Sip Trunk Setup Trix Box Setup

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Trix Box - VoIPtalk IAX Trunk Setup GuideThe VoIPtalk IAX service requires numbers to be sent with the full country code. For example, if you wish to call our office you would send the number in the following format (e164) to our call server:90. A rule can be setup to do this in the GUI. For Outgoing calls:Go to asterisk - FreePBX, then click Setup, and click Trunks.

I just bought a SPA-3000 from e-bay to play with and connected the FXO port to the MagicJack line. I am trying to setup a SIP trunk between SPA-3000 and TrixBox but I am not able to.I create a SIP trunk in TB with the following settings:Outgoing Settings:username=pstn1type=peersecret=pstn1qualify=yeshost=192.168.1.136port=5061dtmfmode=inbandAnd I set the following on the SPA-3000:Proxy: 192.168.1.128 (IP of TB)Register: yesMake Call Without Reg: yesAns Call Without Reg: yesUser ID: pstn1Password: pstnThe Registration Status still shows 'Failed'. Anyone know why this is. What other setting do I need to make this trunk register.I am able to make calls through the trunk if I reboot the SPA-3000 but after about a day I get the 'All trunks are busy now'. Another reboot fixes it for a wile longer but goes back to busy trunk message. Anyone know what other setting I need to get this to work?Thanks.

Outbound on Asterisk (FreePBX):username=2030type=friendsecret=qualify=yesport=5061nat=neverincominglimit=1host=dynamicdtmfmode=rfc2833disallow=allcontext=from-trunkcanreinvite=noallow=ulawUserID on your SPA needs to match the trunk number 'username'. I gave my SPA a SIP trunk of 2030 is asterisk, therefore the username is 2030.Also, your passwords ('secret') do not match.

You have 'pstn' and 'pstn1'.I remember that configuring the SPA 3000 with asterisk was a serious PITA. It took a lot of time to get it 'just right'. On a 'normal' Asterisk trunk, you can set it so Asterisk registers with the provider, as is required by Callcentric, VoIP.ms, etc. Or, you can set it to not register, e.g. Voxbeam, bandwidth.com, etc. There is no provision for the provider to register to Asterisk! Your 3000 is acting as a 'provider', so it's not surprising that it can't register to Asterisk.

Of course, the SPA doesn't accept registrations, either, so Asterisk can't register to the SPA.A peer can be set up to accept registration, yet still retain trunk characteristics. This is required, e.g. If you had a remote Asterisk on a dynamic IP as part of your system. PBXes.com has a 'Sub PBXes' feature that does this, you could set one up and go to Source View to see what would be needed in your config files. However, I don't see any benefit in doing that for your setup. IMO, in the 3000, just set Register to 'no' for the PSTN line and call it a day.

If you are really lucky, that will also fix your busy trunk issue and you'll be a happy camper.Otherwise, set up and test a softphone, ATA or IP phone so it can call out through the 3000. When you get the all trunks busy condition from Asterisk, see if you can still access the 3000 from the other device. If not, you'll know that the 3000 has the corrupted state - check the Info page or use its debug features to find the cause. OTOH, if the 3000 is still working fine, you'll know that the corrupted state is in Asterisk - with luck, you'll find some clue in its logs. Said by:I just bought a SPA-3000 from e-bay to play with and connected the FXO port to the MagicJack line. I am trying to setup a SIP trunk between SPA-3000 and TrixBox but I am not able to.I create a SIP trunk in TB with the following settings:Outgoing Settings:username=pstn1type=peersecret=pstn1qualify=yeshost=192.168.1.136port=5061dtmfmode=inbandAnd I set the following on the SPA-3000:Proxy: 192.168.1.128 (IP of TB)Register: yesMake Call Without Reg: yesAns Call Without Reg: yesUser ID: pstn1Password: pstnThe Registration Status still shows 'Failed'.

Anyone know why this is. What other setting do I need to make this trunk register.Why not just configure your SPA3K to register to your TB as an extension?

Thanks to everyone for your help. I finally got the trunk (extension?) to show registered on the SPA3K. I had to make the trunk name (2030) the same as the username in the outbound settings of my trunk.But now I have the same issue. Yesterday I was able to make test calls through the SPA3K but today I tried to make a call and I get 'all circuits are busy now.' I checked the SPA3K info page and it still shows as registered. I checked the TrixBox status page and it doesn't show any trunks as being active.

Power cycling the SPA3K resets the trunk somehow and I am able to make calls again. Anyone know why this is?Thanks. Said by:I checked the SPA3K info page and it still shows as registered. I checked the TrixBox status page and it doesn't show any trunks as being active.

Power cycling the SPA3K resets the trunk somehow and I am able to make calls again. Anyone know why this is?I am just curious.

