Configuring Grandstream GXW-4104
dicembre 16th. News December 16th. 2007, 12:12 amIn the two units so far I've set up, the starting point was to update the firmware. The easiest way to do this is to download the latest firmware from the appropriate page of the site GrandStream : the latest available at the time of writing this article is 1.0.1.2.
We will make sure that you have a webserver which make our unity GXW-4104 can download the firmware. We'll have to unpack the zip with the firmware in a subdirectory of your webserver, for example in a folder called "firmware /". Assuming that the server responds to the IP 192.168.0.1, and opening a browser, we must ensure that - by going to http://192.168.0.1/firmware is displayed a list of the contents of the folder extracted previously, namely:
- boot64.bin
- boot64a.bin
- gxw4100.bin
- load64.bin
Without these basic steps, we will configure our unit so that it can take the firmware from our web server. By accessing the administration panel via our trusty browser notification, we will:
- Activate the upgrade through HTTP protocol;
- Set the path of the firmware ("Firmware Server Path" should be filled with the value - following the example given above - "http://192.168.0.1/firmware")
- Enable auditing for the upgrade (Always check for New Firmware) and the date by which the check must be made ("check for upgrade every"): the lowest possible value is 60 minutes.
Without this, after 60 minutes set the firmware will be downloaded from the web and the next time unit we find the latest release of firmware you have time to grab a coffee and read the latest news on voip-PBX ![]()
After updating the firmware, do not forget to check by "Status" that the version of firmware is actually equivalent to that discharged.
Without the upgrade, we can do the configuration itself. We list the first steps:
- Set the FXO ports and parameters of the PSTN tones
- Set the SIP Server
- Configure Asterisk
The configuration shown in this mini-tutorial aims to help you:
- receive calls on any port FXO and dispatch them - all - into a single sip account
- make calls using the PSTN directly from your SIP phones
1. Setting FXO ports
In "FXO Lines":
Under "FXO Termination"
- Enable Current Disconnect: Yes
- Enable Tone Disconnect: Yes
- Enable Polarity Reversal: No
- AC Termination Impedance: 270 Ohm + (750 Ohm | | 150 nF) and 275 Ohm + (780 Ohm | | 150 nF)
- Unconditional Call Forward to VOIP:
- Username: ch1-4: 111; (It means that for channels 1 to 4 calls are redirected all'extension SIP/111 the sip server specified for each channel)
- SIP Server: ch1-4: p1 (It means that for channels 1 to 4 should be used to configure the sip server specified in p1 = Profile number 1)
- SIP Destination Port: ch1-4: 5060; (It means that for channels 1 to 4, the sip server is on port 5060)
In "Channel Dialing":
- Wait for Dial-Tone (Y / N): ch1-4: N;
- Stage Method (1/2): ch1-4: 1;
In the "Channels":
Under "Call Progress Tones":
- Dial Tone: ch1-4: f1 = 425 @ -14, f2 = 425 @ -14, c = 20/20-60/100;
- Ringback Tone: ch1-4: f1 = 425 @ -14, f2 = 425 @ -14, c = 100/400;
- Busy Tone: ch1-4: f1 = 425 @ -14, f2 = 425 @ -14, c = 20/20-20/20;
- Reorder Tone: ch1-4: f1 = 425 @ -14, f2 = 425 @ -14, c = 20/20-20/20;
- Confirmation Tone: ch1-4: f1 = 425 @ -14, f2 = 425 @ -14, c = 20/20-20/20
Under "Specific Channel Setting":
- DTMF Methods (1-7): ch1-4: 2;
2. Set the SIP Server
We will use a single profile, under "Profile 1", set the IP of your Asterisk server is in "SIP Server" in "Outbound Proxy".
3. Configuring Asterisk
Edit the file sip.conf:
[gxw410x]
type=peer
context=from-grandstream
host=ip_di_asterisk
insecure=port
dtmfmode=rfc2833
[111]
type=friend
secret=111
qualify=yes
nat=no
host=dynamic
canreinvite=no
context=gxw410x
Edit the extensions.conf file and add a rule for outgoing calls to the GXW 4104:
[grandstream]
exten => _0X.,1,Dial(SIP/${EXTEN:1}@gxw410x,30,r)
At this point you just have to modify your dialplan to handle incoming calls according to your needs.
