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Grandstream HT503 ATA Configuration with Asterisk FreePBX
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- Опубліковано 13 бер 2019
- This is a long and boring video walking through the setup process for configuring an HT503 to work with a Raspberry PI running Asterisk FreePBX. The was prepared for Amateur radio operators.
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You, my man are a life saver.. I've spent hours and hours before finding this video and finally managed to configure my grandstream ht813.. great tutorial..
You are welcome!
Excuse me, do you have a problem with a call back from the Gradsteam HT813? I have this problem but don't know how to fix it yet
Your video helped immensely getting a Grandstream HT813 (the replacement for the discontinued HT503) working with a POTS line and my RasPBX. (I know what your AREDN is as I've got an FCC ARS license.) I'm simply coming in the FXO port to allow inbound calls to be answered on my PBX system - and it provdes caller ID in numerous other locations around my house. One added wrinkle is having a PSTN phone connected to the FXS port which now has an extension number and I'm still tweaking that to get it working exactly the way I'd like it to. Your setup with just the Grandstream GXP1625 doesn't need the FXS port. I've got two PSTN lines and will be adding the second one to another HT813. My setup isn't identical but yours got me close enough to get it running. Looked at FXO gateways and they're made for commercial/business PBX systems - and they're comparatively costly.
Setup in the HT813 is not quite identical to the HT503 as there are more settings, but it parallels what you did quite closely. Passing this on to you if you need to work up another system like your AREDN and cannot find an HT503. The HT813 should do the job. BTW, I've got a GXP1625, GXP1628, and now a GXP1782, GXP1760W, and a GXP2170 all running on the RasPBX with four DIDs on a VoIP provider SIP. A second HT213 will unify all six phone numbers on my VoIP system and maintain the pair of PSTN lines with some PSTN 2-line phones and a FAX machine. Had to make certain the POTS trunk was the only one tied to the POTs line.
I had help setting this up and don't understand much of it.
hello.. i plan to buy HT813. i'm using ONT modem and has RJ11 output to my analog phone. can i connect that RJ11 port on the ONT into FXO port on HT813 and using raspberry (asterisk) on same network to make a system where i can call or receive my landline number from smartphone using zoiper app? as u also using HT813, would u please give me your configuration?
Thank you for making this; I've had a HT503 hooked up to my freepbx for a while for just messing around and while I could get incoming calls to work, I never could get outbound to work properly. This did the trick; thanks for burning a trail for the rest of us!
Glad it helped. I understand your frustration - I messed around with this unit for over a week before getting help.
Hello ! Thanks for the video. its full instructions served me very well and I managed to setup an HT813 in Freepbx, with outbound and inbound calls, and incoming id caller. Greetings from Colombia
Thanks Commsprepper for this great video.
The only thing, outgoing calls was not working for me showing a "forbidden" message in Asterisk logs. I changed the "FXO Port->Stage Method (1/2)" to 1 as opposed to the default value and it started working fine.
I will I could help but I had to get help to get this working in my shack. I am clueless what all the settings do. Telephone is not my thing. Sorry.
thank you very much, i tried this and finally it worked for me
sam bchb Good to hear
crucial info: if using the ht813 make sure to toggle the FreePBX setting to yes inside Settings>>SIP settings 'Non-standard g726' otherwise it won't work!
Thanks for the comment. I am not an expert with this unit - at all - and appreciate folks adding to the comment section. Respectfully.
In Mexico we say Eres un Chingón , when you are a MASTER, Thanks very usefull!
Thank you for the comment. I can't take credit for all the content. I had allot of help.
Well done fantastic work
Thanks for doing this. I was struggling to get my Grandstream HT802 to connect to FreePBX until I came across your config setting with the x.x.x.x:5160 for the SIP server instead of just the IP address.
I'm trying something similar to you with a HT802 and FreePBX but haven't had any luck getting it to work, any chance you can share your Grandstream Device Config settings?
This video was done extremely well. Thanks!
Thank you for the comment and channel support.
THAT... was a lot of work! Thank you for your efforts!
Thanks for the comment and channel support.
I am going to try and adapt this for my magic jack, great work!
Awesome video. I used watched this video and got HT 813 to work
Thanks
Thank You very much for this video, you saved me alot of time. Informative, clear and to the point. Job Well Done
Thank you for the comment and channel support.
