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</o:shapelayout></xml><![endif]--></head><body lang=ES-AR link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Good day Jay,<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 lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>For what I´m seeing on the debugs on the inbound call the invite is sending an empty called number:<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Received:<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>INVITE sip:5856786019@66.66.214.216:60954 SIP/2.0<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK4ab70570;rport<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>From: "+15856944923" ;tag=as2ccf6d86<o:p></o:p></span></p><p class=MsoNormal><b><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>To:<o:p></o:p></span></b></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Contact:<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>That´s why you are sending:<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Sent:<o:p></o:p></span></p><p class=MsoNormal><b><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>SIP/2.0 400 Bad Request - 'Invalid Host'<o:p></o:p></span></b></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK4ab70570;rport<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>From: "+15856944923" ;tag=as2ccf6d86<o:p></o:p></span></p><p class=MsoNormal><b><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>To: ;tag=1A0D2FC-D89<o:p></o:p></span></b></p><p class=MsoNormal><b><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></b></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Is that coming from the provider? Check that out with them…<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>I don’t have good experiences with NAT and anything related to voice, but with the NAT configuration you´ve got, packets are at least getting to the CME… You could have one way audio issues after solving the empty called number problem. In that case you might need to configure inspection to solve that problem, but I´m not sure if that would work with PAT. Check this doc (it´s for an ASA): <a href="http://www.cisco.com/en/US/docs/security/asa/asa82/configuration/guide/inspect_voicevideo.html#wp1204403">http://www.cisco.com/en/US/docs/security/asa/asa82/configuration/guide/inspect_voicevideo.html#wp1204403</a><o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Regards.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> <o:p></o:p></span></p><div><div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm 0cm 0cm'><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"'> cisco-voip-bounces@puck.nether.net [mailto:cisco-voip-bounces@puck.nether.net] <b>On Behalf Of </b>Jay Stants<br><b>Sent:</b> martes, 22 de marzo de 2011 11:53 a.m.