Hi,   Are your ephones configured as dual lines? if not can you delete the ephones involved from CME and recreate it with dual-line mode and see if it makes a difference.   instead of:  ephone-dn use:  "ephone-dn dual-line" when creating the ephone.   Thanks! Jinto. ... View more

Hi Siva,   For PRI utilization alert, here is what you can do login to your CUCM through RTMT and go to System-->Performance you can see your call manager servers listed here, now choose the one on which your gateway is registered, you will get a big list of parameters here to choose, now click on callsInProgress -->right click on this and now choose an option   Counter instances-->choose your H.323 gateway and now you can see that there will be a graph created with current calls information in RTMT right-click on the graph now a new tab will get opened called Alert properties, give a description and recommended action and click next, here you can set up a threshold for example if the callsInprogress exceeds certain number it will send you a notification from RTMT to your email.   For monitoring a specific interface (D channel), you might want to use an SNMP monitoring tool and you might want to enable SNMP traps for interface status change. https://www.cisco.com/c/en/us/support/docs/ip/simple-network-management-protocol-snmp/13506-snmp-traps.html check for ISDN SNMP traps in this doc, it will give you an idea on how to configure it.   Hope this helps Thanks! Jinto.           ... View more

There are a couple of ways,   See below thread https://community.cisco.com/t5/ip-telephony-and-phones/pri-voice-utilization/td-p/745860 ... View more

Have a couple more questions, Please confirm below diagram is correct if this is correct your CUCM is sending 488 Media not acceptable, Do you have "MTP Required" option checked in the SIP trunk (If yes remove it), are you using standard SIP profile (If no use the standard SIP profile for this trunk)?  in the SIP trunk security profile for outgoing requests have you set it as UDP ("The default is TCP but for the right fax, it should be UDP ")   Also in SIP trunk under SDP Information make sure that you have to Send send-receive SDP in mid-call INVITE       ... View more

I would like to see below interaction,   ... View more

If you click those sip signaling diagram on RTMT you can View the contents on it (Click on invite and you can see complete Invite sip signal), from your non-working configuration can you post the contents of each message such as Invite, trying and also the contents of 488 SIP message so that I can review what's happening between CUCM and Rightfax. ... View more

In the signaling Diagram can you specify these signals from Which servers, you can replace the IP address with a name for example    if your CUCM IP is 192.168.0.12 replace it with CUCM in the signaling diagram so that it will be easier for me to understand the SIP signals.   Can you post that complete Invite message that CUCM sends what codec are you seeing in that, What's the region settings on your SIP trunk from CUCM to the FAX server (Make sure that its 711ulaw)   488 can be mostly due to a codec mismatch, but we would want to know where it's happening and based on that we can change settings to resolve this.     ... View more

Hi,   could you please try to directly hard code this to the dial-peers, don't use Voice class codec 2 (or just use one codec here)   codec g711ulaw fax rate 14400   Also, use    fax protocol t38 version 0 ls-redundancy 2 hs-redundancy 0 fallback none       let me know if it makes a difference.   Thanks! Jinto. ... View more

One solution for this will be to create a Dial-peer with higher preference than the ephone DN and then apply a translation rule to manipulate the digits however the way you want. Hope this helps For example   dial-peer voice 1002 voip destination-pattern 1019 . . . preference 0 <<=====This dial peer has precedence and will match first. ephone-dn 1 number 1019 preference 1 <<======Configure lower preference for prebuilt DN.   ... View more

Could you please post your dial-peer configuration from CUBE to CUCM, also the router configuration ... View more

Hi Ravi Kumar,   You might want to talk to your account manager to get that access issue resolved, only Cisco Employees or registered partners have access to the sizing tool.   if you don't have access, I would say check the OVA template you have deployed, based on that you can decide how many CTI route points, CTI ports, and Agent phones you have currently, you will get an idea, hope this helps     Thanks! Jinto. ... View more

Hi Ravi Kumar,   There is no limit to the number of Unified CCX clusters supported with a single Unified CM cluster as long as the combined agent phones, CTI ports, and CTI route points that are utilized by all Unified CCX clusters are used to size Unified CM. So in your case check maximum ports, CTI route points and Agent phones that your cluster can support.   https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cust_contact/contact_center/crs/express_11_0/design/guide/UCCX_BK_U3AF2742_00_unified-ccx-design-guide-11/UCCX_BK_U3AF2742_00_unified-ccx-design-guide-11_chapter_0100.html   Thanks! Jinto. ... View more

Hello, Check if you have a translation pattern, route pattern that overlaps with that phone's number, try to change its directory number and test.   Thanks! Jinto. ... View more