Cannot get normal audio from PSTN line. (GS4104)

[bertarecchia]

I have another problem with the setup of grandstream 4104. Maybe I should have gone with the patton.

I looked in the forum and on the web for similar problem. Some people have clicked the setting "PBX Delivers Audio". I did that under the extension setting and under the PSTN settings but I still have:

-Terrible call quality
-Some keys ( e.g. the * to access my cell phone voicemail ) do not work.

I spent 3 days on this setup ... a bunch of issues....

Any suggestion?!

 

[leejor]

My experience with MJ is...Outgoing only (via SPA3000), incoming is permanently forwarded to a PSTN line. It is run on a machine that runs no other SIP/VoIP programme/entity, and it's on a different LAN.

Some of your issues might be a result of how/where you are running MJ. You may want to remove it temporarily, to see if there is any change to the quality of the other (Comcast) line.

 

[bertarecchia]

You will need to play around with the detect/hangup on silence detection settings on the GS device.

I'll look for this and the gain settings and test them.

Did you have to do any firewall settings to get the MJ working?

No, just plug and play. This device has been a blessing for my first years of "mobile" office-type of work.

My guess is that you can leave the application on the same machine as 3CX given that you are essentially using 3CX in a PSTN mode.

If things don't improve I'll set up my laptop just for MJ. It's worth the try.

I also think that call quality will be more likely the result of the source line. If the quality is great with Comcast and it is also being routed through the GS device, then there is no reason to think that it or 3CX is able to single out and make the MJ port/line of a lesser quality.

Actually the quality is exactly the same on the comcast and MJ line. The extra problem with MG is the hanging up part.

On a side note, I'm trying to sort something out, the link to the configuration page posted in this thread, http://www.grandstream.com/products/gxw_series/gxw410x/documents/gxw410x_interop_3cx.pdf has conflicting instructions from the one that was suggested to me initially: http ://www.3cx.com/voip-gateways/Grandstream-GXW-41044108/ - NO LONGER AVAILABLE
The first one says to go to "FXO lines" and:
b. Set the “Off hook Auto Dial(VoIP)” field to “ch1-4:yyyyy;”
There is no such setting in my GS. The only one close to that is "Channel Dialing to VoIP" (suggested in the second link).
In one instruction it's suggested "ch1-4:yyyyy;" and the other "ch1-4:yyyyy+;" .
Should I set mine to "ch1-4:10000+;", with the + or without it?

Going nuts here...

 

[bertarecchia]

Update.
Some good news. I'm sure you guys were crossing your fingers No worries, I'll find more topics for new posts....

The quality has improved a lot ( let's see if it lasts ).

I changed the “Off hook Auto Dial(VoIP)” field to “ch1-4:10000+;”, the first SIP user ID ( which I still don't know what it does)
and
re-ran the FXO line test. One line at a time and one test at a time = 6 tests.
The Preferred Vocoder didn't change since the settings you suggested were already the default.
I think the fxo test did the trick.

Now the quality is decent but I can't use the *.# or any number so there's no way to call an office that has an automated system. Kinda sucks that a phone system ( 3CX and GS ) has this limitation. This is a problem since we call offices with automated systems often.
So, I'll have to keep the MJ out of the 4101 and use it for outbound calls.

Now I can spend some time on figuring out the 3cx phone book and the one that pops on the left side of the softphone.

Thanks for your patience and help.

 

[leejor]

Now the quality is decent but I can't use the *.# or any number so there's no way to call an office that has an automated system.

Once connected (answered), you should be able to send * and # as long as the DTMF Method in the phone matches that of the gateway. You will not be able to DIAL the * and # in an initial dial string (right after dialtone from the PSTN) sent from 3CX to the gateway.

 

[bertarecchia]

Once connected (answered), you should be able to send * and # as long as the DTMF Method in the phone matches that of the gateway.

You should see my face when I read message like the one above. To me it is the same as saying "you are the only one that can stop a nuclear war". :S

I *think* I found the page on the GS where to change the DTMF settings. The Dial Plans is the only page that has DTMF settings. Even in the grandstream so-called manual they don't tell the average joe how to configure strings.

My biggest question though, where is the DTMF method on the PHONE that you mentioned. If you have the time please tell me which lines and basic string I need to modify.

Thanks for your time.

 

[leejor]

When sending DTMF tones AFTER the called party has answered, when using a Gateway with PSTN lines usually relies on the gateway regenerating the tones. The phone will send a SIP message for each button press. On the 3CX phone (I may be using an older version so settings may vary from what you have) go to Connection Settings/ Advanced Settings.
There should be 3 check boxes with DTMF options INBAND, SIPINFO and RFC2833. I believe that as a default, all 3 are checked.

Most SIP phones also have these options, although they are sometimes given different names.

On your gateway there will be a similar set of options (I haven't used that one, so you're on your own). If the phone is already set to make use of any DTMF method available, then you may need to "adjust" the settings on the gateway, until you are able to send tones.

Be aware that the Magicjack site (when you log in to your settings) has a setting, that I found that I had to change, to allow DTMF to pass properly.

Was this just a MJ issue or were you also having problems on the Comcast line?

