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ht-503 FXO port as a trunk in asterisk (using freepbx as front end)

neel23
Posts: 8
 Member Since:
2008-08-01
Submitted by neel23 on Sat, 08/02/2008 - 9:42am.

there are other threads on this topic, but i have not managed to do this and therefore am starting a new thread.

I created a trunk as per advise in this thread, and entered the trunk name and password in the FXO port of HT-503. However, HT-503 status page does not show registered after this. Makes me believe that my FXO port setup as a trunk is not yet working.

Can others who have made it work share whether their HT-503 shows as registered on the status page of the device? Any help will be appreciated. Thanks!!

‹ GXW-4004 and GXW-4104 setup for Trixbox gxp 2000 and callforwarding ›

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GameGamer43
Posts: 30
 Member Since:
2006-11-16
I have an HT503 from
Sat, 08/02/2008 - 2:23pm

I have an HT503 from Grandstream working with Elastix. You can check out my writeup on the setup at http://voipcoop.org/viewtopic.php?t=65

--

Bryan Walters
Schmooze Com, Inc.
Website: www.schmoozecom.com

-FreePBX OTTS Certified-


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neel23
Posts: 8
 Member Since:
2008-08-01
strange thing -- i located
Sat, 08/02/2008 - 9:08pm

strange thing -- i located your post earlier. power of net. tried the solution a few times. but my HT-503 would not register. i tried again after i saw your post. and HT-503 registered. wondering how.

OB calls now work!
IB calls did not work. Until I enabled "anonymous inbound SIP calls". Now they work.

isn't the last option unsecure? is that you how you have it working?

thanks for the post!


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neel23
Posts: 8
 Member Since:
2008-08-01
i finally managed to get
Wed, 08/13/2008 - 9:26am

i finally managed to get ht-503 working as trunk. had to enable anonymous calls in freepbx.
did you also have to enable that option in general settings?

another question: are you able to receive caller id on FXO (from calls that are received on the PSTN line)?


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GameGamer43
Posts: 30
 Member Since:
2006-11-16
Sorry for the late response.
Wed, 08/13/2008 - 3:52pm

Sorry for the late response.

As for your first question, you are absolutely correct, you need to have anonymous sip calls enabled for this to work.

As for your next question, I am receiving caller id on the HT503. As I explained in the voipcoop.org post, Caller ID Minimum RX Level (dB) may need some tweaking based on your Landline provider. If you play with that setting, you should find one that allows you to receive the caller id of calls coming in on the pstn line.

--

Bryan Walters
Schmooze Com, Inc.
Website: www.schmoozecom.com

-FreePBX OTTS Certified-


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neel23
Posts: 8
 Member Since:
2008-08-01
alright. i will play with
Wed, 08/13/2008 - 10:00pm

alright. i will play with all the RX level from the min to max. hope it works.

thanks!


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khakserv
Posts: 26
 Member Since:
2008-07-22
Configuration
Wed, 11/05/2008 - 3:34pm

Could somebody please paste the full configuration that is working? i also have a HT-503 and want to add it as a trunk, i use trixbox 2.6.1.13


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neel23
Posts: 8
 Member Since:
2008-08-01
this might help you,
Thu, 11/06/2008 - 4:28am

this might help you, khakserv

http://img.skitch.com/20081106-m9yrfk7c5d8tqp5tkqs9h9xger.jpg
http://img.skitch.com/20081106-tn5piyaij5pqpu7cq6bja5h1at.jpg
http://img.skitch.com/20081106-euus8md7bteg1c6qp1dk9mh1sy.jpg

also, you need to enable Allow Anonymous Inbound SIP Calls? to yes in general settings on the freepbx page.


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khakserv
Posts: 26
 Member Since:
2008-07-22
Settings
Thu, 11/06/2008 - 12:12pm

Thankyou for your help, i changed some settings and added the nat=yes line in my trunk settings, but it is still not working, i have a question, is the extension 705 you are forwarding the calls to a valid extension? because i dont want to send my calls to an extension but to the IVR, i already have allow anonymous inbound sip calls set to yes and my inbound used to work in version 2.2.9, but not anymore.
In the outbound i get rings to the number i try to call but noone ever answers even in numbers i know someone will pickup.
any ideas?


