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<font face="DejaVu Sans"><small><big><small>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.<br>
</small></big></small></font>
<div align="justify"><font face="DejaVu Sans"><small><big><small>However,
I'm a bit confused as the problems that I'm experiencing are with calls
made via the</small><br>
<small>Sangoma analog card. <br>
So the voice goes from the SIP phone through asterisk through the
sangoma card then </small><br>
<small>directly into the PSTN and vice versa. There are no firewalls in
the way. </small><br>
</big></small></font></div>
<div align="justify"><font face="DejaVu Sans"><small><big><small>Furthermore
incoming calls are OK, the problem is only with outgoing calls when I
can hear the other party well </small><br>
<small>but they barely understand me.</small><br>
</big></small></font></div>
<small><font face="DejaVu Sans"><small><big><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</big></small></font><br>
</small><br>
Salvatore Giudice wrote:
<blockquote
cite="mid00e201c83459$ccba1520$662e3f60$@Giudice@VoIPSecurityTraining.com"
type="cite">
<pre wrap="">When you take your packet capture, you'll need to look at the sip messages
with SDP attached to get the ip's and ports used for both media streams.
Make sure that the ips are correct and that the port used can traverse
between those ip's without being blocked by a packet filter or firewall. A
lot of times, administrators will set a range of UDP ports that are allowed
to pass their packet filter for media and your pbx or phones may be using a
different range. This can cause audio loss. You'll need to eliminate that
possibility. Sometimes checking your firewall/packet filters for blocks may
also prove helpful in identifying problems. You should be aware that the
logs from certain firewall products may not be comprehensive. For example,
in the past I have seen packets dropped going through netscreens because of
invalid headers and no entries appeared in the logs. If you ultimately
believe a firewall may be blocking your traffic make sure you setup a
capture port or a span on each side of the device and verify the traffic
going to and leaving from the firewall using ethereal on a laptop or maybe a
Nixon box if you are in a large distributed environment. Never trust a
potentially broken device to report accurate information about it's
function.
TDM = Time Division Multiplexing
TDM describes how channels are separated on T1's, etc. It's common to refer
to those types of connections as TDM.
<a class="moz-txt-link-freetext" href="http://en.wikipedia.org/wiki/Time-division_multiplexing">http://en.wikipedia.org/wiki/Time-division_multiplexing</a>
--------------------------------------------------
Salvatore Giudice
<a class="moz-txt-link-abbreviated" href="mailto:Salvatore.Giudice@VoIPSecurityTraining.com">Salvatore.Giudice@VoIPSecurityTraining.com</a>
VoIP Security Training, LLC
<a class="moz-txt-link-freetext" href="http://VoIPSecurityTraining.com">http://VoIPSecurityTraining.com</a>
848 N. Rainbow Blvd. #1676
Las Vegas, NV 89107
Phone: (617) 959-7625
Fax: (214) 279-2906
-----Original Message-----
From: <a class="moz-txt-link-abbreviated" href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>
[<a class="moz-txt-link-freetext" href="mailto:asterisk-users-bounces@lists.digium.com">mailto:asterisk-users-bounces@lists.digium.com</a>] On Behalf Of Veselin
Kantsev
Sent: Friday, November 30, 2007 8:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Outgoing PSTN calls , unusable voice quality
Thank you much for the prompt reply Salvatore.
Would you have the time to explain further how should I go for verifying
that SDP and RTP are OK.
Also what is reffered to as the TDM site.
Veselin
On Fri, Nov 30, 2007 at 05:01:17PM -0500, Salvatore Giudice wrote:
</pre>
<blockquote type="cite">
<pre wrap="">Take a packet capture of your VoIP segment and verify that the SDP is
correct and that the RTP is making it to the correct places. If all that
looks good and this is a straight out quality problem, then you need to
figure out if it's happening on the voip side or on the TDM side. You
</pre>
</blockquote>
<pre wrap=""><!---->should
</pre>
<blockquote type="cite">
<pre wrap="">make calls (with captures) VoIP to Voip passing the media through your
asterisk and also try routing a tdm call in and back out. If you have the
equipment, take a mos score of the TDM loop.
Without any of the above, you will not be able to isolate the issue.
--------------------------------------------------
Salvatore Giudice
<a class="moz-txt-link-abbreviated" href="mailto:Salvatore.Giudice@VoIPSecurityTraining.com">Salvatore.Giudice@VoIPSecurityTraining.com</a>
VoIP Security Training, LLC
<a class="moz-txt-link-freetext" href="http://VoIPSecurityTraining.com">http://VoIPSecurityTraining.com</a>
848 N. Rainbow Blvd. #1676
Las Vegas, NV 89107
Phone: (617) 959-7625
Fax: (214) 279-2906
-----Original Message-----
From: <a class="moz-txt-link-abbreviated" href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>
[<a class="moz-txt-link-freetext" href="mailto:asterisk-users-bounces@lists.digium.com">mailto:asterisk-users-bounces@lists.digium.com</a>] On Behalf Of Veselin
Kantsev
Sent: Friday, November 30, 2007 2:47 PM
To: <a class="moz-txt-link-abbreviated" href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>
Subject: [asterisk-users] Outgoing PSTN calls , unusable voice quality
Hello,
I have an Asterisk running with a Sangoma A200 card with Hardware Echo
cancelling connected to the UK PSTN.
If a PSTN call comes in, voice both ways is OK, however if an outgoing
call over the PSTN is made I can hear the other party OK but they can
not, they can barely understand what I am saying, my voice is unclear
fading and skipping.
Internal SIP and IAX2 calls are OK, incoming/outgoing calls over IAX2
are OK too. I've tried gsm/ulaw/alaw codecs so far.
Tried disabling the echo cancelling as well.
Any suggestions will be greatly appreciated.
Regards,
Veselin
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</pre>
</blockquote>
<pre wrap=""><!---->
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</pre>
</blockquote>
<br>
<br>
<pre class="moz-signature" cols="72">--
Regards,
Veselin Kantsev
Campbell-Lange Workshop
</pre>
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