Ok for incoming call, setting on basic setting (why??) reach PBX incoming channel trunk.
Now is time to resolve how FXO registration work and outgoing channel too…
Not as expected as it where a remote trunk, user/administration manuals lack too much information about.
Ok for incoming call, setting on basic setting (why??) reach PBX incoming channel trunk.
If you need help point what is not working.
Hi Marcin, good point, what is not working is registration of FXO channel…
So it behave as an extension, but it is not an extension: it must be a trunk.
Question: how can I configure HT to get registered or not then make call from an Asterisk TRUNK interface?
What is tel URI? Enabled how it behave? Phone=User, disabled what this means? Is useful an user/administration manual that mention captain obvious?
Is not question of risk more opening network where this is, OLD analog PBX can remain there for another short while, I cannot risk open a network to unsecure people.
I cannot waste more time preparing an open test bench, Asterisk has no cost than knowledge and skill, HT has defect, it has to be GS loaded no more mine loss (see next).
Registration how it really work? (on HT clearly, I am aware of how SIP work)
Make call without registration, how and when has some scope.
Edit: in general a more detailed user/administration manual on how setting behave on SIP transport/signalling.
Also some other details need to be addressed with more details.
On firmware, I feel FXO interface settings can be better addressed in a more different layout:
Call forwarding in a sip serting page and not in basic setting, all SIP on a page,
all Office SLIC on another page.
As is appear as a messed intermix where details about a single interface are spread where they don’t have sense to be in.
User manual can be a simple description
Administration manual MUST be an administration one, addressing thecnical topic in deep and not a mere copy of user one.
In this perspective I feel GS product of many Year ago where more and more accurate in documentation and firmware, today appear as less professional.
On early day of SIP I remember I used 286, 386 and 486, then 488 succeded, these product where less accurated on SLIC section, all device I owned failed on SLIC interface circuits but software was well and strong enough.
Today with reversed HW FW rules, is still a good product? Yes FW design has a cost, skilled software designer cost, I know about but GS has to decide about how much cost a defective product.
Can be mass production plagued by a poorly designed software and less responsive quality test and customer support too?
HT813, bought 2 unit:
one was never touched,
the first has now been repackaged then stored on wait room.
We are now planning what to do about all GS product, not sure to wish still have at office.
Now i can reply
- If you want register to trunk, you must make it possible on Asterisk that trunk will accept register from HT (not really know how to set it, on UCM i have option for his).
- Peer : turn of register on FXO HT port. Then it will start as “peer” and is how you usually do it.
Most documentation from 503 be directly used by HT813, so check this.
Ht 503 / Ht813 is one of most hard gateways to configure as it require a lot of knowledge: How pstn work, how sip work. You also need knowledge about PSTN you use (CID type, SLIC type, call signals).
As for setting forward it straight easy (if you know how) but you need knowledge.
Forward is on basic so USER can change it (no idea why) without messing true settings, this gateway is designed as home use more then business. GXW series is designed for business.
FXS gateways are easy to set, while fxo forward is not easy.
Hi Marcin, I don’t need knowledge about how SIP s working.
I need information on how configuration of HT work and if bugs where solved too.
You say I can infer from HT508, I read and is not so clean as necessary.
Is not my job to use more in deep wireshark to debug a defective product or discover information forever hidden.
Knowledge on how to configure HT can be grasped by having missing information all there are hiding.
How to solve? I lack information from GS, only one solution is to switch off device repackage and put back to GS.
You need knowledge, what you trying to do is not simple matter whatever device you buy. You trying create bridge between PSTN, HT VOIP and SIP server. All of them must be correctly set and if you have no idea what you do then hire specialist. There is no one way to configure this it depend on your country PSTN signals, and how PBX is set. General GS document is clean if you understand what you read.
You need set sip PBX correctly, then you need set VOIP account on FXO (HT) then you need configure PSTN correctly (FXO).
We can try help with problems but not with documentation. I personally feel that if something is more complicated then i hire special which do it for me, it is less costly then wasting my time over something i do not need in life
PS. FXS is very easy to do, FXO is magnitude or two harder…
Hi Marcin, As firmware/hardware developer I can say the last sentence:
I know how to setup My FXO drivers to mate what I know.
I CANNOT setup foreign hardware I don’t know in details and feel they lack documentation.
SLIC settings seems correct, no problem detected and I feel no need to use instrument to measure nothing.
At software layer (Stack) I suppose it is quite impossible when setup is missing behaviour.
Marcin, please don’t say word without knowledge, I never hire specialist/colleague to setup a scrap device.
Checkup, competitor manual are thousand pages, GS are two similar short and quite useless manual for casual home user.
Try read competitor software manuals:
As specialist I just give back and say to colleague never use GS product, maybe just hobby IP phones to small home pbx if limited to connect to a provider/simple pbx.
GS documentation lack important details and cannot be used as gateway without them.
Forum and service continue mess up about my knowledge.
My knowledge now are complete: HT813 is a crappy product with buggy firmware and useless manuals.
As from your word, yes I need knowledge on how HT813 setting works. If GS is not able to handle this gap and “expert” reiterate same offending word, HT813 remain useless… then just few word remain:
Good bye GS.
