The screenshots/settings below were taken from a Panasonic KX-TDE100 with a GW16 card behind a standard consumer grade firewall. The configuration was done through the LAN. Note that by default Panasonic uses ports 16000 to N for RTP traffic. This has been modified in the example below, but you may wish to use their default ports. Starting at the main screen to configure the SIP card, you should see a screen similar to: • On the card you wish to configure, and select 'Shelf Property.' On the shelf property screen you will need to change you settings to match the image below. Be sure to enter the public IP address of the device or site depending on your needs for the 'NAT Fixed Global IP Address.'
We're staying on the subject of sip trunking today. This time we have installation guidance for SIP Trunks for FreePBX. This setup information and the screenshots were kindly provided by a customer who has documented his PBX setup with the telgo.SIP Connector. To help speed up your setup, we would. How to configure FreePBX with a Voys SIP Trunk This manual will help you set up your FreePBX server to work with a Voys SIP trunk in combination with a static IP address.
• Click 'OK,' the card configuration screen will now be in focus. And select 'Card Property' from the list. On the screen that has just opened select 'Common Settings' and change the settings to reflect your needs as seen in the image below. NOTE: You may or may not wish to use Google's DNS which is what is listed in the image: • Select 'OK' on the 'Common Settings' screen, and then 'OK' again on the 'Card Property' screen. You will now be returned to the card configuration screen.
Yu Gi Oh 5ds Episode 1 English Dub Download on this page. Right click on the card you are configuring and select 'Port Property'. You will be brought to the 'Main' tab. Please enter your settings in accordance with the image below, however, you must enter your SIPTRUNK.COM 'Trunk #' for a subscriber number. NOTE: DO NOT enter an IP Address unless you are using IP Authenitcation on your trunk. • Select the 'Account' tab and enter your SIPTRUNK.COM 'Trunk #' as the User Name and Authentication ID. Enter your trunk's 'Password' as the Authentication Password. This is represented by X's and Y's respectively in the image below.
• Select the 'NAT' tab. Enter settings exactly as in the image below: • Select the 'Voice/FAX' tab. Ensure that the 1st codec used is 'G.711Mu' as depicted in the image below: • Select 'OK.' The card should now be ready to be placed in service. Once in service, the trunk status should show as 'Registered' in your control panel.
Well, today I was cleaning up and found an old Linksys SPA-3102 device that I purchased 3-4 years ago. I originally purchased this device to connect my Trixbox (Asterisk) PBX to my land line at my house. The SPA-3102 is a device manufactured by Linksys/Cisco that provides one FXS terminal, and one FXO terminal. This device can connect your PSTN phone line to your VoIP PBX, and it can also allow you to connect a standard phone to your VoIP PBX as an extension, all at the same time. While I wasn’t to happy with performance of the solution, nonetheless I figured it out and got it running. I decided to write up a little blog post as a How-To get the SPA-3102 working with Trixbox.
This solution is mostly just a bunch of config, so excuse the lack of How-To and the bulk of config dumps: Update (2017): This also works on new versions of FreePBX Asterisk Linux Distro. I can confirm T.38 faxing works on the FXO line, but have not been able to get it working on the FXS PSTN line (fails to re-negotiate). 1) Configure the Asterisk extension (this configures the line you hook up to a phone on the SPA-3102): Create a extension inside of Trixbox or FreePBX. Leave everything default except: Display Name: Fax Machine (change this to whatever you want) Extension: 199 secret: password (choose you password) canreinvite: yes host: dynamic type: friend nat: yes qualify: yes 2) Configure the Asterisk Trunk for the SPA-3102 Go to the Trunk Menu inside of Trixbox or FreePBX PBX configuration. Add a new SIP Trunk. Leave settings default except: Outbound Caller ID: (Change the number to your PSTN line, if the number doesn’t match, it could break things) Trunk Name: spa3102 PEER Details: username=spa3102 type=friend secret=P4SSw0rdz (replace with your password) qualify=yes port=5062 nat=no host=dynamic dtmfmode=rfc2833 context=from-trunk canreinvite=yes 3) Configure Outbound and Inbound Routes The configuration for the Outbound route is normal and doesn’t require any special configuration other than the normal outbound route you’d normally create for a trunk.
