Asterisk with Analog SIP Trunk Gateway

I currently have broadband cable internet access and I want to setup Asterisk with a D-Link DVG-3004S Analog SIP Trunk Gateway and a Linksys WRTG54 router so I can terminate calls to the PSTN through my ITSP. I would like to know the best way to set this up and if I need any additional equipment. Also, I would like to know how to connect to a remote location (WAN, VLAN, etc) so calls can go back and forth. The specifications for the Gateway is on the link below.
Or can I simply have Asterisk setup also as a gateway and also use it to connect to a remote site? … 04S_ds.pdf


I’ve not used the DLINK product, but from reading it’s spec sheet, it sounds like it should be able to perform what you are trying to do without Asterisk.


With or without the gateway, can Asterisk performthese functions? And if so, what kind of configuration will you suggest?

Asterisk can do everything you are asking about without the gateway. It will require an analog PCI card from Digium.

You can get a TDM400P from Digium. They are good cards and are designed to work well with Asterisk. This will get you up to four analog phone lines to the PSTN or to analog stations.

The best way to go between two remote destinations when there are Asterisk boxes on both ends is to use the IAX protocol. It works the best around firewalls.

You can set Asterisk to connect to a ITSP through SIP.

Just all depends on what you require.


Thanx for the response. If I intend to go with Asterisk by itself, I am looking to implement a pure VoIP setup such that all I need is setup one or more SIP connections with my ITSP, then be able to also communicate with other remote gateways (Asterisk or other remote gateways). And this will be without purchasing any of the Digium cards since I don’t have to terminate to the PSTN locally, all connections are strictly through the internet with my ITSP and remote gateways. What recommendations do you have on this?

Pure VoIP is a pretty easy setup with Asterisk. If your ITSP supports it I would recommend using the IAX protocol. It works better around firewalls and has some bandwidth saving techniques. If your ITSP doesn’t support it, SIP works fine as well.

Choose you codec wisely and make sure you have enough bandwidth to support the number of VoIP lines you need.

Lastly do some latency test (ping / traceroute) to your ITSP first to make sure there isn’t too much latency between you and your ITSP.


I am currently running Fedora Core 4.0 and Asterisk 1.2.5 and I am trying to at least make this machine register with broadvoice to no avail. I would like to know what and how to configure asterisk to work with broadvoice on a strictly VoIP setup.

How do I change the sample voicepulse sonfig files back to the original config files in asterisk?

I have resolved the previous issues but now I get the error: NOTICE[30154]: chan_sip.c:5272 sip_reg_timeout: – Registration for ‘’ timed out, trying again (Attempt #3)
I would like to bresolve this.