I think your problem is that CM can only be authoritative to one domain. If one SIP user is calling another and they have different domains, the originating or terminating application sequence is going to fail for one of them. I don't have a good answer for you beyond changing all your domains...
I think you could try doing just plain G711 and not offer t38. AFAIK, the Viper phones do a V150.1 negotiation that is encapsulated within T38. I think you might be in a situation where t38 is offered by the carrier, and is transported end to end, but the media gateways in the PSTN between you...
Pretty sure that's a feature request. Otherwise, you can run a little script against the web APIs for IX and probably change it that way if you want the automation. Not fun.
Yeah. This part:
Calls hit the group normally through a vdn/vector.
Have a global variable 1-3 for 3 people on call
If inside of business hours, do the normal thing
if out of business hours, go to step 10 if global variable = 1, 20 if variable = 2, 30 if variable = 3
The treatment to look for...
EC500 is still a call on your station. A virtual station can't take a call, it can only use a coverage path. You can try a real x ported station to see if it's any different, but AFAIK, once the call hits coverage, it's not on the station anymore so you can't record the station.
So everybody needs to get calls on their extension and hit their EC500 after hours for personal calls but there needs to be a separate configuration so that calls to a hunt group or other department number would go to just the one person that's on call and you want a button on each phone to...
If you can record all a station's calls and you're always going out to the same number, I'd try setting up EC500 on a station to the destination number and try recording the station that way
It's on WebLM. Either standalone or in SMGR or in System Platform. Should be an option to export licenses from there to a zip file on the local file system of whatever is hosting WebLM.
I have no idea. I'd imagine you're just recording whatever codec the stream is in, which means if you're using H323 via DMCC that you're limited to the codecs supported by H.323 would be up to G.722.
If you have an Avaya Media Server and not G430/450, you could try setting up opus only in a...
Did you check for that trusted tick box on the entity or entity link? First thing in your call processing trace is that the remote host is not trusted. Could be that.
No, you'll need an SBC.
You can try doing a traceSM with call processing enabled to confirm the idea that the invite doesn't match a known entity link.
I have no idea why you`re having so much trouble...
Do you have "enable grooming" on the SM server? You posted that you have it on KCell.
I'm pretty positive you're getting a 407 because SM wants to match you to an entity link on same IP/port and if that doesn't line up, then it tries processing you as a set and answers with a 407 to get...
I'm just looking over your initial post and the 407 comes from SM. I guess you didn't post that message.
In the advanced tab of the server configuration in the SBC towards Session Manager, can you tick the enable grooming check box
So it's something wrong between your SBC and the SIP carrier. The 407 isn't coming from SM, or at least you haven't posted that message. The initial 407 is from the SBC to the carrier. Are you using registration trunks with your carrier? Maybe it's something about them sending you calls on a...
Yeah, so it's probably in the server configuration where you punch in the IP for SM in the SBC. If memory serves, it has TCP and UDP enabled and if you don't remove UDP, the SBC will use that first and that's why SM would be giving a 407 - because you don't have an entity link for UDP...
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