
@font-face
{font-family:"DejaVu Sans";
panose-1:0 0 0 0 0 0 0 0 0 0;}
/* Style Definitions */
p.MsoNormal, li.MsoNormal, div.MsoNormal
{margin:0in;
margin-bottom:.0001pt;
font-size:12.0pt;
font-family:"Times New Roman","serif";
color:black;}
a:link, span.MsoHyperlink
{mso-style-priority:99;
color:blue;
text-decoration:underline;}
a:visited, span.MsoHyperlinkFollowed
{mso-style-priority:99;
color:purple;
text-decoration:underline;}
pre
{mso-style-priority:99;
mso-style-link:"HTML Preformatted Char";
margin:0in;
margin-bottom:.0001pt;
font-size:10.0pt;
font-family:"Courier New";
color:black;}
span.HTMLPreformattedChar
{mso-style-name:"HTML Preformatted Char";
mso-style-priority:99;
mso-style-link:"HTML Preformatted";
font-family:Consolas;
color:black;}
span.EmailStyle19
{mso-style-type:personal-reply;
font-family:"Calibri","sans-serif";
color:#1F497D;}
.MsoChpDefault
{mso-style-type:export-only;
font-size:10.0pt;}
@page Section1
{size:8.5in 11.0in;
margin:1.0in 1.0in 1.0in 1.0in;}
div.Section1
{page:Section1;}
-->
</style>
<!--[if gte mso 9]><xml>
<o:shapedefaults v:ext="edit" spidmax="1026" />
</xml><![endif]--><!--[if gte mso 9]><xml>
<o:shapelayout v:ext="edit">
<o:idmap v:ext="edit" data="1" />
</o:shapelayout></xml><![endif]-->
</head>
<body bgcolor=white lang=EN-US link=blue vlink=purple>
<div class=Section1>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Well do you have a packet filter between the asterisk box and
your phone? Is the phone or the asterisk behind a NAT? Do you have an asymmetric
route for traffic in your network? Does the media take the same path inbound
and outbound between the asterisk and the phone? If you take a packet
capture of the voip segment of both an inbound and outbound call, how do they
differ? Are those calls using different codecs? <o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>If you want to rule out the analog card, construct a loop. When an
inbound call comes in, send it back out to the PSTN. Then create the call and
get a MOS score if you can.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>You should also consider terminating an inbound and an outbound
call against the echo application in asterisk and see if you can further isolate
the problem to either the PSTN channel or the voip channel.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>You really need to try to isolate the problem to a particular
call segment. Once you do that, make sure your test is repeatable. I would be
very hesitant to declare a hardware issue with the Sangoma card. Unidirectional
audio problems are usually caused by a Voip leg since media is handled
separately during that segment of the call. I would be extremely surprised if this
was a hardware issue with a sangoma card.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Good luck, SG <o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<div>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:#1F497D'>--------------------------------------------------<br>
Salvatore Giudice<br>
Salvatore.Giudice@VoIPSecurityTraining.com<br>
<br>
VoIP Security Training, LLC<br>
http://VoIPSecurityTraining.com<br>
<br>
848 N. Rainbow Blvd. #1676<br>
Las Vegas, NV 89107<br>
Phone: (617) 959-7625<br>
Fax: (214) 279-2906<o:p></o:p></span></p>
</div>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<div>
<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'>
<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif";
color:windowtext'>From:</span></b><span style='font-size:10.0pt;font-family:
"Tahoma","sans-serif";color:windowtext'>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>Veselin
Kantsev<br>
