Hi, I’m fairly new to Grandstream stuff, I’m trying to set up phones around my boss’s house, there are 24 GRP2601P units and a UCM6301 interface. I have spent literally over 12 hours trying and trying and trying to get just one to work but can’t, I can get incoming calls working sometimes then other times when I try to ring the landline number it just says number busy. I know I must be going wrong somewhere, I have set up Analog Trunk, Incoming and outgoing Routes, but when I try to make a call it just says “the number you have dialed is incorrect, please check the number and dial again” even when I redial a missed call from my mobile, would somebody be able to assist me on where I’m going wrong? I have tried Numerous Youtube Videos but everybody is setting them up as VOIP phones and we need them set up through a normal phone line. Starting to drive me a little insane because I feel like I’ve tried everything so far, any help would be greatly appreciated, Thanks
this is a forum not the Grandstream assistance that you can find at the link below, if you are not familiar with the SIP protocol I suggest you contact a professional in the area, in any case you can ask for suggestions here, however below I will insert the link of the documentation:
Do you monitor calls from System Status > Active Calls? That might show if the analog port is already in use when you try to call and it’s busy. What patterns did you set up in your inbound and outbound routes? The number you dial has to match a pattern set there. You can use _X which is a generic catchall pattern.
Have you looked at CDR yet? It has info on all of the calls that go through the UCM and maybe you will find some useful info there.
You do need to set up all of the GRP2601P’s as VOIP phones. The easiest way is probably using zero config under Other features on the UCM.
There’s a lot more to all of this but hopefully that helps you a bit.
How these 12 phones are connected to your UCM ?
how did you configure them?
what is the format of the number u r dialing nd ur phone’s dial plan?
As you indicating that you are using an analog line, then the outbound rules need to conform to what he analog provider requires.
So, it could be that when you redial from your mobile, the mobile phone or provider is sending more than what is needed or allowed by the analog provider. The message you hear is more like those that providers feed back and not the UCM. Dial the destination using 7 digits, then 10 digits and finally 11 digits and see which ones get through.
Additionally, as there is only 1 analog line, only one call can be accommodated at any one time.
Hi, I have progressed to being able to dial out, the phone call comes through on my phone but when I answer it the GS phone is still ringing with the dial tone, also when I call inbound from my mobile there is no audio coming through from my mobile to the GS phone, what might be causing this issue?
On a side note I hope your Boss understands you only trying to help him but if this is on your job description and your responsibility you need to read the admin manual. getting the basics is not so hard many here can help you but you need to provide much more clear information
You need to confirm if you using sip trunk or Analog line or combination.
if sip trunk in use then confirm SIp ALG on a router is off has a starting point.
Did I not say I was using an Analog line in the Original Post?
you said a lot of things any way out of my scope I don’t use Analog lines. good luck
I think thats the problem I am having all the information out there isn’t for Analog, Seems not many use GS for analog
In my country you cant even order an analog line they been discontinued you can only use SIP.In any event you have 24 phones and single line maybe you have more but you have a 6301 which I think can take only one FXO so even if you get it to work your problems with 24 users may be only starting
It’s a House, So a majority of the phones will only be used for internal calling, if there are any issues with too many people trying to make outgoing calls we will have to deal with that then at the moment I just need the get it working first
Did you set up Pbx settings, SIP settings to include NAT? While not using SIP trunks you are using the phones.
Did you remove all the codecs by setting them same in the phones and the remove all but or two needed codecs in the extension setting in the UCM?
Try it one one test extension.
How were the phones provisioned?
THANK YOU THANK YOU THANK YOU!! Them bloody Codecs I literally tried to remove them from the extension earlier but they kept re-adding for some reason, I made a new extension and done it fresh and now it works One other thing tho, when I hang up from my mobile the line doesn’t hang up on the GS device Have tried re detecting pstn again but still doesn’t want to hang up on its own, what might be causing this?
It seems as thought UK users have more issues with this, than most. The key is identifying what CPC or disconnect signal is being sent and adjusting the UCM to accommodate. I am uncertain if there is a standard that is universally adopted in the UK, as there seems to be some variance which may be due to a different carrier or a different type of service (copper landline or cable modem).
You may be able to hook up an analog phone to the PSTN line and then repeat the call and hang-up the cell while listening closely on the analog phone to see if you hear tones or even a click or maybe both.
You can also do a forum search for analog line will not hang-up or disconnect and see what comes up.
Or, you can state who the existing carrier is and if copper, cable modem or other and perhaps one of the other UK users has an answer.
I am in the UK and struggled for weeks to get the UCM6301 to hang up analogue calls when closed by the remote user. I tried every single configuration possible with great help from this forum. I just could not get it to work so eventually I just created an announcement on the analogue which asked people to re-dial the SIP line instead and this way the UCM was clearing the analogue call once the announcement was made. Not ideal, but all my customers are now beginning to learn the new number.
Don’t you have number porting in your country… maybe you call it something else where you move a number from one network to another
Unfortunately we have an issue whereby our Broadband connection is tied to the number is it running across. Therefore if you try to port the number across it will automatically cancel your Broadband connection. Personally I have one analogue telephone number and one broadband connection, therefore if I port the number to SIP then everything stops working. The UK is literally a third world country now…