that becasue the reinvite is using a private ip probably..<br><br>sip debug <br><br>pastebin the results..<br><br>look in the re-invite part..<br><br><br><br><div><span class="gmail_quote">On 6/4/07, <b class="gmail_sendername">
Compnet Bobby</b> <<a href="mailto:compnetbobby@hotmail.com">compnetbobby@hotmail.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div link="blue" vlink="purple" lang="EN-US">
<div>
<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. </p>
<p> </p>
<p>I want your input on 2 problems, they are the following:</p>
<p> </p>
<p>1.</p>
<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). </p>
<p> </p>
<p> </p>
<p> -- Executing [9097406868@from-sip:1]
Answer("SIP/9097406868-09e110f8", "") in new stack</p>
<p> -- Executing [9097406868@from-sip:2]
BackGround("SIP/9097406868-09e110f8",
"menus/welcome-to-exec") in new stack</p>
<p> -- <SIP/9097406868-09e110f8>
Playing 'menus/welcome-to-exec' (language 'en')</p>
<p> == CDR updated on SIP/9097406868-09e110f8</p>
<p> -- Executing [103@from-sip:1]
Dial("SIP/9097406868-09e110f8", "SIP/103|50|m") in new
stack</p>
<p> -- Called 103</p>
<p> -- Started music on hold, class
'default', on SIP/9097406868-09e110f8</p>
<p> -- SIP/103-09dedd68 is ringing</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" onclick="return top.js.OpenExtLink(window,event,this)">LAXMGC0120070529230718052251@209.244.63.13
</a>
for seqno 1 (Critical Response)</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" onclick="return top.js.OpenExtLink(window,event,this)">LAXMGC0120070529230718052251@209.244.63.13</a> - no
reply to our critical packet.</p>
<p> -- Stopped music on hold on
SIP/9097406868-09e110f8</p>
<p> == Spawn extension (from-sip, 103, 1) exited
non-zero on 'SIP/9097406868-09e110f8'</p>
<p> </p>
<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. </p>
<p> </p>
<p> </p>
<p>Thanks for the help in advance!!!</p>
<p> </p>
<p> </p>
<p> </p>
<p> </p>
<p> </p>
</div>
</div>
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</a><br><br></blockquote></div><br><br clear="all"><br>-- <br>Mike<br>Sales Manager<br><a href="http://www.voicemeup.com">http://www.voicemeup.com</a><br>Making it happen<br>1.877.807.VOIP (8647)<br>1.514.312.7030