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<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'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Where do I get oej’s patch, and how do I install it?<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>
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<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"'>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>Andreas
Brodmann<br>
<b>Sent:</b> Tuesday, June 05, 2007 2:58 AM<br>
<b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br>
<b>Subject:</b> Re: [asterisk-users] Calls being dropped<o:p></o:p></span></p>
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<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal style='margin-bottom:12.0pt'>We have a similar problem at
our place, since a few months.<br>
<br>
oej, mentioned a patch he has made after the release of asterisk-1.4.4. So
we're<br>
all desperately waiting for asterisk-1.4.5 to be released; unless you want to
install <br>
from svn.<br>
<br>
<br>
<o:p></o:p></p>
<div>
<p class=MsoNormal><span class=gmailquote>2007/6/4, Compnet Bobby <<a
href="mailto:compnetbobby@hotmail.com">compnetbobby@hotmail.com</a>>:</span><o:p></o:p></p>
<div>
<div>
<p> <o:p></o:p></p>
<p>We have the latest version of asterisk, on a xeon dell server (2gb ram),
with 6 snom320's(latest firmware) and 3 grandstream gxp 2000's (latest stable
firmware) and are having a few problems. We have a basic menu that transfers
calls to different extensions. The problems can be found on all extensions. We
have 2 different incoming providers and the problem happens on both providers. <o:p></o:p></p>
<p> <o:p></o:p></p>
<p>I want your input on 2 problems, they are the following:<o:p></o:p></p>
<p> <o:p></o:p></p>
<p>1.<o:p></o:p></p>
<p> <o:p></o:p></p>
<p>60% of the time everything works fine and there are no problems, 40% of
times when the calls are transferred to an extension, after a few seconds
, the call drops. The log from the server is below(this is from pickup to
hangup, the main area of concern is where it says warning). <o:p></o:p></p>
<p> <o:p></o:p></p>
<p> <o:p></o:p></p>
<p> -- Executing [9097406868@from-sip:1]
Answer("SIP/9097406868-09e110f8", "") in new stack<o:p></o:p></p>
<p> -- Executing [9097406868@from-sip:2]
BackGround("SIP/9097406868-09e110f8",
"menus/welcome-to-exec") in new stack<o:p></o:p></p>
<p> -- <SIP/9097406868-09e110f8> Playing
'menus/welcome-to-exec' (language 'en')<o:p></o:p></p>
<p> == CDR updated on SIP/9097406868-09e110f8<o:p></o:p></p>
<p> -- Executing [103@from-sip:1]
Dial("SIP/9097406868-09e110f8", "SIP/103|50|m") in new
stack<o:p></o:p></p>
<p> -- Called 103<o:p></o:p></p>
<p> -- Started music on hold, class 'default', on
SIP/9097406868-09e110f8<o:p></o:p></p>
<p> -- SIP/103-09dedd68 is ringing<o:p></o:p></p>
<p>[May 29 09:05:22] WARNING[2678]: chan_sip.c:1899 retrans_pkt: Maximum
retries exceeded on transmission <a
href="mailto:LAXMGC0120070529230718052251@209.244.63.13" target="_blank">LAXMGC0120070529230718052251@209.244.63.13
</a>for seqno 1 (Critical Response)<o:p></o:p></p>
<p>[May 29 09:05:22] WARNING[2678]: chan_sip.c:1916 retrans_pkt: Hanging up
call <a href="mailto:LAXMGC0120070529230718052251@209.244.63.13" target="_blank">LAXMGC0120070529230718052251@209.244.63.13</a>
- no reply to our critical packet.<o:p></o:p></p>
<p> -- Stopped music on hold on SIP/9097406868-09e110f8<o:p></o:p></p>
<p> == Spawn extension (from-sip, 103, 1) exited non-zero on
'SIP/9097406868-09e110f8'<o:p></o:p></p>
<p> <o:p></o:p></p>
<p> <o:p></o:p></p>
<p>2. When a call comes in or is transferred(not on outgoing), there is a delay
until the person on the incoming line can hear you. We can hear them, but they
can't hear us. Sometimes there is no delay, sometimes for person calling in
cant hear you for 6 seconds. <o:p></o:p></p>
<p> <o:p></o:p></p>
<p> <o:p></o:p></p>
<p>Thanks for the help in advance!!!<o:p></o:p></p>
<p> <o:p></o:p></p>
<p> <o:p></o:p></p>
<p> <o:p></o:p></p>
<p> <o:p></o:p></p>
<p> <o:p></o:p></p>
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