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<DIV><SPAN class=542235707-24032007><FONT face=Arial color=#0000ff size=2>Hi
men,</FONT></SPAN></DIV>
<DIV><SPAN class=542235707-24032007><FONT face=Arial color=#0000ff
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=542235707-24032007><FONT face=Arial color=#0000ff size=2>I have
already encountered some issue like this with few switches (very known great
brand) which doesn't like VoIP traffic !</FONT></SPAN></DIV>
<DIV><SPAN class=542235707-24032007><FONT face=Arial color=#0000ff size=2>Check
by drectly connected the VoIP equipment - if you can - with temporary long
Ethernet cables bypassing the tested switch to see what happens in this
case.</FONT></SPAN></DIV>
<DIV><SPAN class=542235707-24032007><FONT face=Arial color=#0000ff size=2>You
can also tell to "qualify" with a longer delay, but this could not help in case
of regulary frames losses.</FONT></SPAN></DIV>
<DIV><SPAN class=542235707-24032007><FONT face=Arial color=#0000ff
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=542235707-24032007><FONT face=Arial color=#0000ff size=2>Good
luck !</FONT></SPAN></DIV>
<DIV><SPAN class=542235707-24032007><FONT face=Arial color=#0000ff
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=542235707-24032007><FONT face=Arial color=#0000ff
size=2>Francois BERGERET,</FONT></SPAN></DIV>
<DIV><SPAN class=542235707-24032007><FONT face=Arial color=#0000ff
size=2>France.</FONT></SPAN></DIV>
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<DIV></DIV>
<DIV class=OutlookMessageHeader lang=fr dir=ltr align=left><FONT face=Tahoma
size=2>-----Message d'origine-----<BR><B>De :</B>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <B>De la part de</B> Rajeev
Natarajan<BR><B>Envoyé :</B> samedi 24 mars 2007 08:14<BR><B>À :</B>
Asterisk Users Mailing List - Non-Commercial Discussion<BR><B>Objet :</B>
Re: [asterisk-users] SIP/IAX peers UNREACHABLE and audio
loss<BR><BR></FONT></DIV>Well, we have add similar issues - do you use a media
gateway /.IP Phones / softphones as your extensions?<BR><BR>We were running
Audiocodes and for some reason (I suspect a poor ethernet switch), when there
are more than 15 people using the line, Audiocodes will not respond to a
qualify and asterisk will drop the call. Turned off qualify (removed
qualify=yes) and <still keeping fingers crossed> things seem fine.
<BR><BR>Rajeev<BR><BR>
<DIV><SPAN class=gmail_quote>On 3/23/07, <B class=gmail_sendername>Edoardo
Serra</B> <<A
href="mailto:edoardo.serra@webrainstorm.it">edoardo.serra@webrainstorm.it</A>>
wrote:</SPAN>
<BLOCKQUOTE class=gmail_quote
style="PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: rgb(204,204,204) 1px solid">Hi
all,<BR> I'm having a problem
with some Asterisk servers interconnected with<BR>each other using IAX (I
also tried with SIP without solving the problem)<BR><BR>Sometimes, with
apparently no reason, some peers become UNREACHABLE <BR>(I have qualify=yes
in iax.conf) and REACHABLE again as soon as<BR>another qualify test is
made.<BR><BR>Our users are also complaining about audio loss during their
calls,<BR>apparently randomly, everything goes ok for days and bad for
another few <BR>days.<BR><BR>I strongly believe the 2 problems are strictly
related because in the<BR>logs I see REACHABLE / UNREACHABLE messages only
for certains days<BR>without regularity.<BR>The days in wich i see a lot of
messages are exactly the days with <BR>most of complaint about audio
loss<BR><BR>I just noticed that timestamps of the logs (REACHABLE /
UNREACHABLE)<BR>are quite always during business hours, this makes me think
at somewhat<BR>related to load (cpu load, badwidth load, calls load, etc...)
<BR><BR>But, looking at hardware specs of our lan, servers and average load
I<BR>don't think they are over-stressed.<BR><BR>Our servers are all:<BR>2 x
Intel(R) Xeon(TM) CPU 3.20GHz<BR>1 GB RAM<BR>2 x IDE HDDs Software RAID 1
<BR>Asterisk 1.2.13 with res_perl<BR>Gentoo Linux<BR>Some of them has a
Sangoma card connected with an E1<BR><BR>Most ot these are on the same LAN,
interconnected with a 1 GB switch<BR>(I don't think it should be a bandwidth
problem). <BR><BR>Load averages of these server is varying from 0.5 to
1.0<BR>(I guess it should be ok)<BR><BR>On each server we don't have more
than 50 concurrent calls<BR>(bridged SIP <-> IAX2 or IAX2 <->
ZAP)<BR><BR>Used codec is mostly G729<BR><BR>Sometimes on asterisk cli i see
some messages like<BR>"Avoided initial deadlock for '0x9fd130', 10
retries!"<BR>I don't know if it could be somehow related.<BR><BR>Someone of
you can point me in the right direction ?<BR><BR>Tnx in
advance<BR><BR>Regards<BR><BR>Ing. Edoardo Serra<BR>WeBRainstorm
S.r.l.<BR><BR>_______________________________________________<BR>--Bandwidth
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