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The use of D-Link products DPH-140S and DVG-6004S together with Asterisk SIP Server and Skype SIP Gateway – PE (referred here as ssgwpe) is reviewed here.

The D-Link DPH-140S Business IP Phone is a full-featured, cost-effective, standard-compliant telephone that can be easily plugged to your home or office network via Ethernet cable.

The D-Link DVG-6004S VOIP trunk gateways connect an IP network to POTS (plain old telephone service) analog lines to send IP voice and data to the conventional phone sets and fax machines. Designed as a cost-effective VoIP solution for a small to medium-sized business and the branch office, these devices provide 4 FXO (Foreign eXchange Office) ports for 4 phone line connections. IP network connection can be through any of the 4 LAN ports or the WAN port (all 10/100BASE-TX Ethernet interfaces).

DVG-6004S has to work in concert with a SIP Server. For this review, the SIP Server is Asterisk. In my next review, Asterisk shall be replaced by another D-Link product, namely, DVX-1000.

DVG-6004S provides many other features beyond the 4 PSTN FXO ports. It acts as a router and has a WAN port that can be used to connect to the internet. It also provides 4 LAN ports that other IP based products can connect to it.

The simplest network setup to get everything going can be as follows:

Connect the analog telephone lines to FXO ports, the internet modem to WAN port and the rest to the LAN ports. With appropriate configurations of the various devices and servers, the above should work. I have to add that I did not try out this configuration because I already have an existing network and decided against reconfiguring my networks.

Hence, my network diagram is a little more complicated and it can be represented as follows:

In the above diagram, DVG-6004S functions purely as a FXO trunk gateway. The router in the middle provides all the other connectivity.

Using the above configuration, I had the following working:

• Register SIP phone (DPH-140S) to Asterisk
• Make Skype call from internet to SIP phone
• Make SIP phone call to Skype
• Register DVG-6004S to Asterisk
• Make outgoing calls to PSTN via DVG-6004S
• Receive incoming calls from PSTN via DVG-6004S
• Make Skype call from internet to PSTN line via DVG-6004S

The general impression is that the D-Link products are relatively easy to configure. The manual for DVG-6004S is a little hard to understand but with good support from D-Link, I managed to get things working well.

There is no work done on the fax features provided by DVG-6004S. This may be covered in the next review, together with DVX-1000.

Now for the details.

Connecting DVG-6004S

This description is for network diagram 2 above. The following assumptions are made:

• The router to internet is properly setup and have access to the internet. The router shall also provide DHCP service.
• The PC with Asterisk and ssgwpe is properly setup and connected to the router. For information about setting up ssgwpe, please click here.

Do the following steps:

• Connect DVG-6004S via its WAN interface to the router.
• Connect a notebook or PC to LAN interface of DVG-6004S to access its web interface.
• Open browser and point to 192.168.8.254. This is the default LAN IP address for DVG-6004S
• Select “Network Settings” and set “Network Settings (WAN)” to use “DHCP” or “Static IP”. If you choose the latter, ensure that this “Static IP” is out of the range of DHCP client addresses provided by the router.
• Click “Accept”
• Select “System Operation”, check “Save Settings” and “Restart” followed by clicking “Accept”. DVG-6004S will reboot with the new settings.
• Once DVG-6004S has rebooted, go back into “Network Settings” and note the IP address that is assigned by the router (if DHCP was choosen earlier).
• Ping router. This should work.
• Move the notebook or PC to the LAN ports provided by router.
• Ping DVG-6004S via the assigned IP address from router. This should work
• Connect to DVG-6004S web interface via the assigned IP address from the router.

