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


Member Since:
2008-08-01