This guide is not exhaustive and is an indication of the main steps you should take a basic configuration of this product. Possible configurations are much more advanced, but I leave this task to you: When you're familiar with the structure of the configuration panel, everything should become easier!
Tags: Call Center Systems | VoIP PBX | Asterisk Consultant Naples | PBX Phone | VoIP | Asterisk CTI | PBX | IP Phones | Networking | Linux
Development of IVR systems, call center, VoIP PBX.


This is certainly the only guide on the internet that runs GXW4104 with asterisk in Italy. Should be put on the official website. Congratulations.
1000 Thanks for this great help.
Do not mention it. I have some news on the configurations of this system, especially with Elastix. If you need, let me know.
I'm configuring a anchio GXW4104 and the beginner.

I followed this guide to the basic configuration, but some things have changed in the latest version of the firmware ...
The latest news you are referring to are related to the latest firmware?
What are they?
Thanks very much and congratulations again for your help
Actually I was referring to the new configuration options with regard to the fact that you can forward calls to each channel to a different inbound route.
If you need some examples or help, please feel free to contact me with the email form.
Hello!
I am trying to configure GXW 4108, if I understand it should be a 4104 but with 8 FXO ports ...
Unfortunately I just can not configure it, the main problem is I do not see the line. In the "Status" section of the configuration panel, the "Connected FXO Line" shows all the lines as "NO". I tried different ports FXO, but without success.
So I wanted to ask you: the Grandstream, led lights up the corresponding port FXO only transfer data, or just plug the RJ-11 to make it light up? Why I never lit any of the LED on the FXO ports. I do not know what to do ...
I do not have the unit in front of me right now, but I'm pretty sure that lights the door only when transferring data.
What matters is that you say "FXO Line Connected: Yes"
I suggest you follow the guide step by step and check that the registration be correctly between the server and GXW asterisk.
If you need more help let me know
I solved attacandomi to another phone jack in the house ...
strange ...
Hello, I'm configuring a GXW 4108 connected to Asterisk, I would ask: what is the differreza between one stage and two stage? Thanks Max
Hello
I'm setting up a Grandstream 4108 I say that everything works the only thing I can not figure out what I have not touched me to see the numbers come all the incoming unknown despite the id there is where I check to solve the problem??
Thanks
Try to see under FXO Lines -> PSTN to VOIP Caller ID Setting There are two parameters:
1. Caller ID Scheme: Bellcore uses (1) or etsi ring (2)
2. Caller ID Transport Type: Relay via SIP From (1)
Hello, how can I tell which lines to use 4108 x call?
I have the need 'to divert calls to mobile phones to a specific port ...
Thanks
ps = congratulations for the guidance!
Try looking under the Channels tab, under "Port Scheduling Scheme (VoIP-> PSTN)" you should find an option "Prefix to Specify Port (1 stage dialing method):" which is normally set to "99"
It means that if you post to the GWX made a number like this:
99 + port + number to call (Ex. 99 1 02123123)
this number will be dialed on that port.
I usually, in FreePBX, trunks for each port of call.
Hello!
GREAT!!
hours works great!
Thank you!
So you could
I am very pleased!
Hi, I'm new to this system. I built a test PBX Asterisk, FreePBX and CentOS so with OS. I wanted to ask where or how to insert the values described in the GUI for Asterisk FreePBX.
I tried the guide but I can only receive calls and to send them.
Thanks in advance for your reply.
Hello James,
Gxw4104/4108 guide you refer to?
Hello,
I bought a gxw4104, I was able to receive calls on an extension from the outside, thanks to your guide. but do not go out with FreePBX. He tells me that probably all lines are busy.
I may not have properly configured your pbx, I refer to section 3 of the guide.
I insert in FreePBX, the values in the sip trunk that I created with the name gxw410x in this order:
in the outgoing settings - PEER details:
type = peer
context = from-Grandstream
host = ip_di_asterisk
insecure = port
dtmfmode = rfc2833
in incoming settings - USER details:
type = friend
secret = 111
qualify = yes
nat = no
host = dynamic
canreinvite = no
context = gxw410x
Then I inserted the code in extensions.conf as well as written below:
[Grandstream]
exten => _0X., 1, Dial (SIP / $ {EXTEN: 1} @ gxw410x, 30, r)
I hope I did understand. Thank you.