<br><b>To:</b> cisco-voip@puck.nether.net<br><b>Subject:</b> Re: [cisco-voip] Please Help - Inbound dialing rings busy /Outbound works<o:p></o:p></span></p></div></div><p class=MsoNormal><o:p> </o:p></p><div><p>Good day everyone,<o:p></o:p></p><p> <o:p></o:p></p><p>Still trying to figure this issue out. here is a summary of posts .. Some say that it's possibly an issue of Nat and Sip not liking each other, so i will try to be as detailed as possible in hopes that someone can see something i'm possibly missing. <o:p></o:p></p><p> <o:p></o:p></p><p>Attached is the lab design for the environment - I have both EIGRP and OSPF running and i redistribute both at the core switch - all routes are reachable<o:p></o:p></p><p> <o:p></o:p></p><p><strong><u>WAN router is a 2611XM with the following ACL's Defined</u></strong><o:p></o:p></p><p> <o:p></o:p></p><div><div><p class=MsoNormal><code><span style='font-size:10.0pt'>ip nat inside source list NAT_ACL interface FastEthernet0/1 overload </span></code><o:p></o:p></p></div><div><p class=MsoNormal><code><span style='font-size:10.0pt'>! </span></code><o:p></o:p></p></div><div><p class=MsoNormal><code><span style='font-size:10.0pt'>ip access-list standard NAT_ACL </span></code><o:p></o:p></p></div><div><p class=MsoNormal><code><span style='font-size:10.0pt'> permit 10.0.0.0 0.255.255.255 </span></code><o:p></o:p></p></div><div><p class=MsoNormal><code><span style='font-size:10.0pt'>! </span></code><o:p></o:p></p></div><div><p class=MsoNormal><code><span style='font-size:10.0pt'>ip access-list extended FW_inbound </span></code><o:p></o:p></p></div><div><p class=MsoNormal><code><span style='font-size:10.0pt'> permit tcp any any established </span></code><o:p></o:p></p></div><div><p class=MsoNormal><code><span style='font-size:10.0pt'> permit udp any any </span></code><o:p></o:p></p></div><div><p class=MsoNormal><code><span style='font-size:10.0pt'> permit icmp any any echo-reply </span></code><o:p></o:p></p></div><div><p class=MsoNormal><code><span style='font-size:10.0pt'> deny ip any any </span></code><o:p></o:p></p></div><div><p class=MsoNormal><code><span style='font-size:10.0pt'>! </span></code><o:p></o:p></p></div></div><p> <o:p></o:p></p><p>CCME is configured on a 2811 router with a 16 port PoE module (NME-16ES-1G-P) - recently upgraded to IOS version AdvEnterpriseK9-M 15.1(2)GC / CCME 8.1<o:p></o:p></p><p>Configs are attached for review.<o:p></o:p></p><p> <o:p></o:p></p><p>snet-voip1#sh sip-ua register status<br>Line peer expires(sec) registered P-Associ-URI<br>================================ ========== ============ ========== ============<br>122812_cme -1 82 yes<br>5856786019 20004 101 no<o:p></o:p></p><p> <o:p></o:p></p><p>Debugs that i have run indicating that inbound there is an issue either from voip.ms to me or else on the CCME device with routing the incoming call to a phone. I'm not 100% positive where the issue lies. but dialing my DID just rings busy almost immediatly.(not a fast busy)<o:p></o:p></p><p> <o:p></o:p></p><p><strong><u>debug voip ccapi inout and debug ccsip messages</u></strong><o:p></o:p></p><p>Mar 21 03:18:49.669: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:<br>Sent:<br>REGISTER <a href="sip:newyork.voip.ms:5060">sip:newyork.voip.ms:5060</a> SIP/2.0<br>Via: SIP/2.0/UDP 10.1.10.2:5060;branch=z9hG4bK340EF7<br>From: ;tag=1A0CE20-114C<br>To:<br>Date: Mon, 21 Mar 2011 03:18:49 GMT<br>Call-ID: D60F7CDF-526111E0-8002D2DF-336AE900<br>User-Agent: Cisco-SIPGateway/IOS-12.x<br>Max-Forwards: 70<br>Timestamp: 1300677529<br>CSeq: 380 REGISTER<br>Contact:<br>Expires: 180<br>Supported: path<br>Content-Length: 0<br><br>Mar 21 03:18:49.721: //417/000000000000/SIP/Msg/ccsipDisplayMsg:<br>Received:<br>SIP/2.0 100 Trying<br>Via: SIP/2.0/UDP 