Register SIP phone (DPH-140S) As Client to Asterisk

• Connect DPH-140S to one of the LAN port of router. You may also choose to connect to LAN port of DVG-6004S.
• Create a SIP extension in Asterisk to be assigned to DPH-140S.
• From the LCD display on DPH-140, find the assigned IP address to DPH-140S.
  • Press Menu.
  • Scroll Down until “IP Address”.
  • Note the address.
• Let’s say the IP address in 192.168.2.22. Now point web browser to 192.168.2.22:9999.
• Just press “OK” for authentication. Default is no username and no password. You can choose to change this from the web interface.
• You should see the web interface to DPH-140S.
• Click “SIP Settings” and enter “Registrar Server” values corresponding to IP address of Asterisk. Click “Submit”
• Click “SIP Account Settings” and enter “Account 1 Setting” values corresponding to the created SIP extension in Asterisk. Enable this account. Click “Submit”.
• Click “Restart System”. Allow the phone to reboot and allow it to register to Asterisk server.
• If properly registered, you should see the extension for the phone on the LCD display of the phone.
• Should the above process not work correctly, check network connectivity, and also connect to Asterisk to monitor what is going on.
• If the phone is properly registered, you should be able to make calls according to the dial plans created under Asterisk.

Skype call from internet to SIP phone (DPH-140S)

For ease in describing the process, let’s assume the following:

• The SIP phone is registered as client to Asterisk as sip:101@192.168.1.99.
• The Skype caller from the internet is “tester”.
• The Skype client associated to the ssgwpe is “sgw”.
• Configuration for ssgwpe is done correctly as per “Installation Guide” and is registered as client to Asterisk.

Do the following steps:

• Using the “From Skype” tab of ssgwpe, create a “test” group.
• Add “tester” as buddy on the Skype client “sgw”.
• Press “Refresh” button under “From Skype” tab of ssgwpe.
• Add “tester” to “test” group.
• Set the “Number to call for the group:” to sip:101@192.168.1.99, as per prior assumption.
• Login as “tester” from a Skype client connected to the internet.
• Call from “tester” to “sgw”.
• Your skype call should be routed to the SIP phone.

SIP phone call from DPH-140S to Skype

Extending the assumptions to the prior section:

• ssgwpe is listening on 192.168.1.98 as a SIP server. Again see “Installation Guide” on how this is done.

Do the following steps:

• Connect to web interface of SIP phone, DPH-140S.
• Click “Phone Book”, add in name to correspond to Skype caller “tester”. For example, “Name” can be set as “Skype Tester” and “Number” set as “tester@192.168.1.98″
• Click “New” to add this to the phone book for the SIP phone.
• Press “Phone Book” on SIP phone and scroll down to select “Skype Tester”.
• Press”OK” to dial to Skype client “tester”.
• If configuration is done correctly, you should see ssgwpe picking up the call and then connecting the call to Skype client “tester”.

Note:

• As described under “Known Problems“, despite proper connection, only one way sound may be heard. To rectify this, you can choose to have Asterisk become the bridge between the SIP phone and ssgwpe.
• For example, create a tester@192.168.1.99 (Asterisk) which forward calls to tester@192.168.1.98 (ssgwpe).
• For those familiar with Asterisk, this should be easily done. I used the interface from VoiceOne allowing me to write macros to support one extension to handle all Skype users.
• Edit the phone book for the SIP phone to reflect the changes and then try connection again.

Register DVG-6004S as Client to Asterisk

In this setup, DVG-6004S is used simply as a PSTN gateway.

• Connect to DVG-6004S web interface via the assigned IP address from the router.
• Click “SIP”. For each FXO lines, set each up as SIP client to Asterisk. This is similar to the process done for SIP phone earlier. Ensure that the same SIP extensions are defined in Asterisk.
• Check “Enable Support of SIP Proxy Server / Soft Switch” and “Enable SIP Proxy 1″.
• Enter Asterisk IP address to “Proxy Server IP / Domain”.
• Click “Accept”.
• Click “System Operation”. Check “Save Settings” and “Restart”.
• Click “Accept” to reboot DVG-6004S.
• Upon rebooting, DVG-6004S should be registered as SIP clients to Asterisk.
• Connect to Asterisk to ensure this is done correctly.

Making outgoing call to PSTN via DVG-6004S

When the prior setup is done correctly, making outgoing PSTN call is easy. Assuming FXO Line 1 is registered as 701@192.168.1.99 (Asterisk), then dialing 701 on the SIP phone establishes a PSTN line out with a dial tone.

