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Known Bug. upgrade

Hot desking is a joke on J series phones on Version 11. Anyone tried the H323 load on them then connect to version 11?

3 site SCN using one-x mobile. Does anyone have any idea why a user at site A would get the sip login info for one of the other sites randomly? for example user Joe ext 1000 is getting the fqdn of site 3 ipo3.phonesystem.com as the sip server. I can see in system status that Joe is trying to...

You will be able to see the DTMF in vm debug. I just had to do this recently the test call was to 222: I think I had level 9 high trace on the vmpro service. You can also set a vmpro callflow to read you back digits you dial. 17/12 11:23:34.983 vmprov5s (1b,5) 5d0, 690: IClient::DtmfDetect...

I didn't see it in your post, is it on an IP 500 and not server edition? IF 500 I think you have bad VCM resources. I have seen something similar before. you said G711 is fine and sounds like the issue I had.

Echo Canceller: Press HOME (Key with The A on the button, exact center of the phone) Press arrow down button to ADVANCED OPTIONS, press OK Press arrow down button to AUDIO, Press OK. Press arrow down button to ECHO CANCELLER, Press CHANGE to turn off. Press SAVE. Press BACK, Then EXIT.

On what type of trunks? You're using 1608 phones so it has a hard code transfer key. On my 9608 when I make a call I do not get the transfer button to appear until the call is answered on the far end. I'm using PRI. I can't see behind the scenes as to why your other system is doing it, but I...

You won't be able to transfer ringing calls. The IPO doesn't see it a a "call" yet for lack of a better term and can't control it. I have seen this on PRI in the US when users could not transfer calls that were answered because the carrier had not sent an answer supervision message that the...

You mention there are analog lines on the system, but how are the lines delivered? Are they over a cable modem? Based on the limited info I don't think your issue is with the IPO at all, I think the carrier is having an issue. If you disconnect the main line and plug into a butt set or analog...

Pepp77 Are you asking that about my suggestion? If so, I route calls in and out this way all the time. There's no issue with this.

You can't do this with SCN, but you could do it if you turn SCN off and just set the trunk as H323 without SCN. Then you could create shortcodes to allow or disallow dialing to anyone on each side. However you would lose the other SCN features like voicemail pro and BLf, etc. depends on how...

I'm wrong often, but doing this way has never caused me an issue so I will stick with it.

If you went 7->9 that's wrong. should have been 7->8.1.95->9 Probably why you had some weirdness? Can't say.

Do you have coverage of some sort turned on or are they part of a group those 2 phones? Do you have log missed calls answered at coverage and/or Log missed hunt group calls turned on? I have had customers complain of this as well.

It sound like your IP in not in the SBC for the failover. Since your using user and password I would think you wouldn't have this issue. Unfortunately every sip provider does things a little different so maybe they need your IP as well. Did they ask what IP you were coming from?

For this sip trunk are you using ip authentication or username and password? what is the sip provider?

Then don't do what I suggest.

You regulars crack me up.

In my book, if the tick goes away because of something I did. I fixed it. Anytime I don't have to get the carriers involved, whether it be to fix my problem or theirs its a win.

Because clicks = Slips. Look up PRI slip. Its timing. will present itself with analog ports dropping calls. On digital phones its not as cut and dry. Unfortunately I have had a lot of experience with PRI timing issues. Mainly my own doing early in my career. But maybe I'm wrong like joe says.