10.1.10.2:5060;branch=z9hG4bK340EF7;received=10.1.10.2<br>From: ;tag=1A0CE20-114C<br>To:<br>Call-ID: D60F7CDF-526111E0-8002D2DF-336AE900<br>CSeq: 380 REGISTER<br>User-Agent: VoIPMS/SERAST<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Su<br>snet-voip1#pported: replaces<br>Contact:<br>Content-Length: 0<br><br>Mar 21 03:18:49.725: //417/000000000000/SIP/Msg/ccsipDisplayMsg:<br>Received:<br>SIP/2.0 401 Unauthorized<br>Via: SIP/2.0/UDP 10.1.10.2:5060;branch=z9hG4bK340EF7;received=10.1.10.2<br>From: ;tag=1A0CE20-114C<br>To: ;tag=as78e4178e<br>Call-ID: D60F7CDF-526111E0-8002D2DF-336AE900<br>CSeq: 380 REGISTER<br>User-Agent: VoIPMS/SERAST<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Supported: replaces<br>WWW-Authenticate: Digest algorithm=MD5, realm="newyork.voip.ms", nonce="08644cbc"<br>Content-Length: 0<br><br>Mar 21 03:18:49.729: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:<br>Sent:<br>REGISTER <a href="sip:newyork.voip.ms:5060">sip:newyork.voip.ms:5060</a> SIP/2.0<br>Via: SIP/2.0/UDP 10.1.10.2:5060;branch=z9hG4bK3411244<br>From: ;tag=1A0CE20-114C<br>To:<br>Date: Mon, 21 Mar 2011 03:18:49 GMT<br>Call-ID: D60F7CDF-526111E0-8002D2DF-336AE900<br>User-Agent: Cisco-SIPGateway/IOS-12.x<br>Max-Forwards: 70<br>Timestamp: 1300677529<br>CSeq: 381 REGISTER<br>Contact:<br>Expires: 180<br>Authorization: Digest username="122812_cme",realm="newyork.voip.ms",uri="<a href="sip:newyork.voip.ms:5060">sip:newyork.voip.ms:5060</a>",response="81e0f1e1d74178c268f1aec698535939",nonce="08644cbc",algorithm=MD5<br>Content-Length: 0<br><br>Mar 21 03:18:49.781: //417/000000000000/SIP/Msg/ccsipDisplayMsg:<br>Received:<br>SIP/2.0 100 Trying<br>Via: SIP/2.0/UDP 10.1.10.2:5060;branch=z9hG4bK3411244;received=10.1.10.2<br>From: ;tag=1A0CE20-114C<br>To:<br>Call-ID: D60F7CDF-526111E0-8002D2DF-336AE900<br>CSeq: 381 REGISTER<br>User-Agent: VoIPMS/SERAST<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Supported: replaces<br>Contact:<br>Content-Length: 0<br><br>Mar 21 03:18:49.797: //417/000000000000/SIP/Msg/ccsipDisplayMsg:<br>Received:<br>SIP/2.0 200 OK<br>Via: SIP/2.0/UDP 10.1.10.2:5060;branch=z9hG4bK3411244;received=10.1.10.2<br>From: ;tag=1A0CE20-114C<br>To: ;tag=as78e4178e<br>Call-ID: D60F7CDF-526111E0-8002D2DF-336AE900<br>CSeq: 381 REGISTER<br>User-Agent: VoIPMS/SERAST<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Supported: replaces<br>Expires: 180<br>Contact: ;expires=180<br>Date: Mon, 21 Mar 2011 03:18:49 GMT<br>Content-Length: 0<br><br>Mar 21 03:18:50.905: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:<br>Received:<br>INVITE <a href="sip:5856786019@66.66.214.216:60954">sip:5856786019@66.66.214.216:60954</a> SIP/2.0<br>Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK4ab70570;rport<br>From: "+15856944923" ;tag=as2ccf6d86<br>To:<br>Contact:<br>Call-ID: <a href="mailto:73f32c6f3b912602300b0c1814fc7b28@74.63.41.218">73f32c6f3b912602300b0c1814fc7b28@74.63.41.218</a><br>CSeq: 102 INVITE<br>User-Agent: VoIPMS/SERAST<br>Max-Forwards: 70<br>Remote-Party-ID: "+15856944923" ;privacy=off;screen=no<br>Date: Mon, 21 Mar 2011 03:18:50 GMT<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Supported: replaces<br>Content-Type: application/sdp<br>Content-Length: 285<br><br>v=0<br>o=root 2881 2881 IN IP4 74.63.41.218<br>s=session<br>c=IN IP4 74.63.41.218<br>t=0 