I have to say I spent a lot of time parsing CUAC Standard 12.0 and CUCM 11.5 logs trying to figure out why CUAC Standard was sending exactly what I was telling it to, to CUCM, on Transfer direct to Voicemail. I set the CUACS console to use * as the prefix, in a +E164 environment. When trying the transfer, I would just get the "Invalid Destination" error. CUAC logs showed the correct number *+and11dig. CUCM logs showed the *+and11dig coming in, it would show the beginning of the digit analysis showing the CSS and partitions being used to try route the call, but would not do anything with the number. After hitting this post, I tried a Translation Pattern as suggested with *\+ instead of just *+ and it works. Thanks much for posting.
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I found the exact same issue in 2018, on CUCM/IM&P 11.5.1SU5 - my scenario was that we had done a PCD upgrade from 9.1, but due to some complex issues ended up building a new cluster doing a data Export/Import out of the upgraded 11.5 into a new 11.5. IM&P was built clean, not upgraded (due to PCD bugs). Services would take 20-30 minutes to start. HA would never go fully active. Cisco Sync Agent would start, then stop again. After finding this thread, I checked my Service Profiles, and of the 6 Service Profiles, 5 of them were really odd - when I clicked on them in the CUCM GUI, the page where they should have displayed was blank.
I was able to run a system Export of just that table, and determined that the directory area of the CSV was messing it up - a search base has comma-separated values in it, and my search bases all had the first dc= entry, then were missing the remainder of the field.
I added one manually with a good Directory portion, exported that from 11.5, then tried to copy/paste that area into the other Service Profiles and re-import... the Import says it succeeds, but it appears it fails on those because it's a CSV file and tries to process the next fields based on the comma in the Search Base indicating to move to the next column. Any new Service Profile being added using this Import method shows that it works in the Job, but because of the multiple dc= entries with commas, you get the same behavior - click on the Service Profile in CUCM to edit it, and it just shows a blank page.
I was able to get my import to work by just taking one Service Profile and not setting any options in the directory portion, then copying the Directory XML column value for that one Service Profile into the other Profiles in my original export file. I then manually updated the Directory area in each Service Profile once they were imported (and showed up correctly then).
EDIT: Found this bug that applies to the Service Profile failing on import, which is exactly what I ran into. https://bst.cloudapps.cisco.com/bugsearch/bug/CSCvf81790
What they don't tell you in the bug is that it impacts IM&P servers, causing the Sync service to stop.
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debug voip vtsp all may also be helpful. If that's too much, you can do debug voip vtsp default, and then add session as well if you need.
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One thing I just ran across (on 15.3 ( 3 ) M7 at least) is that the call treatment cause-code command has no impact if you use the call threshold interface command to limit calls instead of call threshold global.
My scenario was that the carrier wanted a 486 back on the 101st call on a SIP trunk in order to fail over to the other data center. Using call threshold global total- calls low 95 high 100, and adding the call treatment commands Michael Socher identified above, I was able to get the 486 to be sent back on the 101st call.
My dilemma was that I have a UCCE/CVP environment, and calls to CVP consume 2 legs per the global total-calls tracking. FWIW, forking calls to MediaSense also adds another leg. Because I had normal user calls as well as contact calls ingressing on this SIP trunk, I could not accurately state what the total-calls would be at a global level, because there was no way to determine how many would go to CVP with 2 legs, and how many will be recorded.
I thought I could do this on the WAN interface for the SIP trunk instead, so I used the command "call threshold interface GigabitEthernet0/1 int -calls low 95 high 100 midcall -exceed". This accurately counted only the calls on the carrier interface. However, this only sent back a 580 Precondition Failed message, it would not send back the 486 specified using the call treatment commands. I could not find any way to change what was being generated back. ( is this possible in SIP Profiles?)
The way around this that I found was to use the int -bandwidth command instead of the int -calls command. For whatever reason, the int -bandwidth command at the interface level sends a 486 busy back when the high bandwidth number is hit. This is my working solution:
call threshold on
call threshold interface GigabitEthernet0/1 int - bandwidth low 7840 high 8001
It is interesting to note that you must set the high bandwidth mark to just above the exact bandwidth needed (I had g711 calls coming in, it counts them as 80kbps per call - needed 100 calls, so 80*100=8000). If I left that to 8000, it would trigger the 486 on the 100th call.
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Geoff, I have also seen similar behavior, and did find a video detailing how to do LBCAC for CVP 7.0.2. It does describe the need for the delimiter, as well as specifiying the identifier in the Location name. The options and method for doing this have changed going to CVP 8.5, but this video was the first place I found mentioning that "--" delimiter. You can see it at http://developer.cisco.com/web/cvp/video/-/wiki/Developer%20Video%20Tip%20Series/Call+Server , look for the video titled Configuring Location and VXML Gateway Routing for IP Callers in 7.0.2 Maybe that will help a little... Loren
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I was specifically using voice hunt-groups with cme-srst to ring as many phones as possible at the same time (parallel or blast hunt group), without using shared lines. I did find a workaround for this - apparently all the gateway cares about is having an ephone configuration for at least one of the DNs in the list for the hunt-group. Manually adding a dummy ephone to the gateway and making sure it's dn was in each list allows the "final" command to work now, even if no real phones for the hunt-group list have ever registered to the gateway.
