I was reading anther post but there wasn’t a resolution. I purchased a grandstream ht813 how do Iset it up to accept a call and do a #10 so it pages the pager unit?
This is an HT813 question, not a FreePBX one. You would have to set up the FXS port for use with a paging amplifier. I can’t find any online documentation that even suggests this is possible.
Alternatively the paging amplifier will need to answer the call, and somehow detect the end of the call, and go back on hook, which would make it a Simplex 5100 question.
I’ve been researching various threads about the HT813 setup. Currently, we use a standalone phone connected to the system, dialing #10 to trigger the page.
I’m trying to replicate this setup with the HT813 so that any phone extension within the school can call the HT813, which would then automatically dial #10 and activate the paging system and forward our message .
Most of my schools use multicast paging, but this one has a different setup, so I’m exploring alternative solutions.
Thanks!
One approach:
Set up a pjsip trunk with Registration: Receive and set up the HT FXO to register to the PBX.
may be useful; you can find other guides online.
Using # in called numbers is often problematic, so I suggest that you set your Outbound Route to rewrite some numeric string to #10. For example:
prepend: #10
prefix: 7243
match pattern: (blank)
CallerID: (blank)
Also, set the Dial Plan for the FXO port in the HT to permit #10 to be dialed.
Then, when the user dials 7243 (PAGE), the PBX will send #10 on the trunk and with luck the 5100 will respond.
If you still have trouble, paste the Asterisk log (with pjsip logger on) at pastebin.com and post the link here. If the HT fails to register, paste the log for a registration attempt. Otherwise, paste the log for a call to 7243 (or whatever else you use to trigger the page).
HI thanks for responding I been reading that thread.
for the #10 in the ht should it look like this?
ok, I created the sip trunk and the outbound route. The ht now says Idle Connected and Registered picture is below. When I dial 7243 I get all circuits are busy now.
Can you share some details from the asterisk log? One quick suggestion that may or may not be applicable; If you are using the Class of service module make sure the new trunk has been set as “allow” in the proper group(s)
Also do make sure the route has the trunk associated(sometimes missed), and that no routes ahead of it have anything that would take a priority match prior to this route.
Possibly the dial plan requires the {} syntax. Try
{#x+}
which should allow anything starting with # to be dialed.
If you still have trouble, at the Asterisk command prompt type
pjsip set logger on
make an attempt to 7243, paste the Asterisk log for the call at pastebin.com and post the link here.
HI never pulled a log how do I do it.
asterisk -rvvv
You can make a call at this point and see what this yields, it may have some quick and apparent answers to what the issue is.
otherwise defer to @Stewart1 instructions in terminal:
asterisk -rvvv
pjsip set logger on
make an attempt to 7243, paste the Asterisk log for the call at pastebin.com and post the link here.
After pulling logs be sure to also:
pjsip set logger off
and ctrl+c to exit.
Not sure if i did it right attached is pastebin
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25399. Dial("PJSIP/3099-0002c1ef", "SIP/test1/#10,300,Tb(func-apply-sipheaders^s^1,(4))U(sub-send-obroute-email^#10^7243^4^1730374738^Mr. Avery^3099)") in new stack
25400 [2024-10-31 11:38:58] WARNING[23878][C-00022ecd] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
You are trying to make a call using chan_sip, when the adaptor is, presumably, registered to chan_pjsip.
I changed it trunk to pjsip and it work thank you everyone for you support. I greatly apperciate it.
Now that the HT813 is set up correctly, I’m trying to add a page group to the outbound route so that phones in rooms without speakers can receive announcements. However, when I apply the group and call extension 7243, the phones ring but the phone says incoming call from ext, and the page isn’t audible. How would I remove the incoming call and just have it pick up the page?
Are we now talking about SIP phones?
You cannot force analogue phone to answer, unless the phone can be configured to answer for certain caller IDs.
Most SIP phones can be configured to auto answer, based on an Answer-After parameter, but there should be safety mechanisms in the phone to prevent this, and you will have to disable those. The ability to auto answer makes the phone into an infinity transmitter.
Hi David,
This is regarding the SIP phones. While setting up this unit, I configured a paging extension so that when someone in our main office dials 6400, all the SIP phones in the classroomss automatically answer, allowing staff to hear our admin assistant through their phone speakers.
However, when I add that paging group to the outbound route under notifications picture below, the phones ring and play an ‘incoming call from ext…’ message over the speaker instead of auto-answering to hear our admin assistant directly.
@aavery, great Job on your first pastebin! I see it looks like at @david55 was able to look at things and quickly get you on the right track. If truly resolved please do be certain to mark the user’s answer that provided the solution. This helps others down the line that may find themselves in similar circumstances!
I noticed you mentioned working with schools. I am in a similar role. I wanted to mention @penguinpbx just added a k12/Education vertical, its in it infancy now, but know that will be a resource and space for many of us in the education space to lean into.
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@aavery, I don’t believe for your use case that particular section is not the intended use. Let’s back up a few steps, what are you trying to do?
I am trying to make one master page that will page the simplex system and my sip phones all at once.
Ok, our best bet is to probably look apt the applications>paging module.
Do you have the commercial module? The reason I ask is putting a lot of extensions in a paging group is very taxing on your system, so a multicast group(s) is what I recommend depending on quantity of endpoints.
For your page it may need to look something like this. Connectivity>Extension> Create a Custom extension(Use a number you well outside of any extension range or feature codes you use.
For our example we will say its extension: “775”
This next part can vary and is really contingent on the Simplex Systems behavior
in the Advanced tab>Edit Extension>Dial
I believe one way may look something like this:
LOCAL/7243@from-internal
Submit and Apply config,
Dialing “775” before to adding it as a device in the paging group should give you an idea to its behavior and functionality
This could take a bit more configuring and tweaking.
Can you describe what the simplex system does now when you dial “7243” is the PBX adding #10 for you or is the Grandstream inserting that?
Let us know if this gets you anywhere, also remember to share your logging, to yield the best responses. 