0<br>m=audio 19002 RTP/AVP 0 18 101<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:18 G729/8000<br>a=fmtp:18 annexb=no<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=silenceSupp:off - - - -<br>a=ptime:20<br>a=sendrecv<br><br>Mar 21 03:18:50.909: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:<br>Sent:<br>SIP/2.0 400 Bad Request - 'Invalid Host'<br>Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK4ab70570;rport<br>From: "+15856944923" ;tag=as2ccf6d86<br>To: ;tag=1A0D2FC-D89<br>Date: Mon, 21 Mar 2011 03:18:50 GMT<br>Call-ID: <a href="mailto:73f32c6f3b912602300b0c1814fc7b28@74.63.41.218">73f32c6f3b912602300b0c1814fc7b28@74.63.41.218</a><br>CSeq: 102 INVITE<br>Allow-Events: telephone-event<br>Reason: Q.850;cause=100<br>Server: Cisco-SIPGateway/IOS-12.x<br>Content-Length: 0<br><br>Mar 21 03:18:50.961: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:<br>Received:<br>ACK <a href="sip:5856786019@66.66.214.216:60954">sip:5856786019@66.66.214.216:60954</a> SIP/2.0<br>Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK4ab70570;rport<br>From: "+15856944923" ;tag=as2ccf6d86<br>To: ;tag=1A0D2FC-D89<br>Contact:<br>Call-ID: <a href="mailto:73f32c6f3b912602300b0c1814fc7b28@74.63.41.218">73f32c6f3b912602300b0c1814fc7b28@74.63.41.218</a><br>CSeq: 102 ACK<br>User-Agent: VoIPMS/SERAST<br>Max-Forwards: 70<br>Remote-Party-ID: "+15856944923" ;privacy=off;screen=no<br>Content-Length: 0<br><br>Mar 21 03:19:07.973: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:<br>Sent:<br>REGISTER <a href="sip:newyork.voip.ms:5060">sip:newyork.voip.ms:5060</a> SIP/2.0<br>Via: SIP/2.0/UDP 10.1.10.2:5060;branch=z9hG4bK34220CC<br>From: ;tag=1A115A4-EB0<br>To:<br>Date: Mon, 21 Mar 2011 03:19:07 GMT<br>Call-ID: D62E9D9A-526111E0-8003D2DF-336AE900<br>User-Agent: Cisco-SIPGateway/IOS-12.x<br>Max-Forwards: 70<br>Timestamp: 1300677547<br>CSeq: 456 REGISTER<br>Contact:<br>Expires: 180<br>Supported: path<br>Content-Length: 0<br><br>Mar 21 03:19:08.029: //418/000000000000/SIP/Msg/ccsipDisplayMsg:<br>Received:<br>SIP/2.0 401 Unauthorized<br>Via: SIP/2.0/UDP 10.1.10.2:5060;branch=z9hG4bK34220CC;received=10.1.10.2<br>From: ;tag=1A115A4-EB0<br>To: ;tag=as47fbc031<br>Call-ID: D62E9D9A-526111E0-8003D2DF-336AE900<br>CSeq: 456 REGISTER<br>User-Agent: VoIPMS/SERAST<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, S<br>snet-voip1#UBSCRIBE, NOTIFY<br>Supported: replaces<br>WWW-Authenticate: Digest algorithm=MD5, realm="newyork.voip.ms", nonce="74aab3a6"<br>Content-Length: 0<br><br>Mar 21 03:19:08.033: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:<br>Sent:<br>REGISTER <a href="sip:newyork.voip.ms:5060">sip:newyork.voip.ms:5060</a> SIP/2.0<br>Via: SIP/2.0/UDP 10.1.10.2:5060;branch=z9hG4bK3438DE<br>From: ;tag=1A115A4-EB0<br>To:<br>Date: Mon, 21 Mar 2011 03:19:08 GMT<br>Call-ID: D62E9D9A-526111E0-8003D2DF-336AE900<br>User-Agent: Cisco-SIPGateway/IOS-12.x<br>Max-Forwards: 70<br>Timestamp: 1300677548<br>CSeq: 457 REGISTER<br>Contact:<br>Expires: 180<br>Authorization: Digest username="122812_cme",realm="newyork.voip.ms",uri="<a href="sip:newyork.voip.ms:5060">sip:newyork.voip.ms:5060</a>",response="6ecb543fe1f1bcb70940d5c129e062d4",nonce="74aab3a6",algorithm=MD5<br>Content-Length: 0<br><br>Mar 21 03:19:08.085: //418/000000000000/SIP/Msg/ccsipDisplayMsg:<br>Received:<br>SIP/2.0 