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I have a gateway configured with CME-SRST, using voice hunt-groups, on IOS 12.4.22T5. I ran into an interesting scenario and am wondering if anyone else has seen this. My configuration: voice hunt-group 1 parallel pilot 8001 list 7002,7003 final 7010 ! ! 7002,7003, and 7010 are phone DNs. Scenario 1: Phone 1 with DN 7010 is registered to the gateway in cme-srst mode, but not phones with DNs 7002 or 7003 - these phones have never been registered to the gateway. When I call 8001, I get a fast busy, and the gateway displays the following message: %IVR-1-APP_PARALLEL_INVALID_LIST: Call terminated. Huntgroup '1' does not contain enough valid SIP end-points to proceed with a parallel call. Scenario 2: I register phone 7002 to the gateway in cme-srst mode. Now when I call 8001, it rings on phone 7002, and if I do not answer, it rolls to 7010 just fine. Scenario 3: I unplug phone 7002, so it unregisters from the gateway. However, the command "show ephone unreg" displays that ephone configuration. When I call 8001, it follows the "final 7010" command and rings on my phone with DN 7010. It appears that a voice hunt-group will NOT follow the final command unless the DN's specified in the list have been registered at least once to the gateway. When I do nested hunt-groups, where the final command points to the pilot for another hunt-group, the call actually hangs if the second hunt-group's list members have never been registered and I have to restart the phone that placed the call. I have a customer with many CIPC phones, desiring multiple hunt groups for the CIPC phones, rolling from one hunt group to the next. It is quite possible that none of the CIPCs in the first hunt group will register (computers shut off), and based on my testing, the call will never roll to the following hunt groups using the "final" command if phones in the list for the first hunt group have never registered. Has anyone else tested this, and determined a way to have a hunt-group to follow the "final" command if the phones in the hunt-group have never been registered?
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I had a related question on this - but I have a voice translation profile applied to the voice port, expanding inbound numbers from 4 to 8 digits. Will the num-exp command take affect prior to the voice port translation-profile, or after the number has been expanded on the inbound voice-port? This is prior to any dial-peer matching. Thanks!
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Rob, Thanks for your response. That's what I had suspected based on testing and the description of the service parameter, but was hoping I was missing something. so be it.
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It would, but the customer does not want to do that. The people using their phones are agents in a call center, and the customer wants the agent to be able to use CAD or the phone without having to bother to choose line 2 if they use the phone for business calls. The customer was given the pro's and con's of having the DID as line 2 (and I had this listed as a con), and accepted it at the time of the design phase.
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I have users with two lines on their phones. Line 1 is an agent line with no VM, and line 2 is their DID line. I am trying to find a way to have Line 2 selected whenever the messages button on the phone is pressed. I have found the following service parameter: Always Use Prime Line for Voice Message. If this is true, it will always choose line 1 (which I don't want). If this is false, (which it is), the description says that it will choose a line that has a voicemail. I have some users reporting that when they press the messages button, it doesn't do anything (because I disabled it for line 1). Other users report that when they press the messages button, it automatically selects line 2 and works as expected, going to Unity. My guess is that those users have voicemail (MWI?) on line 2 of their phone, so it chooses that line for them. It always works if they select line 2 first, then press the messages button (which is how they were trained to do it). Is there a way to have it always select line 2 when pressing the messages button?
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Line appearances are working now after I did the following yesterday: associated all phones with the ac user, restarted the TCD service on pub and sub. I had logged into the AC shortly after restarting the TCD service and the line status still did not work. This morning I logged in and all lines show their state. However, the CcmLineLinkState counter is still set to 10. Is this a big deal or not? Thanks for any help.
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CCM 3.3.3(sr1) with AC 1.2.1.1. This is a pretty simple setup, with the operator being the only user logging in to AC. The operator can see calls come in and can answer, park, etc. using the AC. The problem is they can't see the line state of any other user in the directory. I have run the performance monitor report and the CCMLineLinkState counter is 10. Apparently this means that the TCD service is running, but can't pull the line state from each phone? I have restarted the TCD service on the pub and sub. Along with this question, does each phone that the AC sees in the directory need to be associated with the ac user, or do only those phones that need to be controlled by AC need to be associated with the AC? Previously I had only had the operator phone and my phone associated with the AC user, and with the problems tried associating all the phones to the ac user. The line status still does not show up. Any suggestions? Loren
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