Hi, i configured devices like you but when i call (outgoing) the ring after two times state change to connected and then nothing happen
What should i do, please give help me
remarkable video
Cool looking at using a similar setup for my home phone.
oop nope you are doing something different
Thank you for this video. Would you mind helping me setup Grandstream DP750 to for in-office communication.
Hi just a question would you know why i can only receive calls but can't call out ? any idea why ?
This is a really great video and I hope to make use of it soon. I have been trying to do this very thing with a Linksys SPA3000 or a 3102 using the Issabel PBX software on a HP mini PC but have gotten nowhere after several weeks. Have you made a video using either of the Linksys ATAs?. I am now starting to set up a FreePBX system as it seems the most popular now and have watched the series from Chris of crosstalk solutions called FreePBX 101 v15 but I have gotten stuck on the firewall page at the moment.
this setup can work for connecting landline port from ONT modem to FXO port in HT813? the goal is, my smartphone can be used to do a call or receive incoming call from my landline number using zoiper wherever i am?
Good work, brother. Thank you.
Thanks for the comment and channel support.
Great video, I used to build Asterisk PBXs as part of my first IT job about 11 years ago, those were great instructions and the interface has improved a lot. This is a cool peace time idea, but without rule of law the IP to IP calling would be sufficent I would imagine? I mean you can use the PBX if you have a lot of phones in the system and want to make internal dialing easier and setup conference calls (OPs calling into CP), otherwise IP dialing would be sufficient. Any other thoughts on that I may be missing?
i have setup done but i have Problems from the Grandstream HT813 Ata is always auto CallBack to my FreePBX with inbound extension, so anyone can help me fix this Problem please? Many thanks
I tested with Asterisk server running over the Internet behind a NAT and didn't works for incoming calls.
can you make your pi into a pstn auto dialer like those in a home alarm system without a voip account? i have an existing land line and wondering if i could use my pi to make calls via the land line if an alarm is triggered
Great video. Does the ht503 work in sending the incoming caller id to the pbx?
Did you manage to do this?
Oh wow. You went through a lot of work to make that work
It was not easy. I think this is the first video on the subject. I know it's the my best video but something had to be posted.
@@Commsprepper I'm not the best with computers but I'm learning from you and others and from trying different things also. I had to program a wind turbine inverter over the phone with a laptop the other day. There's more to it then one thinks
No firewall ports open are needed since this is using ATA right?
If your calls are not going out is because you didn't put the stage method to 1 like me. Spent 3 days and finally noticed I had my stage method set to 2
Thanks for the comment.
I followed this to set up an HT-813 and it works great other than the number passed with an incoming call. I always get the dialed number, not the calling number. Any idea what could cause this?
No. I don't fully understand how this works. I had allot of outside help making this video.
@@Commsprepper I found the issue, have to forward it after 2 rings and not 1 :(
Hey, I've been trying to find an "Elmer" for a long time to help me get into radio for self-sufficiency and prepping. I only talk to family on the phone anyway. Work coms is through SIP phone and equipment they provide. Can you help with that? My local radio clubs here are not very helpful. Not many old school guys left. I know a bit about tubes and can build basic amps with them, but I really want to learn more about radios hobby wise but also setting up a legal licensed short wave...I've got CBs but that is a whole different issue there.
Not sure what your question is. Can you rework it.
Can you share a pdf of this presentation?
Can it work with a FreePBX server PC (Desktop) or is it just only Raspberry Pi?
do you have the comand configuration of the trunk for the asterisk server ?
Saw Bolivia hit with 2 earthquakes. EMCOMM to action. Hope your ok?
All is well. Did not feel them in La Paz. Thanks for asking!
FXO Grandstream HT503 = RJ11 PORT
My PBX server using RJ45 connect to WAN port, but PSTN using RJ45 connect to FXO with RJ11.
so, can i using RJ45 to RJ11 converter ?
that is still work?
thankssssssssssssssssssssss
You are welcome.
Hi, I wrote you in one of your videos.
I would like to get in touch directly with you. This videos is great for emergency, I'm ham radio and also member of the Chilean Red Cross , also electronic engineer but old-fashioned.
Please help me to get you help.
73's