I've got a problem with Diversion headers.  our DN's are configured as +E.164.  When forwarding calls we send out the Diversion header with the number of the phone.  This should work but the number that gets populated is +1XXXXXXXXXX. And the carrier is not accepting that.  It will accept it without the +1.  So I am trying to use SIP profiles to remove the +1 from the Diversion header but it is not working.   Here is the invite:   5047483: Oct 18 11:48:32.649 MDT: //4713576/06089D800000/SIP/Msg/ccsipDisplayMsg: Sent:   INVITE sip:8018197960@165.117.119.158 SIP/2.0 Via: SIP/2.0/UDP 10.225.255.249:5060;branch=z9hG4bK4521622390 From: "WIRELESS CALLER" <sip:4802001918@10.225.255.249>;tag=E2FD44C4-E3E To: <sip:8018197960@165.117.119.158> Date: Thu, 18 Oct 2018 17:48:32 GMT Call-ID: DD39E44A-D23411E8-A37FA40F-83E7177C@10.225.255.249 Supported: 100rel,timer,resource-priority,replaces,sdp-anat Min-SE:   1800 Cisco-Guid: 0101227904-0000065536-0000031816-0184869130 User-Agent: Cisco-SIPGateway/IOS-15.4.3.M5 Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 101 INVITE Timestamp: 1539884912 Contact: <sip:4802001918@10.225.255.249:5060> Expires: 180 Allow-Events: telephone-event Max-Forwards: 66 P-Preferred-Identity: "WIRELESS CALLER" <sip:4802001918@10.225.255.249> Diversion: "Kaye Parrish"<sip:+18015634115@10.225.255.249>;privacy=off;reason=unconditional;screen=yes Session-Expires:   1800 Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 349   v=0 o=CiscoSystemsSIP-GW-UserAgent 7233 8884 IN IP4 10.225.255.249 s=SIP Call c=IN IP4 10.225.255.249 t=0 0 m=audio 16574 RTP/AVP 0 18 100 101 c=IN IP4 10.225.255.249 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:100 X-NSE/8000 a=fmtp:100 192-194 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20   Note that the Diversion header is there and it is: Diversion: "Kaye Parrish"<sip:+18015634115@10.225.255.249>;privacy=off;reason=unconditional;screen=yes   Here is my SIP profile voice class sip-profiles 1001   request INVITE sip-header Diversion modify "<sip:+1801563(.*)@" "<sip:801563\1@"   and my Dial-peer dial-peer voice 1001 voip   description Outbound calls to SIP-801-563-Midvale-UT-01   translation-profile outgoing ToSIPPSTN   destination-pattern 801563*T   session protocol sipv2   session server-group 20   voice-class codec 1     voice-class sip profiles 1001   voice-class sip options-keepalive profile 20   dtmf-relay rtp-nte   no vad   For some reason it does not appear to match the profile that I have setup so it is not changing the Diversion header and removing the +1. ... View more

I'm trying to register the fxs ports on a VG310 to CUCME on 4331.  Telephony-Service has been configured but I cannot configure the voice-gateway system command. Is there some other command that I need to set to get this to show up? ... View more

I'm getting the following errors on the CLI Apr  4 15:00:00 USWEJVS02-1 local99 0 : 778: USWEJVS02-1.am.dynonobel.com: Apr 04 2018 21:00:00.63 UTC :  %UC_CERT-0-CertExpired: %[Message=Certificate expiration Notification. Certificate name:USWEJVS02-1.am.dynonobel.com.der Unit:CallManager-trust Type:own-cert Expir][AppID=Cisco Certificate Monitor][ClusterID=][NodeID=USWEJVS02-1]: Certificate has Expired and needs to be changed at the earliest Apr  4 15:00:00 USWEJVS02-1 local99 0 : 779: USWEJVS02-1.am.dynonobel.com: Apr 04 2018 21:00:00.65 UTC :  %UC_CERT-0-CertExpired: %[Message=Certificate expiration Notification. Certificate name:USLAVVS02-1.am.dynonobel.com.der Unit:CallManager-trust Type:own-cert Expir][AppID=Cisco Certificate Monitor][ClusterID=][NodeID=USWEJVS02-1]: Certificate has Expired and needs to be changed at the earliest   The problem is that these are a Unity Connection cluster and so they don't have a CallManager-trust certificate to renew.  If I try to renew from the cli it says that it can't find the cert.  Any help would be appreciated. ... View more

