About NicP270591

NicP270591 · ‎07-28-2019

Hi Maren, Yes you are correct. I am referring to phones in CME. While I understand that dial-peers and translation rules can indeed provide this functionality, I was interested in a more future-proof solution in case of growth. If we did it between just these two sites, then yes peers and rules would be fine. But, if we grew and needed to deploy more than a few extra locations, that would likely require logging in to each router every time and adding translations for all the extensions at other sites. However, if each site/extension could just be pointed out to GK and register call processing/destination info there, then I would prefer to handle it that way. Fine on 2900/3900 hardware, but 4000 ISR's seem to have done away with this functionality - hence I am now wondering what the "modern" alternative to this functionality is. Nic.

NicP270591 · ‎07-25-2019

Hi all, I have noticed in recent years Cisco is migrating away from SCCP and H323 and doing everything via SIP protocol. While I have not used the gatekeeper functionality personally yet, I have seen it implemented in other deployments, for inter-site dialling. eg. - Create SCCP extension, add a secondary number which registers with GK, to call in-between two sites, dial secondary number instead of primary ext. I may need to implement the above example functionality soon. While I have SCCP deployed on 2900 routers, it could be done easily. But it doesn't sound future proof the way things are. - I have also seen that ISR 4000 doesn't support GK functionality. What is the new/modern alternative to my above example? I have searched but cannot seem to find a simple answer. Is there a way to register secondary numbers on both SIP and SCCP extensions for inter-site dialling without a H323 GK? Nic.

NicP270591 · ‎05-28-2018

Hi All, I have a router where I have custom ringtones installed and working fine. When a phone is set to have a custom tone as a default ring, parked calls which timeout and return to the phone ring back with the custom ringtone at a slightly faster speed than normal to signify it is not an external call coming in. This is the correct behaviour. However, I have another router where, the custom ringtone is set. Regular Internal Calls to the phone ring with the custom ringtone, yet when a call times out from park, it returns to the phone and plays the Cisco default Chirp1 ringtone instead. I am sure this behaviour is usually handled by DistinctiveRingList.xml, which I have created by coping the RingList.xml file, name changing to Distinctive, copying and flashing to the router. Is this correct, or should I be looking elsewhere to solve? Nic.

NicP270591 · ‎05-28-2018

A Physical Power-cycle seemed to fix the issue! I originally reloaded via CLI but it did not take. This seems to have done the work. Thank you for all your help! Greatly appreciated, I have rated as helpful. Nic.

NicP270591 · ‎05-28-2018

Hi Maren, The first time I reloaded via CLI it did not have any affect. However I have just done a physical power cycle at the router switch, and this has cleared the issue. Thank you! Nic.

NicP270591 · ‎05-28-2018

The only way I have tested is via two internal SCCP phones, as there is not yet a SIP trunk service delivered for outbound calling. I have had telephony-service shut for 5 minutes now, I can still call between phones. Neither one of the two I am testing with has unregistered or reset. They act like nothing has happened. I shall reconfigure telephony, but given the above results, I'm not too hopeful. Nic.

NicP270591 · ‎05-28-2018

Shut/No Shut produced no results. When I did shutdown telephony-service, phones still continued to be up and running and could still call between one-another, I assumed this command would halt all telephony activity, am I correct or is something wrong there? I have not deleted telephony-service and started again, as I am sure it would have little impact, but I may do that tomorrow and see what happens. Any other possible ideas? This is extremely unusual behaviour. Still TOH and no MOH. Nic.

NicP270591 · ‎05-28-2018

Nipun, followed those commands, redid commands with music on hold from the CME 12 fileset. Same result. No-go. Skinny Music-on-Hold Process is NOT YET STARTED! ------------------------------------- Skinny Music On Hold Status - group 0 Active MOH clients 0 (max 210), Media Clients 0, B-ACD Clients 0 File flash0:MOH/music-on-hold.au (not cached) type INVALID Media_Payload_G711Ulaw64k 160 bytes G.729 Music-on-Hold File Status No MOH file loaded Incorrect naming structure from the MOH folder on flash? Tried multiple ways. Otherwise, something else wrong. I used the no form of the moh command, redid it, no created cnf and recreated cnf under telephony service. flashed new moh file to tftp. Nothing. You mentioned an ip route being needed due to the phones being on a different SVI. Can't see anything wrong but its early hours here so I may be missing the obvious. Nic.

NicP270591 · ‎05-28-2018

Indeed a stretch Maren, unfortunately did not work. But thanks for the suggestion :) Nic.

NicP270591 · ‎05-28-2018

Hi Nipun, Phones are SCCP. Please see below outputs: Skinny Music On Hold Status - group 0 Active MOH clients 0 (max 210), Media Clients 0, B-ACD Clients 0 File MOH01.au (not cached) type INVALID Media_Payload_G711Ulaw64k 160 bytes Moh multicast 239.0.0.1 port 2000 route 192.100.20.1 192.100.21.1 G.729 Music-on-Hold File Status No MOH file loaded Statistic Count ------------------------- ----------- Send queue enqueue error 0 Socket send error 5 Msgs discarded upon error 4 As for the file, it has been a very long time since used, but it was pulled from a working router, have opened it and rechecked. It meets all the listed requirements Looks like it could be a routing issue as mentioned. Nic.

NicP270591 · ‎05-28-2018

Hi Sam, Thanks for the reply! Enabled proxy-arp on both voice interfaces, still getting tone on hold, MOH is still not operational. Nic.

NicP270591 · ‎12-26-2017

Thanks Brandon, So I would need a seperate voice port for every seperate simultaneous call into the system and pre-guess the call volume levels? I'm assuming if all licensed ports are busy when someone calls they would just get the automated "sorry, voicemail is busy, try again later" message followed by an automatic disconnect?

NicP270591 · ‎12-26-2017

Hi All, Soon to implement my first CUE system on ISR G2. In regards to port licenses, what constitutes a "simultaneous connection to the system" for licensing purposes? For example, if the CME auto attendant (B-ACD) on my main business line is setup to redirect to the CUE module after hours, and I have 10 people calling that inbound number at once, and they all choose to leave a voicemail at the same time, does each one use up a port license at the same time? or are port licenses only for my people dialling into CUE system internally to check their voicemails? Anything else I may have missed? My apologies for the "basic" question. Searched all through the forum and Cisco docs referenced on similar posts but couldn't find a definitive answer on this version of the question. Please clarify. Thanks in advance!

NicP270591 · ‎10-26-2015

Hi all, I have recently acquired two Cisco 7965 IP Phones, booting them up and placing them side by side, I noticed that the text size of the actual Soft Keys, system message, and extension number is different on each phone. One is large font, the other is smaller. They both work fine, and I believe they are both loaded with the same SCCP revision (SCCP45.9-3-1SR1-S). Has anyone noticed this behavior before, and is there any way to get them to the same size? Nearly insignificant I know, but as these phones will be placed side by side we'd prefer them to both have the larger font and look virtually identical, the larger size seems to help a little with employees who have difficulty seeing clearly. I'm assuming the fix is to check revisions and then upgrade/downgrade firmware on the smaller text phone to get it back to larger size, but just wanted to confirm? Thanks, Nic.

NicP270591 · ‎10-22-2013

I shut down my 1007 for testing and the call failed. But normally it works fine. Very strange, I will do some testing next time I get the chance and report back. May take a couple days however before I can get on-site. Nic.