 

[lneblett]

If the GX, then the default settings should work as is "{x+}" without the " ". This essentially allows the use of all digits on the keypad. Use rfc2183 or in audio for DTMF your choice. I just used the in-audio and was successful using both * and the # on a FXS GS ATA.

 

[bertarecchia]

Really good info but as I went to apply it... all of a sudden I can't login the GS at all. I just get a white browser page, no error, nothing.
I unplugged everything and left the unit without power for an hour and I still can't login.

The leds look normal.
The 3CX softphone says Not Connected.

Suggestions?

Sorry, eventually I'll stop posting so much....

If I use the reset button, will I lose all the modifications?

 

[lneblett]

After it being unplugged, you get nothing or do you get a message that ie cannot find the requested page? did you accidentally switch LAN/wan cables (assuming you may be using both). Do you know the ip address that was issued and can you ping it? If you use the reset and hold for 7 seconds it will reset to default to factory settings. Hook it up to your laptop you mentioned earlier and see. On the surface, it sounds like an issue preventing comm and not a GS issue.can you get to the Internet?

 

[mylove4life]

If you reset it, DHCP will turn on and it will give it an IP. You can download there tool to find the IP. Just go to support and you will see it.

After it being unplugged, you get nothing or do you get a message that ie cannot find the requested page? did you accidentally switch LAN/wan cables (assuming you may be using both). Do you know the ip address that was issued and can you ping it? If you use the reset and hold for 7 seconds it will reset to default to factory settings. Hook it up to your laptop you mentioned earlier and see. On the surface, it sounds like an issue preventing comm and not a GS issue.can you get to the Internet?

 

[bertarecchia]

I forgot to mention that I did try to ping the IP address that has been assigned to it and it was responding. It's strange, there are only 2 IPs live now and neither one would answer with the /login.htm after it. After runing the GS sniffer i was able to login again, the same exact address ?!
OK, I'll try the suggestions tomorrow with a clear mind.

Thanks!!!!

 

[bertarecchia]

Thanks to you guys' help I was able to improve the quality of the audio, use the * and # keys and change the silence/timeout. It's very much appreciated.

Considerations / worries.

Ok, the flicker in the power yesterday must have rebooted my devices and the GS got a new IP. Here are the strange things / thoughts.

1-The IP that found by the Grandstream IPquery/sniffer tool was the same I was trying to use to connect.(only 2 IPs in the lan, so not a difficult guess) The http://192.168.0.100/login.htm (.101 or 102) I was typing wouldn't let me in but the IPQuery was able to get in, with the same url. What am I missing?

2- After the IP fiasco, the 3cx soft phones wouldn't connect and I'd get "server failure". I had to statically assign the old IP address to the GS to make the system work. I even tried to match the new, dynamically assigned IP to the GS on the phone and the GS system console but to no avail.

Is this normal when there is a power failure and the network devices get rebooted therefore the IPs change?

One other thing I don't understand: I created a global phone book on the system console but it's not getting downloaded on the individual 3cx phones. Am I missing something else?

If you guys ever need some discounts on Faux Stone Panels please let me know.

Thanks one more time!

 

[lneblett]

Well there are a couple of things at play. Keep in mind that this is not a traditional phone system. The devices register with the system and then reaffirm after some period of time as programmed in the devices themselves. This prevents the need for continual registrations which in most cases is only eating up time, bandwidth and system resources. Of course, if there is a need for more strenuous registrations - wireless extensions, remote, critical, etc., - then you can always chage the interval.

Not knowing the exact network setup - who is issuing ip, who is dns, and how each element may have been affected (or not) by the power glitch - makes it a more speculative answer than factual. Hopefully, now that you have a working system, future similar issues might arise, but will correct themselves as the registration periods expire and re-registrations occur. Consider a UPS to mitigate power issues; after all it is your phone system.

 

[leejor]

I'm not sure if I'm understanding correctly. You said that your gateway, the Grandsteam got a new IP?

If you haven't done so, assign the Gateway a fixed IP. Gateways should not be using DHCP as you are required to put the gateway IP in the 3CX settings for the trunk(s).

I prefer using fixed IPs for all SIP devices on the network, just for ease of remote "access", but it isn't necessary for Phones/ATA's.

 

[bertarecchia]

OK, I set my network to static ( I guess I could just do a static IP for the GS's MAC).
Let's see what happens.
I had to make it dynamic because the GS wouldn't come back after saving.

Thanks.

 

[bertarecchia]

Scratching my head again.
After hours of tribulations ( this must be the most counter-intuitive system I used in a while ) I managed to setup a personal phonebook from MyPhone page - still having issues with the company wide phonebook though.

Note that with entries I manually created on my local phone before, all I had to do to dial out was click on the phonebook entry and it would go. Now that the softphone is retreiving the phone entries from the server, when I select a contact to call ( same 10 digits as before) , I get " destination not found". What curious is that anytime after that if I press the Enter key on my keyboard the call starts fine.

I have setup an outbound rule before and it's still there.
So my question is: why the same 10 digits now don't work with a "downloaded" entry?
Do I need to add some kind of a carriage return in some rule?!

:S
Thanks.

 

[bertarecchia]

Grrrrrr... the frigging dashes........

Done, fixed, solved. Sorry for bugging.