Need some clarity on regions and audio codec preference lists with CUCM ver 11.5 and 8800 series devices. Without going into too much details we started doing additional testing when we noticed some failed recordings with our call recording server and was due to the server not supporting any codecs with higher bit rates than g711. In the below example I will refer to my regions as A and B.   When testing inter and intra site calling and verifying codecs we noticed that a inter site call between 2 8800 series devices the call was setup with AMR_WB (this is because we have OPUS codec disabled in CUCM). We then did the same test between the same phones but with g722 and iSAC disable on the device side and it then negotiated G729.   G729 is what we want between regions.  We do not have a region relationship configured between region A and region B so the regions not listed will all use the system default which is what is set on the service parameter clusterwide for audio codec preference list and maximum audio bit rate.  These default service parameters are set to 8kbps for max audio bit rate for interregion and factory low loss for default audio codec preference list between regions. Screenshot attached.  So my questions are below and if I need to clarify more information let me know.   - Why would disabling iSAC and G722 on the devices change the codec negotiated for inter-region calls if they both using the system defaults? - Are codec preference lists invoked with both inter-region calls and intra region calls or just between regions. - Does the default inter region and intra region audio codec preference lists and default max audio bit rate work together for codec negotiation. ... View more

I am having issues with one of our sites that is setup with a cisco 2921 that has 4 POTS lines in a hunt connected to FXO ports on the gateway.  Local calls complete fine but when dialing long distance I get the message "we're sorry your call cannot be completed as dialed".  I have looked and am matching the correct incoming and outgoing dial peers.  I have several other sites setup in the same fashion without any LD issues.  I would normally think this issue would be with the carrier but I can successfully complete a long distance call directly from the gateway via the CSIM start command.  An example of the outgoing dial peer is below.  I have tested with both the forward-digits 11 and no digit strip command and LD is still unsuccessful.  I have also attached a call log with several debugs enabled.  Any suggestions??   dial-peer voice 93 pots  description Local Outbound Dialing  preference 7  destination-pattern 9.T  progress_ind progress enable 8  progress_ind connect enable 8  port 1/0/11   rtr-haw-231#show dial-peer voice 93 VoiceEncapPeer93         peer type = voice, system default peer = FALSE, information type = voice,         description = `Local Outbound Dialing',         tag = 93, destination-pattern = `9.T',         voice reg type = 0, corresponding tag = 0,         allow watch = FALSE         answer-address = `', preference=7,         CLID Restriction = None         CLID Network Number = `'         CLID Second Number sent         CLID Override RDNIS = disabled,         rtp-ssrc mux = system         source carrier-id = `', target carrier-id = `',         source trunk-group-label = `',  target trunk-group-label = `',         numbering Type = `unknown'         group = 93, Admin state is up, Operation state is up,         Outbound state is up,         incoming called-number = `', connections/maximum = 0/unlimited,         DTMF Relay = disabled,         URI classes:             Destination =         Destination route-string = None         huntstop = disabled,         in bound application associated: 'DEFAULT'         out bound application associated: ''         dnis-map =         permission :both         incoming COR list:maximum capability         outgoing COR list:minimum requirement         Translation profile (Incoming):         Translation profile (Outgoing):         incoming call blocking:         translation-profile = `'         disconnect-cause = `no-service'         advertise 0x40 capacity_update_timer 25 addrFamily 4 oldAddrFamily 4         mailbox selection policy: none         type = pots, prefix = `',         forward-digits default         session-target = `', voice-port = `1/0/11',         direct-inward-dial = disabled,         digit_strip = enabled,         register E.164 number with H323 GK and/or SIP Registrar = TRUE         fax rate = system,   payload size =  20 bytes         supported-language = ''         preemption level = `routine'         bandwidth:             maximum = 64 KBits/sec, minimum = 64 KBits/sec         voice class called-number:             inbound = `', outbound = `'         dial tone generation after remote onhook = enabled         mobility=0, snr=, snr_noan=, snr_delay=0, snr_timeout=0         snr calling-number local=disabled, snr ring-stop=disabled, snr answer-too-soon timer=0         Time elapsed since last clearing of voice call statistics never         Connect Time = 102355, Charged Units = 0,         Successful Calls = 58, Failed Calls = 0, Incomplete Calls = 0         Accepted Calls = 0, Refused Calls = 0,         Last Disconnect Cause is "10  ",         Last Disconnect Text is "normal call clearing (16)",         Last Setup Time = 47290722.         Last Disconnect Time = 47292069. ... View more

We have recently moved some of our toll free numbers that live on a PRI over to a newly provisioned SIP trunk (Level 3).  We did however get a report from our operators that when the calls come trough on the SIP trunk that their is a delay when they pickup the call.  The UCCX agent answers but the calling party may miss the initial part of the greeting making the agent repeat themselves.  They have also said that their is a 2-3 second delay on when getting off one call and receiving the next.  These issues do not happen on the toll free numbers that still reside on the PRI and it only seems to be related to the SIP trunk calls and not the UCCX script itself.  Has anyone experienced this before when migrating inbound telephone #'s to SIP?  Please let me know if I am being unclear or if I need to add some additional information version and call flow below.   CUCM Version: 10.5.2 UCCX Version: 10.6.1   Call Flow:  ITSP SIP Trunk ==> CUBE ==> CUCM ==> UCCX RP ==> CTI Port ==> Agent ... View more

We have a CTI port (7022) configured in CUCM specifically used for voicemail so the forward all to voicemail on the line should be checked at all times. This is also a line(shared) on some users phones  so they can select the line and hit the messages button and unity automatically prompts them for the PIN recognizing 7022 as the mailbox ID. I have noticed at times the check box would get cleared/removed from the line and calls will ring to it instead of going directly to voicemail. I believe a user that has this line on their phone is somehow checking the voicemail improperly in turn removing the forward all to VM. My question is how am I able to pull CUCM logs when this happens to see which user or phone will be doing this and what parameter do I search for in the logs? I assume a user is selecting the line then hitting the call forward softkey and then the messages button but I am not 100% sure. Any help would be appreciated. ... View more