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?!

 

[mylove4life]

have you run the line test on it?

 

[bertarecchia]

Do you mean this test here?
http://www.3cx.com/blog/voip-howto/test-voip-provider/
That seems to be for an actual VOIP deidcated line. I don't have a server. Nowhere in the grand stream is things like INVITE server status. There is no DID number with these lines.
I tested the other things I can understand and the test I can perform.
I can make calls, transfer to extensions and hold successfully. I can hear the hold tone and DR announcement.

Are you asking if I did a test before getting the Grandstream? Yes these come out of the magic jack and comcast modem as regular phone lines ( not VOIP ) so I made calls for months.

Calling INTO the lines e.g. from my cell phone, the greeting and other sound are very clear. The problem seems to be only dialing out

I'm thinking the grandstream is not the best product for this use.
I know someone on the forum told me he's using the same setup and it works for him but I'm getting the same exact problems ( quality and can't use the * ) with both the MJ and comcast lines.

I'm including the log of a call. I see some errors but I'm not sure they mean anything:

[CM503007]: Call(12): Device joined: sip:[email protected]:5064;transport=udp
[CM503007]: Call(12): Device joined: sip:[email protected]:50215;rinstance=d527c09e2b281fd2
[CM505002]: Gateway:[Grandstream_GXW4104] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Grandstream GXW4104 (HW 2.0, Ch:10) 1.3.4.10] PBX contact: [sip:[email protected]:5060]
[CM503002]: Call(12): Alerting sip:[email protected]:5064;transport=udp
Currently active calls - 1: [12]
[CM503025]: Call(12): Calling PSTNline:540xxxxxxx@(Ln.10002@Grandstream_GXW4104)@[Dev:sip:[email protected]:5064;transport=udp]
[CM503004]: Call(12): Route 1: PSTNline:540xxxxxxx@(Ln.10002@Grandstream_GXW4104)@[Dev:sip:[email protected]:5064;transport=udp,Dev:sip:[email protected]:5066;transport=udp,Dev:sip:[email protected]:5060;transport=udp,Dev:sip:[email protected]:5062;transport=udp]
[CM503010]: Making route(s) to <sip:[email protected]:5060>
[CM505001]: Ext.101: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CXPhone 6.0.20943.0] PBX contact: [sip:[email protected]:5060]
[CM503001]: Call(12): Incoming call from Ext.101 to <sip:[email protected]:5060>
[CM503008]: Call(11): Call is terminated

 

[bertarecchia]

Does anyone have a suggestion for the above?
I'm stuck with a setup that isn't working.
Am I wrong to think that the grandstream is the problem, given that both magic jack and comcast lines have the same audio quality issues for outbound calls?

 

[leejor]

The quality should not be different in only one direction , when using the same phone line. I'm assuming that you've tested the lines with a "nornmal" phone and tried moving the lines to spare ports on the gateway? You might have a bad port (firmware issue/corruption?) on the gateway,always a possibility.

and can't use the * ) with both the MJ and comcast lines.

That is normal. I've found that neither the * or # will pass properly from a 3CX extension onto and out of an analogue Gateway (there may be some exceptions). On a PSTN line (in most cases) you can substitute "11" in place of the * for short code activations.

 

[bertarecchia]

The quality should not be different in only one direction , when using the same phone line.

Sadly It is very different. Dialing out is very low in volume, noisy and crackly. I hooked up one line at a time to make sure I dial in and out from the same one.
What makes me believe is something with the gateway is that the exact problems are on both lines. Inbound good, outbound bad.

I'm assuming that you've tested the lines with a "normal" phone and tried moving the lines to spare ports on the gateway? You might have a bad port (firmware issue/corruption?) on the gateway,always a possibility.

Actually no. I used the lines with regular phones until I got the grandstream. After that I did all kinds of testing with it and the 3CX phone. That's why I mentioned testing the transferring, hold and others.

and can't use the * ) with both the MJ and comcast lines.

That is normal. I've found that neither the * or # will pass properly from a 3CX extension onto and out of an analogue Gateway (there may be some exceptions). On a PSTN line (in most cases) you can substitute "11" in place of the * for short code activations.

I tried to type "11" but no go.

What would you do at this point? Get a replacement / try a patton? The setup is useless right now.

What kind of testing were you referring to for a line that is more like a PSTN than a voip?

Thanks!