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neel23
Posts: 8
 Member Since:
2008-08-01
705 is indeed an extension.
Thu, 11/06/2008 - 1:21pm

705 is indeed an extension.

other than that, i do not have any ideas...


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khakserv
Posts: 26
 Member Since:
2008-07-22
Got it working somehow
Fri, 12/12/2008 - 10:36am

i can now dial the number of the extension and get a dialtone, but i cannot make it dial automatically, that is without dialing the extension number before dialing the local number
any ideas?


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neel23
Posts: 8
 Member Since:
2008-08-01
it is not an extension as
Fri, 12/12/2008 - 12:45pm

it is not an extension as much as it is a trunk.
use it as a trunk and then pass suitable numbers to this trunk.


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AlfredF
Posts: 1
 Member Since:
2009-04-23
Couple more things I found
Thu, 04/23/2009 - 8:56pm

One thing that tripped me up, is that you need to remember to set port=6052 in the Asterisk trunk. Otherwise it tries to connect to the FXS port of the 503.

I setup the following changes from this thread and got it working. Some are obvious (like the max channels) but since I hadn't seen it mentioned, thought I'd toss them all out.

And a couple of things gotten from the Grandstream forum:
FREEPBX SETUP:
SIP TRUNK:
Maximum Channels: 1 (you can only receive/make 1 call at a time over a PSTN)
OUTGOING SETTINGS:
type=peer
port=5062
HT503 SETUP:
FXO SETUP:
DTMF audio=no
DTMF via RFC2833=yes
Enable Call Features=no
Dial plan={ [x*]+ }
PSTN ring-thru=No

Hope this helps. I love when things end up working like expected. I just got this in the mail a couple hours ago, and both my FXO trunk and FXS are working. No comments yet on quality, I getting a little line noise, but it's probably the 100' cheap phone line I used to test. (How can I not have a short cord around, every time I clean the office I must find 50 of them)

Good luck guys.


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osviweb
Posts: 11
 Member Since:
2009-04-20
Me I'm struggeling with the
Fri, 04/24/2009 - 5:04pm

Me I'm struggeling with the HT-503 as you can see in my post.
I had similar issues like yours which I solved as you describe here but I still have.

Horrible sound with very low volume and some echo when receiving a call from fxo to trixbox. This especially on announcements which came with a delay as well.

Caller id not passing by . Tried to change configurations.

Call comes is but it does not go on the inbound route but to the first available extension .

In particular the first problem of BAD AUDIO is very bad for me and it looks I'm not getting any solution.

if you want to hear how crap is my audio from ht-503 to sip you can hear this recording Audio recording has been made on my mobile and even if the message is in italian you can notice how much noise + low volume + delay I have on this.

It looks like a common issue but no idea on how to solve it.

--

Osviweb
travel agency Italy


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lostInTheFog
Posts: 1
 Member Since:
2008-09-04
I have been playing getting
Tue, 05/26/2009 - 12:39pm

I have been playing getting caller ID as well. Since new to asterisk (now testing freepbx and trixbox CE) doing a lot of experimenting.

HT-503 is connected to a Vonage device as the POTS line into it. To get caller ID working I have found the following in the FXO setup on the HT503

Number of Rings effect on receiving caller ID
1=can't get this to work (all but caller ID works fine)
2=works if change the Caller ID Minimum RX Level (dB): -60
3=works using the default of -40dB


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hootjr29
Posts: 47
 Member Since:
2007-05-15
Yes, lostInTheFog is
Sat, 03/27/2010 - 2:08pm

Yes, lostInTheFog is absolutely right. The only way I was able to get my callerID to work properly was to use the following:

2=works if change the Caller ID Minimum RX Level (dB): -60

THe others may work too, but I need to test and see.


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