I have a Hybrid call setup that is working with the Spark App.  I'm now trying to register some devices for remote users.  Can I register a 7841 phone to use Hybrid call?  I don't have any call licenses since I am using Hybrid.  When I go to add a device in the admin for Desk Phone is says "Your organization must purchase Cisco Spark Call licenses first."  Is there any way to register phones to spark and use them with Hybrid call?   ... View more

Just moved from PRI to sip trunk on a 2851 with CME running IOS 12.4(24)T.  Inbound and outbound calls are working fine as well as direct calls to a phone that forwards to voicemail.  It is only when transferring to a phone and then forwarding to voicemail that I get the dead air. Found this bug https://bst.cloudapps.cisco.com/bugsearch/bug/CSCtl79911/?referring_site... But I already have codec g711ulaw on the inbound dial peer.  All call legs are g.711ulaw.  already have no supplementary-services moved-temporarily and no supplementary-services sip refer Any advice would be greatly appreciated. ... View more

We are moving to +E164 numbering plan.  Currently we only have a couple of users.  Hard phones are working fine with no issues.  However, CSF devices when used as soft phone the call is delayed by 15 seconds.  This is happening to internal 5 digit extensions as well as external calls. There is no overlapping patterns causing a problem.  The hard phone uses the same CSS as the CSF.  If I change the extension back to 5 digits everything works fine.  The trace shows that the invite is sent to CUCM and I see the digit analysis.  But then nothing for 15 seconds.  Then the call continues. Please help me understand why this is happening. Thanks CUCM version 10.5(1) Jabber versions tried 10.5 and 11.8 Information for trace Call was made at 19:04:50 Call continues at 19:05:05 Device name is CSFNCTEST with extension +18652204288 Dialed digits 54592 ... View more

For some reason I keep matching dial-peer 0 and I can't get it to match dial-peer 200.  Here is the config voice class uri CallManagers sip host ipv4:192.168.222.21 host ipv4:192.168.222.20 voice class codec 1 codec preference 1 g711ulaw codec preference 2 g729r8 ! ! ! voice class custom-cptone cptone dualtone disconnect frequency 425 cadence 75 ! voice class e164-pattern-map 1 description *** To CUCM *** e164 +6562916603 e164 +6562916653 e164 +656291659. ! ! voice class dpg 1 dial-peer 100 ! voice class dpg 2 dial-peer 200 ! voice class server-group 1 ipv4 192.168.222.21 preference 2 ipv4 192.168.222.20 preference 5 ! voice class sip-options-keepalive 1 description *** CUCM *** down-interval 40 up-interval 20 retry 6 ! voice-port 0/1/0 trunk-group PSTN supervisory disconnect dualtone mid-call supervisory custom-cptone cptone cptone SG timeouts call-disconnect 1 timeouts wait-release 3 connection plar opx +6562916603 caller-id enable ! dial-peer voice 100 voip description *** CUCM *** session protocol sipv2 session server-group 1 destination dpg 2 destination e164-pattern-map 1 incoming uri via CallManagers voice-class codec 1 dtmf-relay rtp-nte sip-kpml no vad ! dial-peer voice 200 pots trunkgroup PSTN description *** ITSP *** destination-pattern .T destination dpg 2 incoming called-number +656291.... ! Please let me know if you see anything wrong with this config. Also, when the call comes in the phone shows no codec and I'm getting one way voice. ... View more

CUCM 10.5(2) Integrated with LDAP. Four CUCM clusters integrated to a single AD for the whole company All of a sudden some users (not all) email id and directory uri have changed and removed the '@' from the address.  Since all of the clusters are integrated with the same AD I can search for the user in the other clusters and they show up correctly.  Only in their home cluster is it showing up this way.  Performing an LDAP synch doesn't change it.  I have tried to convert to Local user or make any other changes to it but it comes up with a popup saying "A directory URI is in the username@host format, where the host can be an IPv4 address or fully qualified domain name."  So it won't let me make any changes.  I'm not sure how this happened or what to do to fix it.  is there a way to force delete a user and re-import them in? if so will I need to re-associate all their devices? Thanks for any help. ... View more