<b>Sent:</b> Sunday, December 02, 2007 9:52 AM<br>
<b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br>
<b>Subject:</b> Re: [asterisk-users] Outgoing PSTN calls , unusable voice
quality<o:p></o:p></span></p>
</div>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"DejaVu Sans","serif"'>Dear
Salvatore and Joanna,<br>
Thank you much for both your detailed explanations.<br>
I will surely check my firewall configuration and logs to make sure the VoIP
traffic is passing correctly.</span><o:p></o:p></p>
<div>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"DejaVu Sans","serif"'>However,
I'm a bit confused as the problems that I'm experiencing are with calls made
via the</span><span style='font-family:"DejaVu Sans","serif"'><br>
</span><span style='font-size:10.0pt;font-family:"DejaVu Sans","serif"'>Sangoma
analog card. <br>
So the voice goes from the SIP phone through asterisk through the sangoma card
then </span><span style='font-family:"DejaVu Sans","serif"'><br>
</span><span style='font-size:10.0pt;font-family:"DejaVu Sans","serif"'>directly
into the PSTN and vice versa. There are no firewalls in the way. </span><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"DejaVu Sans","serif"'>Furthermore
incoming calls are OK, the problem is only with outgoing calls when I can hear
the other party well </span><span style='font-family:"DejaVu Sans","serif"'><br>
</span><span style='font-size:10.0pt;font-family:"DejaVu Sans","serif"'>but
they barely understand me.</span><o:p></o:p></p>
</div>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"DejaVu Sans","serif"'><br>
Is there any major difference in the way that Incoming/Outgoing calls are processed
in the above scenario?<br>
Any way that I could trace those processes for faults?<br>
<br>
Thank you again.<br>
<br>
Regards,<br>
Veselin</span><span style='font-size:10.0pt'><br>
</span><br>
Salvatore Giudice wrote: <o:p></o:p></p>
<pre>When you take your packet capture, you'll need to look at the sip messages<o:p></o:p></pre><pre>with SDP attached to get the ip's and ports used for both media streams.<o:p></o:p></pre><pre>Make sure that the ips are correct and that the port used can traverse<o:p></o:p></pre><pre>between those ip's without being blocked by a packet filter or firewall. A<o:p></o:p></pre><pre>lot of times, administrators will set a range of UDP ports that are allowed<o:p></o:p></pre><pre>to pass their packet filter for media and your pbx or phones may be using a<o:p></o:p></pre><pre>different range. This can cause audio loss. You'll need to eliminate that<o:p></o:p></pre><pre>possibility. Sometimes checking your firewall/packet filters for blocks may<o:p></o:p></pre><pre>also prove helpful in identifying problems. You should be aware that the<o:p></o:p></pre><pre>logs from certain firewall products may not be comprehensive. For example,<o:p></o:p></pre><pre>in the past I have seen packets dropped going through netscreens because of<o:p></o:p></pre><pre>invalid headers and no entries appeared in the logs. If you ultimately<o:p></o:p></pre><pre>believe a firewall may be blocking your traffic make sure you setup a<o:p></o:p></pre><pre>capture port or a span on each side of the device and verify the traffic<o:p></o:p></pre><pre>going to and leaving from the firewall using ethereal on a laptop or maybe a<o:p></o:p></pre><pre>Nixon box if you are in a large distributed environment. Never trust a<o:p></o:p></pre><pre>potentially broken device to report accurate information about it's<o:p></o:p></pre><pre>function.<o:p></o:p></pre><pre><o:p> </o:p></pre><pre>TDM = Time Division Multiplexing<o:p></o:p></pre><pre><o:p> </o:p></pre><pre>TDM describes how channels are separated on T1's, etc. It's common to refer<o:p></o:p></pre><pre>to those types of connections as TDM.<o:p></o:p></pre><pre><a
