can not dail out using grandstream 4104

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rayhooker

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I have everything registered but when I dail an outside number the grandstream answers with a dail tone. It did not pass on the number I dailed only gave another dailtone. Also when I dail in to the grandstream it answers with a dail tone, but it did
not pass the call on to the attendant??
 
It sounds like you have it set for two stage dialling rather than one stage.
 
How do I set it for 1 stage calling
 
look in the manual for the Grandstream There should just be a entry field with a 2 in it. it will need to be changed to 1 and rebooted.
 
Thanks very much for your help I found and set the grandstream to 1 stage dailing. But when I call in to the grandstream
I received a dail tone. The 3cx is set to forward to an extension, but it answers with a dailtone. Any thoughts on the incomeing side????
 
the grandstream is answering with dialtone, you need to set the forwarding on the grandstream device to your DR or group or what have you.
 
We had some issues with the 4104, it needs to have quite a few settings changed, especially for the UK.
If you get stuck I can send you a copy our config if you are in the UK to compare with.
 
rayhooker said:
But when I call in to the grandstream
I received a dail tone. The 3cx is set to forward to an extension, but it answers with a dailtone. Any thoughts on the incomeing side????

Do you see any log in 3CX during an incoming call to the Grandstream? If not then, (obviously) the call is not even leaving the 4104. I haven't worked with that ATA before, but, if the dialtone is coming from the 4104, then, it almost sounds like some sort of PIN request dialtone for security pass through. Just a guess.

Calls into the 4104 should (essentially) be "forwarded" onto 3CX. If you do see a log in 3CX, and it makes no sense to you, then post it here.
 
If you are getting the dial tone, it is the line settings.

for UK I had to change to this:
Call Progress Tones
[Syntax: ch x-y: f1=val@vol,f2=val@vol,c=on1/off1-on2/off2-on3/off3; ... ]
Note : f1,f2-frequency(Hz); vol-volume(dB); c-cadence(10ms, 0-continuous)
1. Dial Tone: ch1-4:f1=350@-13,f2=0@-13,c=0/0;
2. Ringback Tone: ch1-4:f1=400@-13,f2=450@-13,c=40/20-40/200;
3. Busy Tone: ch1-4:f1=400@-10,f2=0@-10,c=37/37;
4. Reorder Tone: ch1-4:f1=400@-10,f2=0@-10,c=40/35-22/52;


1. Wait for Dial-Tone(Y/N): ch1-4:N;
2. Stage Method(1/2): ch1-4:1;
3. Min Delay Before Dialing Out: ch1-4:500;

That sorted is out when we got the cisci 7940's getting a dial tone (which we could then dial out on).
 
leejor said:
rayhooker said:
But when I call in to the grandstream
I received a dail tone. The 3cx is set to forward to an extension, but it answers with a dailtone. Any thoughts on the incomeing side????

Do you see any log in 3CX during an incoming call to the Grandstream? If not then, (obviously) the call is not even leaving the 4104. I haven't worked with that ATA before, but, if the dialtone is coming from the 4104, then, it almost sounds like some sort of PIN request dialtone for security pass through. Just a guess.

Calls into the 4104 should (essentially) be "forwarded" onto 3CX. If you do see a log in 3CX, and it makes no sense to you, then post it here.

Thanks for your reply. I checked and the 3cx log does not show anything during the incoming call. I agree that the 4104 is answering. I can now dail out with no problem. Is there a setting on the 4104 to stop it from answering and only pass the call on to the 3cx.
 
I'm assuming that you've gone over all of your settings and compared everything to this...http: //www.3cx.com/voip-gateways/Grandstream-GXW-41044108/ NO LONGER AVAILABLE

All it takes is one missing, or misplaced, period, semicolon, or space, to prevent things from working properly
 
leejor said:
I'm assuming that you've gone over all of your settings and compared everything to this...http: //www.3cx.com/voip-gateways/Grandstream-GXW-41044108 NO LONGER AVAILABLE

All it takes is one missing, or misplaced, period, semicolon, or space, to prevent things from working properly


Thanks alot you have been very helpfull. It was a setting (Set the "Unconditional Call Forward to VOIP" field to: "ch1-4:yyyyy+;" where yyyyy is the SIP User ID for the first line defined in the "Channels" page.) Once I corrected this all
is working correctly. Thank you again.
 
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