However, the Inbound route does require special attention. When creating the Inbound route, make sure that the DID Number value exactly matches the 10 digit number you configure for the PSTN. This is how it will recognize this and categorize the incoming call under that specific inbound route.
4) Now for the SPA-3102 Configuration There’s no way I’m writing all the config out for the SPA-3102, so instead I took screenshots for each tab that requires configuration. You now have your SPA-3102 configured to both act as an extension and a gateway to the PSTN. If anyone has any better configuration please write a comment, I’d love to update this article, and I’d like to get this working better than it currently is of possible. One additional note: When the SPA3102 is factory reset, it’s default settings are optimized for the North America region. Posted by at 2:33 PM 10 Responses to “Configure Cisco/Linksys SPA3102 for Trixbox or FreePBX FXS and FXO” •. Good Morning Stephen! Great site!!!
I’m very new to Trixbox, started to play wit 2 days ago, after Callcentric has gone fro excellent free service to useless free( and also pay) service. I have several Spa3102 planted in EU and one here in the US. I would like to ask your help if doesn’t take up much of your time. I’ve tried to make as less changes on the setting form Callcentric to Trixbox, just in case I need to convert back. I was able to register all Spas to my box, but I’m getting one way or no audio on some of them. I see that you’re not using NAT on your settings Do you think that could be part of my issue?
I also have pst line plugged into Spa in EU. The settings on Spa allowing me to dial into this Spa from remote location and utilize the EU land line.
It also forward via pstn dial plan all incoming PSTN call to the Spa in US. But I experience disconnection (or just no audio after 15-20 sec.) I’d like to setup simultaneous ringing on some extensions can I forward call via sip uri to my US land line provider?
I’m not sure if you’re familiar wit IPkall, but I’d like to utilize that option too So I’m not trying any fancy stuff, pretty much just some basic features and I’d like to ask for your help. We could use remote login, if that helps Thanx for your time! File Bomba A Decompression Eliminare Bing more. Hello Attila, Thanks for the compliment on the site! Feedback is always appreciate!
First off, I have to apologize, I’m not firmiliar with Callcentric. And secondly, I’ll be honest, I’m not really a fan of the SPA devices, so I have to admit my knowledge is somewhat limited when it comes to these. However, it does sound like you issues are related to some type of network communication. First, I would make sure that Callcentric is only using UDP for SIP. I’m not sure if it does use TCP, but I’ve seen some other VoIP PBX software do this In my history, I’ve noticed that all issues that relate to one-way audio, or calls not being connected properly are usually always associated with one of these 3 things: 1) RTP – As far as I understand, this is what handles the voice data packets If for some reason these ports aren’t allowed (either through the VoIP PBX software, or the firewall itself), it can cause issues with connections, one-way audio, or even one-way calls. 2) Firewall/NAT – You need to make sure that all devices are aware that they are behind a firewall (only if they are). Keep in mind, if all these devices are behind the same NAT, then you shouldn’t have these issues I would try testing without NAT to see if this resolves the issue.
Unfortunately I know nothing about NAT configuration on CallCentric. 3) Codec misconfiguration – If by chance there is misconfiguration in the codec usage, this could cause issues with calls being dropped after 10-15 seconds since it can’t negotiate a proper codec. Also, sometimes it results in one-way calling since only 1 device can successfully negotiate the codec to be used. As for IPkall, I’m totally unfamiliar with If Callcentric is anything like Trixbox, you should be able to have it simultaneously ring all SPA’s by creating a ring group. On any incomming calls, it would just ring that ring group, which would ring all SPAs. Like I said this is easily done on trixbox, but I’m lost with Callcentric.
I hope this helps ðŸ™‚ Stephen. Stephen, thanx for your quick respond. I’m sorry for being unclear with my issue/request. I’m trying to get away from CC and want to register the devices to the hostname (myname.dyndns.org) of the Trixbox. The registration of all Spas went OK, I can see they all registering. The calls are going out and I’m able to dial in to the remote SPA in EU and to get the dial tone from the local PST line. So may biggest issue is the one way or no audio.
If I could manage that, I think I can somehow figure out the simultaneous ringing.