href="http://en.wikipedia.org/wiki/Time-division_multiplexing">http://en.wikipedia.org/wiki/Time-division_multiplexing</a><o:p></o:p></pre><pre><o:p> </o:p></pre><pre><o:p> </o:p></pre><pre>--------------------------------------------------<o:p></o:p></pre><pre>Salvatore Giudice<o:p></o:p></pre><pre><a
href="mailto:Salvatore.Giudice@VoIPSecurityTraining.com">Salvatore.Giudice@VoIPSecurityTraining.com</a><o:p></o:p></pre><pre><o:p> </o:p></pre><pre>VoIP Security Training, LLC<o:p></o:p></pre><pre><a
href="http://VoIPSecurityTraining.com">http://VoIPSecurityTraining.com</a><o:p></o:p></pre><pre><o:p> </o:p></pre><pre>848 N. Rainbow Blvd. #1676<o:p></o:p></pre><pre>Las Vegas, NV 89107<o:p></o:p></pre><pre>Phone: (617) 959-7625<o:p></o:p></pre><pre>Fax: (214) 279-2906<o:p></o:p></pre><pre><o:p> </o:p></pre><pre><o:p> </o:p></pre><pre>-----Original Message-----<o:p></o:p></pre><pre>From: <a
href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a><o:p></o:p></pre><pre>[<a
href="mailto:asterisk-users-bounces@lists.digium.com">mailto:asterisk-users-bounces@lists.digium.com</a>] On Behalf Of Veselin<o:p></o:p></pre><pre>Kantsev<o:p></o:p></pre><pre>Sent: Friday, November 30, 2007 8:28 PM<o:p></o:p></pre><pre>To: Asterisk Users Mailing List - Non-Commercial Discussion<o:p></o:p></pre><pre>Subject: Re: [asterisk-users] Outgoing PSTN calls , unusable voice quality<o:p></o:p></pre><pre><o:p> </o:p></pre><pre>Thank you much for the prompt reply Salvatore.<o:p></o:p></pre><pre><o:p> </o:p></pre><pre>Would you have the time to explain further how should I go for verifying<o:p></o:p></pre><pre>that SDP and RTP are OK.<o:p></o:p></pre><pre>Also what is reffered to as the TDM site.<o:p></o:p></pre><pre><o:p> </o:p></pre><pre>Veselin<o:p></o:p></pre><pre><o:p> </o:p></pre><pre>On Fri, Nov 30, 2007 at 05:01:17PM -0500, Salvatore Giudice wrote:<o:p></o:p></pre><pre> <o:p></o:p></pre>
<blockquote style='margin-top:5.0pt;margin-bottom:5.0pt'><pre>Take a packet capture of your VoIP segment and verify that the SDP is<o:p></o:p></pre><pre>correct and that the RTP is making it to the correct places. If all that<o:p></o:p></pre><pre>looks good and this is a straight out quality problem, then you need to<o:p></o:p></pre><pre>figure out if it's happening on the voip side or on the TDM side. You<o:p></o:p></pre><pre> <o:p></o:p></pre></blockquote>
<pre>should<o:p></o:p></pre><pre> <o:p></o:p></pre>
<blockquote style='margin-top:5.0pt;margin-bottom:5.0pt'><pre>make calls (with captures) VoIP to Voip passing the media through your<o:p></o:p></pre><pre>asterisk and also try routing a tdm call in and back out. If you have the<o:p></o:p></pre><pre>equipment, take a mos score of the TDM loop.<o:p></o:p></pre><pre><o:p> </o:p></pre><pre>Without any of the above, you will not be able to isolate the issue.<o:p></o:p></pre><pre><o:p> </o:p></pre><pre>--------------------------------------------------<o:p></o:p></pre><pre>Salvatore Giudice<o:p></o:p></pre><pre><a
href="mailto:Salvatore.Giudice@VoIPSecurityTraining.com">Salvatore.Giudice@VoIPSecurityTraining.com</a><o:p></o:p></pre><pre><o:p> </o:p></pre><pre>VoIP Security Training, LLC<o:p></o:p></pre><pre><a
href="http://VoIPSecurityTraining.com">http://VoIPSecurityTraining.com</a><o:p></o:p></pre><pre><o:p> </o:p></pre><pre>848 N. Rainbow Blvd. #1676<o:p></o:p></pre><pre>Las Vegas, NV 89107<o:p></o:p></pre><pre>Phone: (617) 959-7625<o:p></o:p></pre><pre>Fax: (214) 279-2906<o:p></o:p></pre><pre><o:p> </o:p></pre><pre><o:p> </o:p></pre><pre>-----Original Message-----<o:p></o:p></pre><pre>From: <a
