Sorry, I don’t know, it could be any number of things. The flow type, channel order, something with the peer config or routing, if this just started and nothing else changed (which of course it never does) I would go with the reload/replace. There’s a second DSP you look to have in the flex pool for voice, which it could also use on inbound. If you run a call out you can see where it goes with the same show command.
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DSP 0 is broken, there's a number of bugs out for that being stuck in "FW_DNLD_FINISHED"
You would assume that it could try other resources but as "intelligent" as these things are, they're not that intelligent.
At the bottom of the show dsp-g all should be something like this:
DSP groups on slot 1: This command is not applicable to slot 1
1490 DSP resource allocation failure
If it failed to allocate, but, it's possible (speculating) that with the DSP being broken but it thinks it can pull credits from there that it is trying but it fails and you get an out of resources condition. I'd suggest to fix that first unless you know that you don't have an issue with it.
Congestion is equiv to a SIP 503 if it's being generated locally or relayed from a far end host. I can't say if that's the cause code that's generated if you have an actual issue with bandwidth or call limiting or something. But your DSP seems to be broken for sure.
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Also last time I tried this the external to hunt works as long as you don’t translate out - it seems to be looking at the trunk on/off net designator and blocking the calls going through there.
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RDP will trigger from Hunt last time I tried this, but remember that timers can be an issue here for remote and local VM.
This isn’t the best option for an on call page type situation due to this. Jabber will at least eliminate that mobile number component if you don’t go down that road.
If you use Unity Connection Id also suggest using dispatch voicemail and notifications (can outdial). Those do NOT work with Jabber but they do support this scenario.
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It is based on tracked phones and manual phones. So if someone has added manual phones that would do it , or exclude some IP range that’s registering to your UCM that you don’t want to deal with. It loads all registered devices via SNMP and goes from there plus manual devices.
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Sorry, the only S4B type doc I’m familiar with is the federation doc with Expressway or CMS for interworking. If you’re just trunking to S4B for audio conf or something you can just point it at the front end with the requisite return routes and it works okay but this is not a great integration.
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We prepare our own in house doc. The PIN Sync helps a lot, same with the PCA. The wallet card tree and or unity tools generator for that (does this still exist?) helps with navigating the conversation.
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Nope, we run various ATA and analog VG3x0 series with cat6a horizontal and don’t have any trouble. The ATA sometimes have their own problems with DTMF reception but it is not cable related. Given you say 6A I asssume the cabling was certified when installed and it’s not just a case of the 4p2c cable not seating in the jack right?
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In a way it's a mouse click, but it takes a few others to get you there if you don't have those pieces available. Jaime Valencia actually has published a video on how to set this up from beginning to end https://www.youtube.com/watch?v=p6m4gPv0ikE If this is your first time with these systems this may seem a bit daunting, but in reality it is not all that much. You are establishing trust between voicemail and the phone system, creating a credential that can be used to authenticate between, and then clicking the box to enable. I would do a bit of research on this as there's some comment about credential policies needing to be coordinated and that, but, all and all this isn't a bad mini project.
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Yes that in my experience is the same as the associated devices under the DN. I’m not sure what you’re asking about the dependency records. What do you want to know? You probably have to come in from another angle depending on what you want to know if it’s related to another function ( partition , vm profile , etc )
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Assuming it is in the same partition, check that the busy triggers and max call settings on that line appearance are sufficient to let it ring.
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The Self Care Portal is still available for a user to make line setting changes - I’m not sure that you can’t do this with only a CSF device.
Regarding the reorder tone, this is working as it should if there is no active station at the target DN. You can deploy these with a CFUR destination to a recording to let the caller know the person they are trying to reach is unavailable, generically.
Call Back or Camp On is not something that I’ve used personally, however even non Jabber users can display phone presence if tethered to an account, providing the equivalence to a BLF when on the users roster. This does not carry on to anonymous devices, but Jabber is user oriented. The availability state shown in the UCM doesn’t carry over to Jabber unfortunately as far as I can tell, as you can’t subscribe to presence for a line with the client. Not ideal but a possibility.
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Note the behavior for MRA in maintenance in that since the phone or jabber is effective registered through the E, it won’t figure out maintenance immediately and you’ll have a period where calls will fail to progress.
Youll also see alarms about the cluster that take a bit to clear while things reconverge but this is normal.
I ran ran into an issue with 8.10.4 using per domain DNS for MRA, where on one cluster upgrading to X8.11 broke this, which I could reproduce- didn’t affect another. Needed to reconfigure this feature for it to work properly .
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I had a case open for this because the doc is pretty bad. You can’t see these as the admin so it is easy to overwrite them. The “favorites” feature is also pretty useless and doesn’t work as the doc initially described but they slipped in a sly edit to the notes to backpedal on functionality.
If you have standardized layouts you’re golden, or if you can grab a screenshot before you make changes good.
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