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<DIV dir=ltr align=left><FONT face=Arial color=#0000ff size=2><SPAN
class=052514804-12122007>The two boxes are labeled as per the town they are in:
Leongatha and Korumburra.</SPAN></FONT></DIV>
<DIV dir=ltr align=left><FONT face=Arial color=#0000ff size=2><SPAN
class=052514804-12122007></SPAN></FONT> </DIV>
<DIV dir=ltr align=left><FONT face=Arial color=#0000ff size=2><SPAN
class=052514804-12122007>The receptionist is in Korumburra.</SPAN></FONT></DIV>
<DIV dir=ltr align=left><FONT face=Arial color=#0000ff size=2><SPAN
class=052514804-12122007></SPAN></FONT> </DIV>
<DIV dir=ltr align=left><FONT face=Arial color=#0000ff size=2><SPAN
class=052514804-12122007>When a call comes in off the PSTN in Leongatha, the
first number in the call queue is the receptionist. If she answers it, then the
media flow looks like this:</SPAN></FONT></DIV>
<DIV dir=ltr align=left><FONT face=Arial color=#0000ff size=2><SPAN
class=052514804-12122007></SPAN></FONT> </DIV>
<DIV dir=ltr align=left><FONT face=Arial color=#0000ff size=2><SPAN
class=052514804-12122007>PSTN -> Leongatha -> IAX Trunk -> Korumburra
-> Receptionist Phone</SPAN></FONT></DIV>
<DIV dir=ltr align=left><FONT face=Arial color=#0000ff size=2><SPAN
class=052514804-12122007></SPAN></FONT> </DIV>
<DIV dir=ltr align=left><FONT face=Arial color=#0000ff size=2><SPAN
class=052514804-12122007>If she then transfers the call back to a Leongatha
extension, the media path looks like this:</SPAN></FONT></DIV>
<DIV dir=ltr align=left><FONT face=Arial color=#0000ff size=2><SPAN
class=052514804-12122007></SPAN></FONT> </DIV>
<DIV dir=ltr align=left><FONT face=Arial color=#0000ff size=2><SPAN
class=052514804-12122007><SPAN class=052514804-12122007>PSTN -> Leongatha
-> IAX Trunk -> Korumburra -> IAX Trunk -> Leongatha -> IP
Phone</SPAN></SPAN></FONT></DIV>
<DIV dir=ltr align=left><FONT face=Arial color=#0000ff size=2><SPAN
class=052514804-12122007><SPAN
class=052514804-12122007></SPAN></SPAN></FONT> </DIV>
<DIV dir=ltr align=left><FONT face=Arial color=#0000ff size=2><SPAN
class=052514804-12122007><SPAN
class=052514804-12122007></SPAN></SPAN></FONT> </DIV>
<DIV dir=ltr align=left><FONT face=Arial color=#0000ff size=2><SPAN
class=052514804-12122007><SPAN class=052514804-12122007>I believe that it is
possible to stop this 'bouncing' of the call from happening by using the
re-invite feature. However, taking the Trixbox's out of the media path is
undesirable, as they client needs to be able to record calls. Also, doing this
does not 'fix' the underlying problem.</SPAN></SPAN></FONT></DIV>
<DIV dir=ltr align=left><FONT face=Arial color=#0000ff size=2><SPAN
class=052514804-12122007><SPAN class=052514804-12122007>They are also having
some issues with outbound calls from Leongatha (over VoIP), and they are having
no real issues at Korumburra.</SPAN></SPAN></FONT></DIV>
<DIV dir=ltr align=left><FONT face=Arial color=#0000ff size=2><SPAN
class=052514804-12122007><SPAN
class=052514804-12122007></SPAN></SPAN></FONT> </DIV>
<DIV dir=ltr align=left><FONT face=Arial color=#0000ff size=2><SPAN
class=052514804-12122007><SPAN class=052514804-12122007>Many
Thanks,</SPAN></SPAN></FONT></DIV>
<DIV> </DIV>
<DIV class=Section1>
<P class=MsoNormal align=left><STRONG><SPAN
style="COLOR: #666666; FONT-FAMILY: Arial">Daniel Cole<SPAN
style="mso-spacerun: yes"> </SPAN></SPAN></STRONG><STRONG><SPAN
style="FONT-SIZE: 8pt; COLOR: #666666; FONT-FAMILY: Arial">(CCNA)</SPAN></STRONG><STRONG><SPAN
style="COLOR: #666666; FONT-FAMILY: Arial"> </SPAN></STRONG><SPAN
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style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"> </SPAN><SPAN
style="FONT-SIZE: 10pt; COLOR: green; FONT-FAMILY: 'Harmony Text'">Please
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<HR tabIndex=-1>
<FONT face=Tahoma size=2><B>From:</B> asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <B>On Behalf Of </B>Alexander
Lopez<BR><B>Sent:</B> Wednesday, 12 December 2007 3:11 PM<BR><B>To:</B> Asterisk
Users Mailing List - Non-Commercial Discussion<BR><B>Subject:</B> Re:
[asterisk-users] Call Quality Issues With 2 Trixbox's -
RouterIssue?<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV class=Section1>
<P class=MsoNormal><FONT face=Arial color=navy size=2><SPAN
style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">How are the calls being
transferred from Box A to Box B?<o:p></o:p></SPAN></FONT></P>
<P class=MsoNormal><FONT face=Arial color=navy size=2><SPAN
style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><o:p> </o:p></SPAN></FONT></P>
<P class=MsoNormal><FONT face=Arial color=navy size=2><SPAN
style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">On what box is the
receptionist registered too?<o:p></o:p></SPAN></FONT></P>
<P class=MsoNormal><FONT face=Arial color=navy size=2><SPAN
style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><o:p> </o:p></SPAN></FONT></P>
<P class=MsoNormal><FONT face=Arial color=navy size=2><SPAN
style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><o:p> </o:p></SPAN></FONT></P>
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<P class=MsoNormal><B><FONT face=Tahoma size=2><SPAN