href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a><o:p></o:p></pre><pre>[<a
href="mailto:asterisk-users-bounces@lists.digium.com">mailto:asterisk-users-bounces@lists.digium.com</a>] On Behalf Of Veselin<o:p></o:p></pre><pre>Kantsev<o:p></o:p></pre><pre>Sent: Friday, November 30, 2007 2:47 PM<o:p></o:p></pre><pre>To: <a
href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a><o:p></o:p></pre><pre>Subject: [asterisk-users] Outgoing PSTN calls , unusable voice quality<o:p></o:p></pre><pre><o:p> </o:p></pre><pre>Hello,<o:p></o:p></pre><pre>I have an Asterisk running with a Sangoma A200 card with Hardware Echo <o:p></o:p></pre><pre>cancelling connected to the UK PSTN.<o:p></o:p></pre><pre>If a PSTN call comes in, voice both ways is OK, however if an outgoing <o:p></o:p></pre><pre>call over the PSTN is made I can hear the other party OK but they can <o:p></o:p></pre><pre>not, they can barely understand what I am saying, my voice is unclear <o:p></o:p></pre><pre>fading and skipping.<o:p></o:p></pre><pre>Internal SIP and IAX2 calls are OK, incoming/outgoing calls over IAX2 <o:p></o:p></pre><pre>are OK too. I've tried gsm/ulaw/alaw codecs so far.<o:p></o:p></pre><pre>Tried disabling the echo cancelling as well.<o:p></o:p></pre><pre><o:p> </o:p></pre><pre>Any suggestions will be greatly appreciated.<o:p></o:p></pre><pre><o:p> </o:p></pre><pre><o:p> </o:p></pre><pre>Regards,<o:p></o:p></pre><pre>Veselin<o:p></o:p></pre><pre><o:p> </o:p></pre><pre>_______________________________________________<o:p></o:p></pre><pre>--Bandwidth and Colocation Provided by <a
href="http://www.api-digital.com">http://www.api-digital.com</a>--<o:p></o:p></pre><pre><o:p> </o:p></pre><pre>asterisk-users mailing list<o:p></o:p></pre><pre>To UNSUBSCRIBE or update options visit:<o:p></o:p></pre><pre> <a
href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a><o:p></o:p></pre><pre><o:p> </o:p></pre><pre><o:p> </o:p></pre><pre><o:p> </o:p></pre><pre>_______________________________________________<o:p></o:p></pre><pre>--Bandwidth and Colocation Provided by <a
href="http://www.api-digital.com">http://www.api-digital.com</a>--<o:p></o:p></pre><pre><o:p> </o:p></pre><pre>asterisk-users mailing list<o:p></o:p></pre><pre>To UNSUBSCRIBE or update options visit:<o:p></o:p></pre><pre> <a
href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a><o:p></o:p></pre><pre> <o:p></o:p></pre></blockquote>
<pre><o:p> </o:p></pre><pre><o:p> </o:p></pre><pre>_______________________________________________<o:p></o:p></pre><pre>--Bandwidth and Colocation Provided by <a
href="http://www.api-digital.com">http://www.api-digital.com</a>--<o:p></o:p></pre><pre><o:p> </o:p></pre><pre>asterisk-users mailing list<o:p></o:p></pre><pre>To UNSUBSCRIBE or update options visit:<o:p></o:p></pre><pre> <a
href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a><o:p></o:p></pre><pre><o:p> </o:p></pre><pre><o:p> </o:p></pre><pre><o:p> </o:p></pre><pre>_______________________________________________<o:p></o:p></pre><pre>--Bandwidth and Colocation Provided by <a
href="http://www.api-digital.com">http://www.api-digital.com</a>--<o:p></o:p></pre><pre><o:p> </o:p></pre><pre>asterisk-users mailing list<o:p></o:p></pre><pre>To UNSUBSCRIBE or update options visit:<o:p></o:p></pre><pre> <a
href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a><o:p></o:p></pre><pre> <o:p></o:p></pre>
<p class=MsoNormal><br>
<br>
<br>
<o:p></o:p></p>
<pre>-- <o:p></o:p></pre><pre>Regards,<o:p></o:p></pre><pre>Veselin Kantsev<o:p></o:p></pre><pre>Campbell-Lange Workshop<o:p></o:p></pre></div>
</body>
</html>