 

[leejor]

I tried to type "11" but no go.

This will only work on a feature that will respond to the 11 when using a rotary phone (obviously a "regular" PSTN line that will work with a rotary phone). What you are trying to activate, may not be the same thing.

What kind of testing were you referring to for a line that is more like a PSTN than a voip?

Did I suggest something?

What would you do at this point? Get a replacement / try a patton? The setup is useless right now.

It sure seems to be a Gateway problem. Maybe you did get a lemon.
What I would do is...backup/make note of, the settings in the gateway.

Do a factory reset, or even a re-load of the firmware (check that you are using the latest (but not a beta) and that is does not have any bugs). Just put in one line and test it to save yourself some work if it doesn't help and you are doing it manually.
You may want to go to the Grandstream site and make use of their forums, someone else may be having the same issue.

It's a possibility that a slightly older version of the firmware will perform better, for you. I've seen that happen before with other SIP devices.

 

[lneblett]

The line test that mylove4life was suggesting is the embedded test in the 4101. I think it is described on page 21or22 of the manual. There is also a configuration guide for the device and 3cx on the GS website that may also be of benefit.

 

[leejor]

If the gateway uses a separate power supply, the problem could be caused by it being faulty, although you'd think it would be on calls in both directions.

 

[bertarecchia]

The line test that mylove4life was suggesting is the embedded test in the 4101. I think it is described on page 21or22 of the manual. There is also a configuration guide for the device and 3cx on the GS website that may also be of benefit.

Lneblett. I really appreciate you taking the time to understand my question!
I knew I missed something. Although now obvious, I couldn't understand what mylove4life was referring. Frustrating.

I must say the the data, file, pages and basic stuff to learn is a bit overwhelming for a new user.

So, now that I understand what test, I ran it and I see a huge difference in the quality. The only problems left are:

-The audio level coming IN is still pretty low. I changed PC and headset ( thinking it was my PC ) but it's still low. Calling out the volume is much better.
-On one of the lines ( the Magic Jack) after the calls the port stays open.
-During the test I noticed that port 1 went through a long series of blinks but when testing port 2 it just blinked 2-3 times and the test stopped. See attached screenshot for the test setup I had. I'm not sure if it's supposed to but the phone number provided never rang.

Thanks!!!

 

[leejor]

What codecs are you set to use (3CX to gateway)?

 

[bertarecchia]

What codecs are you set to use (3CX to gateway)?

I'm very new at this guys......

I looked everywhere in the grandstream and 3cx console, plus the softphone...where would I find this setting to give you an answer?

Thanks.

 

[lneblett]

You will find the codecs on the advanced page in the device's web browser. Grandstream oftentimes refers to them as "vocoders" or something similar. This determines the type of scheme that will be employed to carry the voice and can play a part in the perceived quality. You will likely want the first to be 711u followed by 711a and then 729.

On another note, keep in mind that you should only test one line at a time. Depending upon which test was conducted, it may not ring through (ac impedance for example).

Also, could you explain your setup a little more? You mention Magic jack and I also seem to recall Comcast in the mix.

Also, keep in mind that if you are also using a soft phone that the PC has a mixer that influences both mic and speaker levels.

Look here as well for other items related to the device and 3cx:
http://www.grandstream.com/products/gxw_series/gxw410x/documents/gxw410x_interop_3cx.pdf

 

[bertarecchia]

You will find the codecs on the advanced page in the device's web browser. Grandstream oftentimes refers to them as "vocoders" or something similar. This determines the type of scheme that will be employed to carry the voice and can play a part in the perceived quality. You will likely want the first to be 711u followed by 711a and then 729.

On another note, keep in mind that you should only test one line at a time. Depending upon which test was conducted, it may not ring through (ac impedance for example).

Also, could you explain your setup a little more? You mention Magic jack and I also seem to recall Comcast in the mix.

Also, keep in mind that if you are also using a soft phone that the PC has a mixer that influences both mic and speaker levels.

Look here as well for other items related to the device and 3cx:
http://www.grandstream.com/products/gxw_series/gxw410x/documents/gxw410x_interop_3cx.pdf

That's better. Thanks. Really.
The only area that has something similar to what you explained in in the attachment. It has a value of "2". Not sure how it translates to lejor's question ((3CX to gateway).

I see so I should fill one line at a time to test. I'll try that.

As far as input/output levels on the PCs setup, I have a good knowledge of.... at least something
It's not an issue with audio levels.