style="FONT-WEIGHT: bold; FONT-SIZE: 10pt; FONT-FAMILY: Tahoma">From:</SPAN></FONT></B><FONT
face=Tahoma size=2><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: Tahoma">
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <B><SPAN
style="FONT-WEIGHT: bold">On Behalf Of </SPAN></B>Daniel Cole<BR><B><SPAN
style="FONT-WEIGHT: bold">Sent:</SPAN></B> Tuesday, December 11, 2007 9:00
PM<BR><B><SPAN style="FONT-WEIGHT: bold">To:</SPAN></B>
asterisk-users@lists.digium.com<BR><B><SPAN
style="FONT-WEIGHT: bold">Subject:</SPAN></B> [asterisk-users] Call Quality
Issues With 2 Trixbox's - RouterIssue?</SPAN></FONT><o:p></o:p></P></DIV>
<P class=MsoNormal><FONT face="Times New Roman" size=3><SPAN
style="FONT-SIZE: 12pt"><o:p> </o:p></SPAN></FONT></P>
<DIV>
<P><FONT face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial">Hello
Everyone,<o:p></o:p></SPAN></FONT></P>
<P><FONT face=Arial size=2><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: Arial">We
have recently installed a pair of Trixbox servers in for a client of our. They
have two locations, with one server each. The servers terminate 3 standard POTS
lines into a Sangoma A200D card. The servers are IBM x3250 1RU servers (1GB Ram,
Raid 1 160GB HDD, Dual Core Xenon Processors). We are using Trixbox 2.2, and
G729 all around.<o:p></o:p></SPAN></FONT></P>
<P><FONT face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial">Each site has two (2) 512k/512k ADSL
connections terminating into a Cisco 877W router (using an additional 'dumb'
modem in a separate VLAN for the extra dsl connection). Using policy based
routing, all Voice Data goes over one DSL connection (the one that terminates
directly into the router), and all other traffic (e.g. Web and VPN) goes out the
second connection (the bridged dumb dsl modem).<o:p></o:p></SPAN></FONT></P>
<P><FONT face=Arial size=2><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: Arial">We
are also the ISP for this client, and as thus we have full monitoring of our
Layer 2 and Layer 3 networks. From our analysis, it doesn't appear that there is
any issue in these networks. We have other customers using the VoIP service, who
have not complained of these issues.<o:p></o:p></SPAN></FONT></P>
<P><FONT face=Arial size=2><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: Arial">Now
for the Fun part!<BR>The client is complaining of issues with inter-site calls.
They are reporting issues with crackly and broken speech, and horrible jitter
(or packet loss). This presents a huge issues, because they have one
receptionist answering all calls for both sites. So if a call comes in from the
other site, it automatically an inter-site call, and the quality falls out of
it. If the call is then transfered back to the originating site, the audio
'bounces' between the two sites, which add to the call quality
degradation.<o:p></o:p></SPAN></FONT></P>
<P><FONT face=Arial size=2><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: Arial">We
have been monitoring the router while these incidents have been reported, and it
does not appear to be a bandwidth issue. The DSL tail used for Voice gets to no
more then 120k in each direction (we have tested the links, and can pull data at
53k/s between sites). CPU usage floats at around 20-25% under load. The router
has only shows major packet loss (that we can tell) when REALLY pushing it in
testing (e.g. 10+ calls between sites).<BR>We have enabled the SIP jitter
buffer, as well as the IAX jitter buffer, which appeared to make a huge
difference, but the issue is still ongoing.<o:p></o:p></SPAN></FONT></P>
<P><FONT face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial">These issues have also been reported
with some outbound VoIP calls. Internal calls, and calls directly in or out of
the Sangoma card are clear, with no issues
reported.<o:p></o:p></SPAN></FONT></P>
<P><FONT face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial">Does anyone have any thoughts on
what could be causing these issues? We have been racking our brains here, and
have tried everything that we can think of. These system is a million times
better then what is what when it was first installed, but it is still not where
it should be in terms of quality.<o:p></o:p></SPAN></FONT></P>
<P><FONT face=Arial size=2><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: Arial">Any
thoughts/ideas are most welcome.<o:p></o:p></SPAN></FONT></P>
<P><FONT face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial">Thank
you<o:p></o:p></SPAN></FONT></P></DIV>
<DIV>
<P class=MsoNormal><FONT face="Times New Roman" size=3><SPAN
style="FONT-SIZE: 12pt"> <o:p></o:p></SPAN></FONT></P></DIV>
<DIV>
<P class=MsoNormal
style="mso-margin-top-alt: auto; mso-margin-bottom-alt: auto"><STRONG><B><FONT
face=Arial color=#666666 size=3><SPAN
style="FONT-SIZE: 12pt; COLOR: #666666; FONT-FAMILY: Arial">Daniel Cole
</SPAN></FONT></B></STRONG><STRONG><B><FONT face=Arial color=#666666
size=1><SPAN
style="FONT-SIZE: 8pt; COLOR: #666666; FONT-FAMILY: Arial">(CCNA)</SPAN></FONT></B></STRONG><STRONG><B><FONT
face=Arial color=#666666><SPAN
style="COLOR: #666666; FONT-FAMILY: Arial"> </SPAN></FONT></B></STRONG><o:p></o:p></P></DIV>
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