Hello,
If you use FreePBX is not necessary to hand the dialplan.
Create a new trunk with these parameters:
Outbound dial prefix: 99
Trunk Name: gxw4104
Peer details:
type=peerqualify=yes
insecure=port
host=ip_del_gxw4104
dtmfmode=rcf2833
nat=yes
canreinvite=no
Then vai outbound routes and navigate a route that uses the output trunk.
Eye to what has already replied to Andrea on 30. October 2011.
If you need, write well
Thanks for the info. now works (but I did not understand why!)
last night did not work despite your configurations, I now access the PBX and voila, outgoing calls working.
conversely, no longer receiving incoming calls ... official!
The GXW is configured to send calls in 101 in:
Channel Dialing to VoIP => 1. Unconditional Call Forward: => User ID: ch1-4: 101;
in the trunk that I created with the name I entered in ch2_gxw4104 Incoming Settings:
USER Context: 101
USER Details:
type = friend
allow = all
context = incoming
host = IP address of the GXW
username = 101
secret = password
and create an Incoming Route - all DID / any CID that the tip extension 101.
nothing to do but does not respond to calls coming GXW.
it off?
Thanks again for your responses. Happy Holidays
Resolved, riaccendeno off and the pc. Is this normal?
I'm still here to humbly ask for help.
ever since I gxw4104 that calls are answered by IVR from ch1.
I have the trunk, I inbond route and IVR. but does not answer when I call.
it off?
Hello,
You need to tell the PBX that must divert incoming calls to 'IVR
Hi, I have a problem with the gxw4101, you can not make outgoing calls.
If I restart the GXW call me first, then the successive FreePBX tells me that the lines may be engaged.
What can be the problem?
the configuration is like the one above.
Thanks in advance ...
I ricontrollerei part FXO Termination and Call Progress Tones. From what you write I have the impression that the channels remain open after the first call.
Hello!
Ok, ricontrollero 'as soon as possible. through 1000. In fact checking the status page of the GXW when I close the call takes a long time to return to Ilde.
If you can give me more details I would be grateful. Thanks
Hi, iparametri are configured as described in the guide to the top.
I just do not find this setting: Confirmation Tone: ch1-4: f1 = 425 @ -14, f2 = 425 @ -14, c = 20/20-20/20
I thought it was removed from the current firmware GXW. (1.3.4.10)
I do not know where to turn. help, thanks
on my (who has the same FirmWare of yours) I find him under that parameter Channels / Call Progress Tones / Voice 4: Tone Recorder ...
Hi, I come to ask for help.
In exchange I could run in part I have configured the trunk sip to GXW4104 and use them with a line pots / RTG and with 2 analog outputs of a stud NT1 + ISDN mono number configured to pass the first call to the port / b1 occupied and if the first door, pass it to a/b2.
I cofigurato an IVR, and it works. the only problem is finding that when the second call comes dall'IDSN (then the second port is already occupied when the first) astrerisk unresponsive. I say this because doing the tests I realized that I use as gateway GXW4104 me in the status window indicates that the call arrives. It seems as if the GXW not send the call to asterirsk or asterisk will not accept the incoming call.
which may be the reason? how can I fix it?