Sip

After you powered cycle your SPA3K, does your TrixBox status page show any trunks as being active? If so, then that explains why you get the all circuits are busy now message. Perhaps you may wan to add qualify=yes to your extension (just a wild guess). Stewart,I don't know what you mean by 'could you reach it from another device?' When the trunk goes into the 'busy' state I am able to ping the SPA3K.

Said by:I don't know what you mean by 'could you reach it from another device?' .You would set up another VoIP device, such as a spare line button on your Polycom, to point directly at 192.168.1.136 port 5061 (and not register). This would allow you to call out via the SPA, independently of Trixbox, which would provide a quick test of whether the problem was on the Trixbox side or the SPA side.However, that's probably all moot now, because we know that the SPA is rejecting the call with a 504 error. I suspect that SPA and MJ are not configured compatibly electrically; unfortunately you didn't record line voltage and loop current to confirm.Try setting Line-In-Use Voltage to 15, and also connect an analog phone to MJ (in addition to the SPA, using a splitter). If the settings change doesn't fix the problem, when a failure occurs, report line voltage and loop current, and pick up the phone to confirm that MJ is still functional.

Sip Trunk Setup Trix Box Setup

I have not yet hooked up another phone in parallel but I changed the 'Line-in-use' voltage to 15 yesterday morning. I was able to make calls all day and I thought my problem was solved but I tried today morning and I got the 'All circuits are busy.'

Said by:PSTN Line StatusHook State:OnLine Voltage:-6 (V)Loop Current:15.4 (mA)This seems like an impossible combination - If it's on hook the current should be near zero, even if the MJ were messed up and providing an incorrect voltage.Possibly, it's an SPA firmware bug and a newer (or conceivably, an older) version will fix it. Though unlikely, reversing the polarity might help (if you have a crimp tool and RJ11 plugs, make a cord with one end upside down).It may be worth looking at how it gets into the bad state - did the previous call end incorrectly, or did some event (e.g. MJ PC rebooted) trigger it?When in the bad state, see if unplugging the Line jack on the SPA for a few seconds will get it right again (without a reboot). If so, possibly a restart of the MJ app will get it going.The report itself may be invalid - check what is shown for voltage, current and hook state, when properly idle, while on a call, and after a call has ended properly, i.e.

Sip Trunk Setup Trix Box Setup

When a new call would work.If all else fails, you might have a script periodically look for the trouble and reboot the ATA when needed.You could also just track down how to make Asterisk work with the MJ account, so the SPA and MJ wouldn't be needed at all;). I tried unplugging the telephone cable when the system got into the funny state but that did not help. I rebooted the system and see that while on hook the line voltage is -51V and loop current is 0.0mA.

Loop current is 15.4 mA and line voltage is -5V when I get the 'circuits busy' message. I am running MJ under a virtual machine (VMware Workstation) and I can't think of anything that might cause this.

I will try moving MJ to real machine and see if that is any better.I used to register directly into MJ without using the dongle but MJ did something to make that not work. Has that been gotten around? Is it now possible to connect to MJ without a dongle?

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Any additional info on that would be greatly appreciated.Thanks. Just for shits and grins, do you have 'real' dialtone at your house from a POTS provider (phone or cable)?If so, is there any way you could hook this up to the real dialtone and see how long it holds?As mentioned, there are MJ hacks for asterisk, so you could eliminate the 'middlemen'.I'd have to agree that the MJ is probably the culprit. I've never used a MJ, is there any special dialing? Are you sending the right dial plan out through your ATA?The V/A specs you have posted are simply an 'off hook' condition. If you send a bad dial patters to the MJ, it's probably going to send back an SIT intercept to the ATA.

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The ATA doesn't know what to do with it. Therefore asterisk doesn't know what to do with it and gives you a generic ACB intercept message. IMO, the first order problem is with the SPA, though the MJ probably is doing something non-standard to provoke it. New english file beginner student's book download pdf. Possibly, newer firmware will help; there is a link to 3.1.20 in ยป.If you can determine when it gets into the bad state, that may be a clue to a workaround.As mentioned earlier, you could automate something that detects the trouble and reboots the SPA.A different FXO device such as Obi, Zoom, or even an SPA3102 may work better.

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If your TB runs on a desktop, you could try an X100P clone.Personally, I gave up on MJ when they made one too many anti-SIP moves, and don't even know which side won the latest battle in the cat-and-mouse game. There are several similar services (netTALK, GV) that are more Asterisk-friendly. A provider with low rates (Voxbeam, Localphone) or inexpensive 'unlimited' (VOIPo, Phone Power) might meet your needs, or spend a bit more for something really good that won't waste a lot of your time(Callcentric, Vitelity, etc.).