This manner of calling, getting a line out followed by actual number dialing, is a little clumsy. This can be improved in two possible ways:

1. Program the “Line” buttons on the SIP phone to get the first free line out. This must be done together with proper settings for DVG-6004.
2. Have Asterisk to support number mapping. For example, dialing 912345678, means get a line out followed by dialing 12345678. This is easily done using this “exten = s,1,Dial(SIP/701,20,gtTwWmD(ww${ARG1:1}))“. You can choose to do something similar to this.

Receive incoming call from PSTN via DVG-6004S

• Define a SIP extension on Asterisk to receive incoming calls from DVG-6004S. Let say this extension is *33@192.168.1.99. What Asterisk does with this extension is up to Asterisk. *33 can be directed to a SIP phone, to “incoming queue”, to “incoming group” or to whatever dialplan defined by Asterisk.
• Connect to DVG-6004S web interface via the assigned IP address from the router.
• Click “Telephony Settings”, enter “*33″ for “Hot Line No.” for the FXO Lines.
• Click “Accept” followed by Save and Reboot.

Make Skype call from internet to PSTN line via DVG-6004S

To extend Skype call to PSTN line via DVG-6004S, a number of steps are needed. The brief details are as follows:

• Group Skype callers under ssgwpe that are allowed to make PSTN connection.
• Create an extension in Asterisk to receive Skype calls from this group.
• Map this extension to the group under ssgwpe.
• Define the extension in Asterisk to do the following:
  • Use interactive voice response to Skype caller for line number to dial out.
  • Bridge the Skype call to line out via DVG-6004S.

Summary

Skype Sip Gateway (PE) works well in the various test cases. However, there is no dial tone during connection from SIP to Skype. A user may be mistaken that a connection is not taking place. This should be improved.

DPH-140S is easy to setup and use. The phone book is especially useful for setting up “non-numeric” call numbers as in the case for calling Skype users. It would be good if phone numbers can also be added via the phone interface, akin to those of mobile phones.

DPH-140S can handle up to 4 SIP accounts making it possible for, say, one connection to a local IP PBX and another to an internet SIP account. The latter case was not tested (This is incorrect. All 4 Sip accounts must be to the same SIP server).

When speaking via the handset, it is best to speak directly into the microphone. Called parties have reported difficulty in hearing when the microphone is displaced slightly. As for the calling end, there is a noticeable low level of noise, albeit not too uncomfortable.

When used with Asterisk, DVG-6004S is a good alternative to Zaptel analog cards. Once DVG-6004S is correctly setup, it performed very well.

There are two noticeable advantages on the use of DVG-6004S over Zaptel analog cards:

• There is no “echo cancellation” problem with DVG-6004S. With Zaptel analog cards, I have to use “open source line echo canceller (oslec)” to solve the problem. This involves patching and recompiling Asterisk.
• With Zaptel analog cards, there is a problem detecting dropped call during start of voicemail prompt from Asterisk. When this happens, PSTN line can only be freed after the voicemail service runs its course. This can take some time. With DVG-6004S, this problem does not happen. DVG is able to detect dropped call correctly and can initiate proper call termination to Asterisk.

Some issues do crop up with the use of DVG-6004S.

• One situation has calls from DVG-6004S to DPH-140S resulting in failure and many warning messages on Asterisk. Strangely, when the dial plan in Asterisk is changed to a group (one is DPH-140S and the other X-Lite), the problem disappears. Setting “a-law” as preferred codec in DVG-6004S also solve the problem.
• Despite trying various settings and firmware updates to DVG-6004S, there is no success passing PSTN incoming caller ID to the SIP phone.
• Firmware update to DVG-6004S takes a different track. One does not upload a file to DVG-6004S. Instead, DVG is a client to TFTP, FTP or HTTP service. Though different from normal, it can be useful in cases where centralized repository of updates are preferred.

DVG-6004S is an all-in-one device offering LAN, WAN and PSTN connectivity. As a suggestion, if may be good if D-Link can offer a simple PSTN gateway with 1 LAN port for network connectivity. This can appeal to many with already existing networks.

This entry was posted on Wednesday, November 7th, 2007 at 2:41 am and is filed under Articles. You can follow responses to this entry through the Comments RSS Feed. You can leave a response or trackback from your site.