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    <title>topic is it sip or h323 between the in IP Telephony and Phones</title>
    <link>https://community.cisco.com/t5/ip-telephony-and-phones/sip-8831-disconnect-cause-47/m-p/2633468#M291663</link>
    <description>&lt;P&gt;is it sip or h323 between the CMEs ?&lt;/P&gt;</description>
    <pubDate>Tue, 17 Feb 2015 19:50:27 GMT</pubDate>
    <dc:creator>Jeff Levensailor</dc:creator>
    <dc:date>2015-02-17T19:50:27Z</dc:date>
    <item>
      <title>SIP 8831 disconnect cause 47</title>
      <link>https://community.cisco.com/t5/ip-telephony-and-phones/sip-8831-disconnect-cause-47/m-p/2633467#M291662</link>
      <description>&lt;P&gt;Dear All,&lt;/P&gt;&lt;P&gt;&amp;nbsp;&lt;/P&gt;&lt;P&gt;Currently i am deploying ip telephony system. In site a we have sccp and sip phone, in site b there is only sccp phone.&lt;/P&gt;&lt;P&gt;the topology is:&lt;/P&gt;&lt;P&gt;ip phone (sccp and sip) -- CME -- WAN -- CME -- ip phone (sccp)&lt;/P&gt;&lt;P&gt;when i call from sccp ip phone in site a to sccp ip phone in site b, the call works well.&amp;nbsp;but when i try to call from sip ip phone in site a to sccp ip phone in site b, ip phone in site b is ringing, but when picked up , the call is disconnected.&amp;nbsp;i try to run "debug ccsip message" i found disconnect cause 47.&lt;/P&gt;&lt;P&gt;&amp;nbsp;&lt;/P&gt;&lt;P&gt;after that, i try to simulate the call. i install new router (as cme) in site a (same site)&amp;nbsp;but in different network segment. i register 1 sccp ip phone to the new cme. then i call from sip ip phone to sccp phone. the call works well.&lt;/P&gt;&lt;P&gt;&amp;nbsp;&lt;/P&gt;&lt;P&gt;i ask them to check if there is blocking port for sip. but they said they won't open it (it's related to their policy). they insist us to create transcoding or mtp to resolve this case.&lt;/P&gt;&lt;P&gt;if i configure mtp, is that going to solve the problem?&lt;/P&gt;&lt;P&gt;&amp;nbsp;&lt;/P&gt;&lt;P&gt;Thanks,&lt;/P&gt;&lt;P&gt;Anju Josua&lt;/P&gt;</description>
      <pubDate>Sun, 17 Mar 2019 09:00:57 GMT</pubDate>
      <guid>https://community.cisco.com/t5/ip-telephony-and-phones/sip-8831-disconnect-cause-47/m-p/2633467#M291662</guid>
      <dc:creator>Anju Josua</dc:creator>
      <dc:date>2019-03-17T09:00:57Z</dc:date>
    </item>
    <item>
      <title>is it sip or h323 between the</title>
      <link>https://community.cisco.com/t5/ip-telephony-and-phones/sip-8831-disconnect-cause-47/m-p/2633468#M291663</link>
      <description>&lt;P&gt;is it sip or h323 between the CMEs ?&lt;/P&gt;</description>
      <pubDate>Tue, 17 Feb 2015 19:50:27 GMT</pubDate>
      <guid>https://community.cisco.com/t5/ip-telephony-and-phones/sip-8831-disconnect-cause-47/m-p/2633468#M291663</guid>
      <dc:creator>Jeff Levensailor</dc:creator>
      <dc:date>2015-02-17T19:50:27Z</dc:date>
    </item>
    <item>
      <title>Hi Jeff, i use h.323 between</title>
      <link>https://community.cisco.com/t5/ip-telephony-and-phones/sip-8831-disconnect-cause-47/m-p/2633469#M291664</link>
      <description>&lt;P&gt;Hi Jeff,&lt;/P&gt;&lt;P&gt;&amp;nbsp;&lt;/P&gt;&lt;P&gt;i use h.323 between site.&lt;/P&gt;&lt;P&gt;8831 register as sip to cme, but i use&amp;nbsp;h.323 dialpeer to other site.&lt;/P&gt;&lt;P&gt;&amp;nbsp;&lt;/P&gt;&lt;P&gt;Thanks,&lt;/P&gt;&lt;P&gt;Anju&lt;/P&gt;</description>
      <pubDate>Wed, 18 Feb 2015 08:34:21 GMT</pubDate>
      <guid>https://community.cisco.com/t5/ip-telephony-and-phones/sip-8831-disconnect-cause-47/m-p/2633469#M291664</guid>
      <dc:creator>Anju Josua</dc:creator>
      <dc:date>2015-02-18T08:34:21Z</dc:date>
    </item>
    <item>
      <title>can you paste your config?</title>
      <link>https://community.cisco.com/t5/ip-telephony-and-phones/sip-8831-disconnect-cause-47/m-p/2633470#M291665</link>
      <description>&lt;P&gt;can you paste your config? they definitely don't need to open any ports for sip if its an h323 trunk, i feel like you need a transcoder or mtp like they are saying.&lt;/P&gt;</description>
      <pubDate>Wed, 18 Feb 2015 16:27:43 GMT</pubDate>
      <guid>https://community.cisco.com/t5/ip-telephony-and-phones/sip-8831-disconnect-cause-47/m-p/2633470#M291665</guid>
      <dc:creator>Jeff Levensailor</dc:creator>
      <dc:date>2015-02-18T16:27:43Z</dc:date>
    </item>
    <item>
      <title>Dear Jeff, sori for late</title>
      <link>https://community.cisco.com/t5/ip-telephony-and-phones/sip-8831-disconnect-cause-47/m-p/2633471#M291666</link>
      <description>&lt;P&gt;Dear Jeff,&lt;/P&gt;&lt;P&gt;&amp;nbsp;&lt;/P&gt;&lt;P&gt;sori for late respon.&lt;/P&gt;&lt;P&gt;hereby i attach my show run.&lt;/P&gt;&lt;P&gt;&amp;nbsp;&lt;/P&gt;&lt;P&gt;Thanks,&lt;/P&gt;&lt;P&gt;Anju&lt;/P&gt;</description>
      <pubDate>Tue, 07 Apr 2015 09:59:34 GMT</pubDate>
      <guid>https://community.cisco.com/t5/ip-telephony-and-phones/sip-8831-disconnect-cause-47/m-p/2633471#M291666</guid>
      <dc:creator>Anju Josua</dc:creator>
      <dc:date>2015-04-07T09:59:34Z</dc:date>
    </item>
    <item>
      <title>Hi,Can you please share the</title>
      <link>https://community.cisco.com/t5/ip-telephony-and-phones/sip-8831-disconnect-cause-47/m-p/2633472#M291667</link>
      <description>&lt;P&gt;Hi,&lt;/P&gt;&lt;P&gt;Can you please share the debug ccsip messages?&lt;/P&gt;&lt;P&gt;If you want to use MTP, then change the codec to g729r8.&lt;/P&gt;&lt;P&gt;&amp;nbsp;&lt;/P&gt;&lt;P&gt;HTH&lt;/P&gt;</description>
      <pubDate>Tue, 07 Apr 2015 11:12:29 GMT</pubDate>
      <guid>https://community.cisco.com/t5/ip-telephony-and-phones/sip-8831-disconnect-cause-47/m-p/2633472#M291667</guid>
      <dc:creator>inderpreetsingh23</dc:creator>
      <dc:date>2015-04-07T11:12:29Z</dc:date>
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