The setup - very simple.
I have my internet connection via comcast, the package includes a phone line. Their modem has a rj11 on the back for it.
I have 2 PCs. On one of them has the Magic jack plugged via USB. The MJ has also an rj11 on the back side.
I connected the comcast line on port 1 of and the MJ on port 2 of the 4104.
Both PCs have the 3CX softphone installed.
At the moment the comcast line works well except for the volume and the fact I can't use the * .
The MJ is like the comcast line but with an extra problem. The line/port stays open even after the call ends. I changed ports and it does the same thing.

That's another link I didn't see before....there are a lot of pages out there, some redundant. I did follow a very similar one to do the initial setup.

I really appreciate the help.

 

[lneblett]

Sorry, I was mistaken about which tab codecs are set. Go to the profile page and set the preferred vocoder. Then go to fxo tab and set stage method to one for ch1-4:1

 

[bertarecchia]

Sorry, I was mistaken about which tab codecs are set. Go to the profile page and set the preferred vocoder. Then go to fxo tab and set stage method to one for ch1-4:1

I may be still misunderstanding but the codecs you mentioned before (711u etc ) aren't in the pull-down list. Please look at the attached.

The stage method seems to be already set to 1.

Thanks!

 

[leejor]

At the moment the comcast line works well except for the volume and the fact I can't use the * .
The MJ is like the comcast line but with an extra problem. The line/port stays open even after the call ends. I changed ports and it does the same thing.

It's a shame you don't have an ATA or SIP phone to test with. Your PC's may be introducing some of the audio issues and without something else to compare you are just guessing.

The best quality (high bit rate) codecs are the G711A/U, and are usually the default first and second choice. You will usually only choose something else when bandwidth is of a concern, which it usually isn't when the call is staying on the local LAN (3CX to gateway).

On your MJ line, when you say the line stays open after the call ends...are you talking about all calls, both directions, even when you call out? Or...is it when a call comes in, and is connected to something like, voicemail? Are you using a MJ plus? Is the PC running MJ also running 3CX?

 

[bertarecchia]

I think we are going in circles.
The Line Test made a huge difference. I would like to complete lneblett's suggestion and see if it helps.
I got the grandstream and 3CX to avoid getting SIP phones.

As mentioned the choices in the Profile Page do not include G711A/U. Please see attached.
In order to complete the test suggested ( it might solve my problem ), which choices would you suggest:
GSM, G.723.1, G.729A/B, PCMU, PCMA? Any specific order? Bandwidth is not a concern.

Outbound calls on the MJ don't leave the line open. Calls coming in that either get answered or that go to the VR/voicemail remain open for almost a minute.
The PC with the MJ also has 3CX but the one I'm using for testing doesn't have MJ. Should I setup a laptop to handle the MJ, without 3Cx, for testing?

Thanks!

 

[leejor]

G711U = PCMU G711A = PCMA


Just put the U first and A second, the others can stay, they will be either unsupported, unnecessary, or ignored.

 

[lneblett]

Leejor is correct. Like many things, there is more than one way to reference a given item. PCMA and PCMU equate to g711a and 711u respectfully; set as he suggested.

As far as MagicJack - this device does not send a hangup tone or signal; hence why the line appears to stay off-hook. You will need to play around with the detect/hangup on silence detection settings on the GS device.i guess the assumption from their point of view is that when a call is completed that both parties will terminate their respective end and will hangup. Without some kind of signal, the device has no way to know if the call is done. Seems somewhat narrow in my view as this may cause some inadvertent hang-ups if one party needs to set the phone down (silence) to attend to or get something else from another room. I also question if MJ has an off-hook indication such that it emits some type of notification if there is no active call and the phone (in this case, not gateway) has a receiver not set in the cradle correctly (off-hook).

Did you have to do any firewall settings to get the MJ working? 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. 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. You can always switch the ports and see if desired. It really becomes a question of how you can use the GS settings to optimize the MJ line. You can use the line tests as well as the gain settings, but my understanding from others is that MJ is seemingly subject to a greater degree of variability than traditional copper, Catv phone, or even Sip trunking; at least this is what I hearfrom others I know who have MJ. Of course, this may also be a function of the end point as most of these folks are using it to call internationally - India, Mexico, etc. where the phone infrastructure is not nearly as good as some of the more developed parts of the world.