Seat configurations:
GXW4104:
FXO Line
FXO termination
2. Enable Tone Disconnect (Y / N): Y
7. AC Termination Impedance: ch1-4: 2;
Channel to PSTN Dialing
1. Wait for Dial-Tone (Y / N): Y
2. Stage Method (1/2): ch1-4: 1;
Dialing to VoIP Channel
1. Unconditional Call Forward:
User ID: ch1: 1001; ch2: 1002; ch3: 1003; ch4: 1004;
SIP Server: @ ch1-4: p1;
PSTN to VOIP Caller ID Setting
1. Number of Rings Before Pickup: ch1-4: 1;
2. Caller ID Scheme: ch1-4: 8;
Chanels
Phone Number Settings
Channel (s) SIP User ID Authenticate ID authen Password Profile ID
1. 1 1001 1001 1 pass1
2. 2 1002 1002 1 pass2
3. 3 1003 1003 1 pass3
4. 4 1004 1004 1 pass4
Call Progress Tones
1. Dial Tone: ch1-4: f1 = 425 @ -12, f2 = 425 @ -12, c = 20/20-60/100;
2. Ringback Tone: ch1-4: f1 = 425 @ -12, f2 = 425 @ -12, c = 100/400;
3. Busy Tone: ch1-4: f1 = 425 @ -12, f2 = 425 @ -12, c = 50/50;
4. Reorder Tone: ch1-4: f1 = 425 @ -12, f2 = 425 @ -12, c = 20/20;
FreePBX: (all 4 trunks are the same conf only change the parameters for the 4-port)
Port gxw4104: FXO1
Name Beam: 1001
Outbound CallerID:
Maximum number of channels: 1
Prefix dialing out: 991
Settings on Exit
Name beam: 1001
PEER Details:
context = from-trunk
host = dynamic
username = 1001
secret = _Password_ (set in GXW4104)
type = friend
dtmfmode = rcf2833
hope you got all the details, I tried to get help in other forums with no response. (Since you are now the only answer and that there are very grateful)
Thank you in advance. James
Hello James,
the status window of the GCW is a little lacking in details. I controllrei logs from asterisk console for additional information: I would try to make two calls and collect the logs. In this way you can understand, from the side of the PBX, what happens.
I'll also note something in the config GXW:
1. Unconditional Call Forward:
User ID: ch1: 1001; ch2: 1002; ch3: 1003; ch4: 1004;
Then
channel 1 will be sent to 1001 @ ip_del_tuo_server, channel 2 is sent to 1002 @ ip_del_tuo_server etc.
Assuming you have a context of inbound configuration in your Asterisk IVR that plays on the 1001, you may try to change that parameter in this way:
User ID: ch1-2: 1001; ch3: 1003; ch4: 1004;
Let me know!
Hello, Thanks for reply.
As I am using Asterisk 1.7.1 distribution. The logs say that I see them going to the CLI?
If yes, I made two calls, but the second there is the shadow in the CLI. displays only the first.
How do I create an instance of all inbound in FreePBX that targets the IVR. The conf 'uncondizional call forward' is so for all the separate channels. is wrong?
Thank you. James
Hello admin, I followed your instructions. That I see in the logs by opening terminal and entering the CLI centos there any indication of the arrival of the second call by ISDN. When you unplug the GXW the first line of ISDN and deal with a traditional phone channel works.
Confiugurando that parameter as you suggested, the trunk in FreePBX not. Respond to any call.
I set the only route in FreePBX Inbound IVR to point. Help, thanks.
PS can be a problem. The GXW?
Hello Again I
I doubt it is built ...
I connected the two ends to the GXW Analog ISDN line and a line of pots / RTG normal, all telecom. Since the problems in receiving calls I have with these two ends, is not that perhaps the gxw4104 and ISDN for some reason are not compatible?
With the line RTG I had no problem ....
I await news, thanks. James
Hello,
Carefully recheck the call progress tone of your configuration:
Your config, covering at least the one you posted is:
That indicated in my post:
Hello,
I replied to your post on January 4. Let me know. Anyway I do not think there is any compatibility problem.
Hello, I changed the values of call progess tone. I can not find field 'Confirmation tone'.
How is it possible?
Meanwhile, through 1000, as always. James
Perhaps we have two different firmware versions. In truth it is quite a bit that I do not update. What version you have?
The latest version: 1.3.4.10 Program-Loader Boot-1.1.3.4-1.1.3.2
I reconfigured as per your instructions and everything I have done some basic tests, two cell types to call the same number of ISDN. the result is, is what I call one line from the outside, is what I call the two lines almost contenporanea (with a slight delay due to the config. the "ISDN to which I respond to first line 1 and the second line 2). The GXW I respond well to the first test, the second test does not get calls in FreePBX.
From the status window of the GXW I get calls but then, as if they stop inside the GXW (from cell. Contiuna the dial tone, no answer).
If I restart the GXW starts from the beginning the first call on all OK, the second thing.
As regards the line RTG completely regular. Call, answer every time.
What could it be? perhaps the ISDN line? I do not know what to do ....
Thanks for your patience. Regards James