Update.  I am able to make outbound calls the new SIP Trunk.  It is using DP 200.   Inbound call to test number rings my phone but when I pick up there is no connection and my cell still shows calling. combing thru the debugs I found that the SIP/2.0 Trying and Ringing are now showing the right IP address (209.x.x.x) but it is now the received portion which show the 10.1.x.x ip address which is the carriers internal IP's.   Is this still an issue on my end with the DP's or should I look to the carrier? Sent: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 209.142.200.14:5060;branch=z9hG4bKdmrim7309otm0lighlo0.1 From: "" <sip:6162831487@209.142.200.14:5060>;tag=5679373 To: <sip:6162775094@209.142.200.14:5060>;tag=283044B8-1A57 Date: Tue, 12 Feb 2019 18:24:15 GMT Call-ID: 1548058228-15951135@SFLDMIUP-C3SIPGW CSeq: 1 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event P-Asserted-Identity: "" <sip:7771350@192.77.235.171> Contact: <sip:6162775094@192.77.235.171:5060> Server: Cisco-SIPGateway/IOS-16.6.4 Session-ID: 0000353e00105000a00080ce623bf878;remote=713a4e8f5fbc5c12a670ee27070804c2 Content-Length: 0       234795: Feb 12 2019 18:24:16.420 UTC: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: INVITE sip:6162775094@192.77.235.171:5060 SIP/2.0 Via: SIP/2.0/UDP 209.142.200.14:5060;branch=z9hG4bKdmrim7309otm0lighlo0.1 To: <sip:6162775094@10.1.6.47:5060> From: "" <sip:6162831487@10.1.6.4>;tag=5679373 Call-ID: 1548058228-15951135@SFLDMIUP-C3SIPGW CSeq: 1 INVITE Max-Forwards: 69 Contact: <sip:6162831487@209.142.200.14:5060;transport=udp> Supported: 100rel Expires: 330 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, PRACK, REFER, SUBSCRIBE, NOTIFY, UPDATE, REGISTER Accept: application/sdp Content-Type: application/sdp Content-Length: 259   v=0 o=- 272265216 1549995856 IN IP4 209.142.200.14 ... View more

Sorry about that.. I was making a few changes last week with DP 100.  I set it back to where it was originally.  Attached is current running config.  The call should be using DP 100 to send the call out to the carrier correct?   ... View more

I believe your right about SIP profiles working.  I made the change but am now getting 408 Request Timeout.  Attached is new debug output.  It looks like when using DP 100 to send the call out is causing the hang up.  this is how DP100 is current configured.  I have it bound to gig 1/0/1 which is the WAN interface.   dial-peer voice 100 voip preference 4 session protocol sipv2 session target sip-server incoming called-number 8T voice-class codec 1 voice-class sip profiles 1 inbound voice-class sip bind control source-interface GigabitEthernet0/0/1 voice-class sip bind media source-interface GigabitEthernet0/0/1 dtmf-relay rtp-nte sip-notify fax-relay ecm disable fax rate 14400 no vad   ... View more

see attached.  thanks. ... View more

Ok, Using DP 200 now and not ringing my phone but instead getting the SIP/2.0 404 Not Found error in the log.   I am fairly sure the CSS on the SIP trunk has access to the extension I am translating the DID to in CUCM.  Still showing the 10.1.6.x IP addresses in the TO: and FROM: fields.   ... View more

Good Call!  Being that DP 200 is configured as below, I makes sense why it would not be up.  I need to configure either an answer-address, incoming called number or destination pattern to the DP to go to an up status.  Being that DP 200 is used to accept the call from the Provider as you stated, I should add the incoming called-number to capture everything.  Correct?   dial-peer voice 200 voip description Incoming dial-peer voice-class sip profiles 201 voice-class sip profiles 200 inbound ... View more

I am expecting one of the 20x dial peers to get matched as well.  So the incoming call should hit dial peer 100 first which connects the call to CUCM right?  But there should be a second leg using a 20x dial peer back to the sip carrier right?  I am not seeing that.  I attached the output show only the debug voip ccapi inout   ... View more

Thanks!  I was just looking at a similar solution reading thru some documentation. I followed your steps but no change.  After testing I show it exclusively uses dial-peer 100 only.  No reference to dial-peer 200.  I also noticed that invite includes the 10.1.6.xx ip address.   Yes - it is always the .14 ip.    Was I supposed to include something further here after the from and to?  voice class sip-copylist 200 sip-header from sip-header to      ... View more

