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<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Sip reload<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>
<div>
<p class=MsoNormal><span lang=EN-US style='font-size:10.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Regards,<o:p></o:p></span></p>
<p class=MsoNormal><b><span lang=EN-US style='font-size:10.0pt;font-family:
"Calibri","sans-serif";color:#1F497D'>Mindaugas Kezys<o:p></o:p></span></b></p>
<p class=MsoNormal><span lang=EN-US style='font-size:10.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='font-size:10.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Kolmisoft UAB <o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='font-size:10.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>VoIP Billing Solutions<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='font-size:10.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>e-mail: <a href="mailto:info@kolmisoft.com"><span
style='color:blue'>info@kolmisoft.com</span></a><o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='font-size:10.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>URL: <a href="http://www.kolmisoft.com"><span style='color:blue'>http://www.kolmisoft.com</span></a><o:p></o:p></span></p>
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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>
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<p class=MsoNormal><b><span lang=EN-US style='font-size:10.0pt;font-family:
"Tahoma","sans-serif"'>From:</span></b><span lang=EN-US 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>jonas
kellens<br>
<b>Sent:</b> Tuesday, March 02, 2010 4:59 PM<br>
<b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br>
<b>Subject:</b> Re: [asterisk-users] rtcachefriends & qualify & sip
reload<o:p></o:p></span></p>
</div>
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<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>On Tue, 2010-03-02 at 11:32 +0000, Ishfaq Malik wrote: <o:p></o:p></p>
<pre><o:p> </o:p></pre><pre>If you are changing RealTime config in your DB you need to do a sip <o:p></o:p></pre><pre>prune realtime either directly from asterisk cli or using AMI. You <o:p></o:p></pre><pre>really do not need to do a SIP reload when changing the config of one <o:p></o:p></pre><pre>sip extension.<o:p></o:p></pre>
<p class=MsoNormal>I notice that after a "sip prune realtime all" I
also loose all of my realtime sip peers. Same result actually as with "sip
reload".<br>
<br>
I close the softphone of gerrie2 (becomes unspecified)<br>
<br>
<span style='font-size:10.0pt;color:blue'>asterisk*CLI> sip show peers</span><br>
<span style='font-size:10.0pt;color:blue'>Name/username
Host Dyn Nat
ACL Port Status Realtime </span><br>
<span style='font-size:10.0pt;color:blue'>gerrie005/gerrie005
192.168.1.106 D N
5060 OK (4 ms) Cached RT </span><br>
<span style='font-size:10.0pt;color:blue'>gerrie002/gerrie002
(Unspecified) D N
0 UNKNOWN Cached RT
</span><br>
<span style='font-size:10.0pt;color:blue'>gerrie001/gerrie001
192.168.1.105 D N
5060 OK (11 ms) Cached RT</span><br>
<br>
I prune the realtime peers to no longer have gerrie002 in cache :<br>
<br>
<span style='font-size:10.0pt;color:blue'>asterisk*CLI> sip prune realtime
all</span><br>
<span style='font-size:10.0pt;color:blue'>3 peers pruned.</span><br>
<span style='font-size:10.0pt;color:blue'>2 users pruned.</span><br>
<span style='font-size:10.0pt;color:blue'>[Mar 2 15:42:19] NOTICE[32498]:
chan_sip.c:16612 sip_poke_noanswer: Peer 'gerrie001' is now UNREACHABLE!
Last qualify: 91</span><br>
<br>
The realtime peers are all gone :<br>
<br>
<span style='font-size:10.0pt;color:blue'>asterisk*CLI> sip show peers</span><br>
<span style='font-size:10.0pt;color:blue'>Name/username
Host Dyn Nat
ACL Port Status Realtime </span><br>
<br>
Internal call fails :<br>
<br>
<span style='font-size:10.0pt;color:blue'>[Mar 2 15:46:38] WARNING[558]:
app_dial.c:1272 dial_exec_full: Unable to create channel of type 'SIP' (cause
20 - Unknown)</span><br>
<span style='font-size:10.0pt;color:blue'>[Mar 2 15:46:38] ==
Everyone is busy/congested at this time (1:0/0/1)</span><br>
<span style='font-size:10.0pt;color:blue'>[Mar 2 15:46:38] ==
Auto fallthrough, channel 'SIP/gerrie001-09f631e0' status is 'CHANUNAVAIL'</span><br>
<br>
I re-register 2 softphones (gerrie001 & gerrie005) :<br>
<br>
<span style='font-size:10.0pt;color:blue'>asterisk*CLI> sip show peers</span><br>
<span style='font-size:10.0pt;color:blue'>Name/username
Host Dyn Nat
ACL Port Status Realtime </span><br>
<span style='font-size:10.0pt;color:blue'>gerrie002/gerrie002
(Unspecified) D N
0 UNREACHABLE Cached RT </span><br>
<span style='font-size:10.0pt;color:blue'>gerrie001/gerrie001
192.168.1.105 D N
5060 OK (11 ms) Cached RT </span><br>
<span style='font-size:10.0pt;color:blue'>gerrie005/gerrie005
192.168.1.106 D N
5060 OK (7 ms) Cached RT </span><br>
<br>
The SIP-peer 'gerrie002' is still in the cache ! Don't know where this is
coming from ??<br>
<br>
I prune again :<br>
<br>
<span style='font-size:10.0pt;color:blue'>asterisk*CLI> sip prune realtime
all</span><br>
<span style='font-size:10.0pt;color:blue'>3 peers pruned.</span><br>
<span style='font-size:10.0pt;color:blue'>1 users pruned.</span><br>
<span style='font-size:10.0pt;color:blue'>[Mar 2 15:51:57] NOTICE[32498]:
chan_sip.c:16612 sip_poke_noanswer: Peer 'gerrie001' is now UNREACHABLE!
Last qualify: 11</span><br>
<span style='font-size:10.0pt;color:blue'>[Mar 2 15:51:57] NOTICE[32498]:
chan_sip.c:16612 sip_poke_noanswer: Peer 'gerrie001' is now UNREACHABLE!
Last qualify: 11</span><br>
<span style='font-size:10.0pt;color:blue'>[Mar 2 15:52:01] NOTICE[32498]:
chan_sip.c:16612 sip_poke_noanswer: Peer 'gerrie001' is now UNREACHABLE!
Last qualify: 11</span><br>
<br>
And again no more peers until I re-register :<br>
<br>
<span style='font-size:10.0pt;color:blue'>asterisk*CLI> sip show peers</span><br>
<span style='font-size:10.0pt;color:blue'>Name/username
Host Dyn Nat
ACL Port Status Realtime </span><br>
<br>
<br>
This realtime thing isn't really working out here... What exactly do I need to
do to clear the cache and thus the old SIP-peers so they can no longer be used
??<br>
<br>
Jonas. <o:p></o:p></p>
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