401 Unauthorized<br>Via: SIP/2.0/UDP 10.1.10.2:5060;branch=z9hG4bK3438DE;received=10.1.10.2<br>From: ;tag=1A115A4-EB0<br>To: ;tag=as47fbc031<br>Call-ID: D62E9D9A-526111E0-8003D2DF-336AE900<br>CSeq: 457 REGISTER<br>User-Agent: VoIPMS/SERAST<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Supported: replaces<br>WWW-Authenticate: Digest algorithm=MD5, realm="newyork.voip.ms", nonce="6de8b321"<br>Content-Length: 0<br><br>Mar 21 03:19:09.029: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:<br>Received:<br>SIP/2.0 401 Unauthorized<br>Via: SIP/2.0/UDP 10.1.10.2:5060;branch=z9hG4bK34220CC;received=10.1.10.2<br>From: ;tag=1A115A4-EB0<br>To: ;tag=as47fbc031<br>Call-ID: D62E9D9A-526111E0-8003D2DF-336AE900<br>CSeq: 456 REGISTER<br>User-Agent: VoIPMS/SERAST<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Supported: replaces<br>WWW-Authenticate: Digest algorithm=MD5, realm="newyork.voip.ms", nonce="74aab3a6"<br>Content-Length: 0<br><br>Mar 21 03:19:09.085: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:<br>Received:<br>SIP/2.0 401 Unauthorized<br>Via: SIP/2.0/UDP 10.1.10.2:5060;branch=z9hG4bK3438DE;received=10.1.10.2<br>From: ;tag=1A115A4-EB0<br>To: ;tag=as47fbc031<br>Call-ID: D62E9D9A-526111E0-8003D2DF-336AE900<br>CSeq: 457 REGISTER<br>User-Agent: VoIPMS/SERAST<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Supported: replaces<br>WWW-Authenticate: Digest algorithm=MD5, realm="newyork.voip.ms", nonce="6de8b321"<br>Content-Length: 0<o:p></o:p></p><p> <o:p></o:p></p><p><strong><u>debug voip ccapi error and debug ccsip error</u></strong><br><br>Mar 21 02:00:37.434: //-1/xxxxxxxxxxxx/SIP/Error/sipSPI_validate_own_ip_addr: ReqLine IP addr does not match with host IP addr<br>Mar 21 02:00:37.434: //-1/D988D9A98006/SIP/Error/sact_idle_new_message_invite: Invalid URL in incoming INVITE<br>Mar 21 02:00:37.438: //-1/xxxxxxxxxxxx/CCAPI/cc_set_post_tagdata:<br>CALL_ERROR; Avlist Set Is Failed<br>snet-voip1#<br>Mar 21 02:01:00.734: //-1/xxxxxxxxxxxx/SIP/Error/debugPrintBranchList: via branch list is:<br>Mar 21 02:01:00.734: //-1/xxxxxxxxxxxx/SIP/Error/debugPrintBranchList: end of list<br>Mar 21 02:01:00.790: //-1/xxxxxxxxxxxx/SIP/Error/debugPrintBranchList: via branch list is:<br>Mar 21 02:01:00.790: //-1/xxxxxxxxxxxx/SIP/Error/debugPrintBranchList: end of list<br>Mar 21 02:01:00.790: //346/000000000000/SIP/Error/ccsip_api_register_result_ind: Message Code Class 4xx Method Code 100 received for REGISTER<br>Mar 21 02:01:01.734: //-1/xxxxxxxxxxxx/SIP/Error/sipSPILocateInviteDialogCCB: Could not find ccb for response<br>snet-voip1#<br>Mar 21 02:01:01.790: //-1/xxxxxxxxxxxx/SIP/Error/sipSPILocateInviteDialogCCB: Could not find ccb for response<o:p></o:p></p><p> <o:p></o:p></p><p>If more detail is needed please let me know and i can provide additional debugs. I've done an Echo test on the sip provider side to the DID and that was successful and the provider sees the call delivered so i believe that to validate the config within the account on voip.ms side to be correct as far as recieving calls from PSTN and delivering to DID / SIP side. <o:p></o:p></p><p> <o:p></o:p></p><p>Any help is much appreciated<o:p></o:p></p><p>Regards,<o:p></o:p></p><p> <o:p></o:p></p><p>Jay Stants<o:p></o:p></p><p> <o:p></o:p></p></div></div></body></html>