Sorry, I was troubleshooting prior and shut that dial-peer off. It was enabled during the captures I provided.  I was just on the call with the carrier and I questioned this piece of the debug output below (specifically the part underlined and bolded.  The 10.1.6.x is the carrier internal network. I should be sending the destination of 209.142.200.xxxx.  From what I gather, I am retaining their internal IP's and not sending the 209.142.200.xxxx in the FROM and TO.  I am unsure how to remedy that.     SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.20.1.98:5060;branch=z9hG4bKD32204 From: "Putnam Troy" <sip:6162831477@209.142.200.14>;tag=DBD39BC-235F To: <sip:6162775094@172.20.1.104> Date: Thu, 07 Feb 2019 15:08:28 GMT Call-ID: F3A2E8B-2A2111E9-AEAAD21B-BC9A1796@172.20.1.98 CSeq: 101 INVITE Allow-Events: presence Content-Length: 0     103409: Feb  7 2019 15:08:28.984 UTC: //11514/0F3832BDAEA4/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.20.1.98:5060;branch=z9hG4bKD32204 From: "Putnam Troy" <sip:6162831477@209.142.200.14>;tag=DBD39BC-235F To: <sip:6162775094@172.20.1.104>;tag=32856804~9e135f52-c30b-4970-9ce6-aa055143ce9a-59865179 Date: Thu, 07 Feb 2019 15:08:28 GMT Call-ID: F3A2E8B-2A2111E9-AEAAD21B-BC9A1796@172.20.1.98 CSeq: 101 INVITE Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY Allow-Events: presence Server: Cisco-CUCM11.5 Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=DESKTOP Supported: X-cisco-srtp-fallback Supported: Geolocation Session-ID: 00002fed00105000a00080ce623bf878;remote=b6f9ac70b0d253ea9657216eb0d80351 P-Asserted-Identity: "Troy Putnam" <sip:7771350@172.20.1.104> Remote-Party-ID: "Troy Putnam" <sip:7771350@172.20.1.104>;party=called;screen=yes;privacy=off Contact: <sip:6162775094@172.20.1.104:5060>;+u.sip!devicename.ccm.cisco.com="CSFTPutnam";video;bfcp   Content-Length: 0     103410: Feb  7 2019 15:08:28.985 UTC: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:    Calling Number=, Called Number=6162775094, Peer Info Type=DIALPEER_INFO_SPEECH 103411: Feb  7 2019 15:08:28.985 UTC: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:    Match Rule=DP_MATCH_DEST; Called Number=6162775094 103412: Feb  7 2019 15:08:28.985 UTC: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:    Result=Success(0) after DP_MATCH_DEST 103413: Feb  7 2019 15:08:28.985 UTC: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:    dialstring=NULL, saf_enabled=0, saf_dndb_lookup=0, dp_result=0 103414: Feb  7 2019 15:08:28.985 UTC: //-1/xxxxxxxxxxxx/DPM/dpMatchPeers:    Result=SUCCESS(0)    List of Matched Outgoing Dial-peer(s):      1: Dial-peer Tag=100 103415: Feb  7 2019 15:08:28.985 UTC: //-1/xxxxxxxxxxxx/SIP/Error/ccsip_reg_search_reg_number_table: No entry found in reg Number Table for 6162831477 103416: Feb  7 2019 15:08:28.985 UTC: //-1/xxxxxxxxxxxx/SIP/Error/ccsip_reg_search_reg_number_table: No entry found in reg Number Table for 6162775094 103417: Feb  7 2019 15:08:28.985 UTC: //-1/xxxxxxxxxxxx/SIP/Error/sipSPI_validate_own_ip_addr: ReqLine IP addr does not match with host IP addr 103418: Feb  7 2019 15:08:28.985 UTC: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:    Calling Number=, Called Number=6162831477, Peer Info Type=DIALPEER_INFO_SPEECH 103419: Feb  7 2019 15:08:28.986 UTC: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:    Match Rule=DP_MATCH_DEST; Called Number=6162831477 103420: Feb  7 2019 15:08:28.986 UTC: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:    Result=Success(0) after DP_MATCH_DEST 103421: Feb  7 2019 15:08:28.986 UTC: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:    dialstring=NULL, saf_enabled=0, saf_dndb_lookup=0, dp_result=0 103422: Feb  7 2019 15:08:28.986 UTC: //-1/xxxxxxxxxxxx/DPM/dpMatchPeers:    Result=SUCCESS(0)    List of Matched Outgoing Dial-peer(s):      1: Dial-peer Tag=100 103423: Feb  7 2019 15:08:28.986 UTC: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:    Calling Number=7771350, Called Number=, Voice-Interface=0x0,    Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,    Peer Info Type=DIALPEER_INFO_SPEECH 103424: Feb  7 2019 15:08:28.986 UTC: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:    Result=NO_MATCH(-1) After All Match Rules Attempt 103425: Feb  7 2019 15:08:28.986 UTC: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:    dialstring=NULL, saf_enabled=0, saf_dndb_lookup=0, dp_result=-1 103426: Feb  7 2019 15:08:28.987 UTC: //11513/0F3832BDAEA4/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 209.142.200.14:5060;branch=z9hG4bKmtpboj20d82rk2hsej70.1 From: "Putnam Troy" <sip:6162831477@10.1.6.4>;tag=4192693 To: <sip:6162775094@10.1.6.47:5060>;tag=DBD39F7-EF6 Date: Thu, 07 Feb 2019 15:08:28 GMT Call-ID: 1548058228-14743757@SFLDMIUP-C3SIPGW CSeq: 1 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event P-Asserted-Identity: "Troy Putnam" <sip:7771350@192.77.235.171> Contact: <sip:6162775094@192.77.235.171:5060> Server: Cisco-SIPGateway/IOS-16.6.4 Session-ID: 00002fed00105000a00080ce623bf878 ... View more

Thanks for the responses.  We are moving from a dynamic sip solution to a static so I  have not defined a SIP profile (request REGISTER, INVITE, etc) thinking I did not need to.  Maybe I do. The  192.77.235.171 is as you stated, our public IP address or Source IP>  The destination IP (SIP provider) is 209.142.200.14.   ... View more

See attached ... View more

Thank you for the response.  Please see new debug file with debug ccsip message enabled. ... View more

slightly but I am not using schedules.  Well, the scheduled used for every location via the sites user template is set to ALL HOURS.   Since I can't predict when a school is going to close due to bad weather, I don't how using a closed greeting for this is possible.  I may looking at this the wrong way however.   ... View more

Thanks for the reply.  We have over 80 schools so I would have to administrate the closed schedule to accommodate when schools close.. which I don't really want to do  :).   Because the alternate greeting takes precedence over other greetings, the person creating the greeting can still enable with an end date.   Same as the closed with the exception the closed is dependent on a schedule.   Sound right? ... View more