As an aside, as I am not able to fully discern the screenshots, so, you may want to take a look at how the ports are used between the FXS, FXO and the PBX.
Both FXS and FXO should register to the PBX using 5060.
The FXS and FXO each have a different local port set so that the response coming back from the PBX is unique for each side. PJSIP channels are in use.
There really should not be any difference in how the FXS and FXO sides register as the only things required are:
SIP server IP and port
and the local SIP port for each.
At the time of the original post, there was a Beta firmware, which by the time you posted, the fix that was contained in the Beta, had been rolled out to the 1.0.2 version and the Beta was removed.
Hi Ipneblett, I decided to use the simplest solution:
Buy a couple of SLIC and a couple of SLAC hybrid modules, made a PCB with a mixed signals processor then interface to Raspberry PI bus.
My old time FXS FXO firmware where based on an MSP430 able to manage two line in parallel, today I use STM32, it offer more processing power and USB so it can offer another gear over old times of tight filtering.
It is more simple to write DSP code then an Asterisk device driver than have complete description on how HT813 setting behave on SIP protocol.
HT813 is the only product on the market so I am sure someone can be interested on a device can use without hassle of:
- how can I place a call on extension? (Behave as extension not TRUNK)
- having a decent manual
- correct assistance
- Marcin comment on “FXO must be enabled” without reading what is on screen shot
- Again Marcing say me I need knowledge
- support say need configure SLIC/SLAC and never caught HT SIP has a lot of problem
- manual don’t say how setting work.
- Administration manual is a mere copy of user and useless.
- discover FXO act as Extension and not as Trunk.
20 day and more day without having a working device are out of acceptable.
Well, “each to his own”, as they say; and glad that you have the skill set to develop your own solution. Hope it all works out for you.
Ipneblett, my FXSO Software is in service since 2002, last sold solution was on 2005, last few hundreds units still operational as they leave space to SIP integration.
Software is few pages of code but over service time I wrote more than hundred pages documentation. Some notes about international impedance and signalling setup, setting templates too.
It is simple to rewrite HAL to adapt to industry standard SLIC/SLAC, adapt to new uC and is simple to maintain filtering interrupt. PSTN is going to sudden death, maybe it remain for last time opening market.
Last SIP application I wrote is 10 yr old, I never touched up but again everything is documented.
See other solution? No competitor so???
Did you see other solution than lose time with network analyser to hack/debug HT settings?—
Is useful have a manual that say URI setting is Disabled peer=phone enabled?
Is useful have manual say registration is yes no?
This can be seen on Web interface but what are changing all these settings?
How can I set settings to register/act as trunk?
FXO SIP must be a trunk not an extension.
I said too many times none answered right way how setting works.
I’m too experiencing problems with the FXO port of the HT813.
The context is as follows: I run a PBX setup with Asterisk software and HT503 ATAs on two different locations that I will call A and B. Both sites have a landline provided by our ISP. I use these systems to provide an in-home telephony system and to filter out unwanted phone calls (mainly telemarketers). The setup is the same at both sites: the landline is connected to the HT503 FXO port, fixed handsets to the FXS port and our Android smartphones with a SIP client (CSipSimple). Thus we are able to pick up an incoming or originate an outgoing call from whatever phone connected to the system (physically or by software). Furthermore, the 2 systems are interconnected by IAX so that the in-home telephony system encompasses the 2 sites and legitimate calls from one of the landlines can be transferred transparently from the called site to the other.
All this has been running like a charm for several months until the HT503 at site B broke down following a switch-off/switch on. The faulty device has been sent back to the after-sales service. I bought a HT813 as a replacement (the HT503 has reached its end of life). The two devices seemed to be quite alike, so I thought updating the configuration to the new device would be a piece of cake.
I didn’t change anything in the Asterisk configuration files (sip.conf, extensions.conf). Some of the parameters of the ATA configuration may seem cryptic, but when I first configured the HT503, I managed to get it working almost at once while keeping most of the default values.
To be honest, I must mention the work of Graham Miln who cleared the way for me:
As for the HT813, it was a piece of cake for the IP and FXS parts, but not for the FXO. I tried to duplicate as closely as possible the HT503 configuration for the FXO port to the HT813. SIP registration is OK for both FXO and FXS, but:
- it is impossible to originate a call to the FXO port: the call is immediately hung up with a 403 (“Forbidden”) response, as seen on the Asterisk console. If the FXO port is unplugged, I get a busy tone on the calling phone and a 486 “Busy here” response.
- it is impossible to process an incoming call from the FXO: the caller gets a “Reorder” tone just after the number has been dialed; no trace of the call on the Asterisk console.
The FXS port works fine: calls between the FXS, locally connected smartphones and the remote Asterisk are OK.
While I can find a solution I keep the HT813 FXS and FXO unplugged and the fixed handset directly plugged to the Internet box. So incoming calls are no longer filtered out.
So what’s wrong with the FXO port? Is it a hardware fault, or a software bug?
Th software version of my device is 22.214.171.124, the last available from GS. Don’t know if my problems have something to do with those above in this forum.
Some people can’t have their FXO port register properly with the same FW version as mine. Here is an excerpt of my sip.conf file from Asterisk, and the registration works fine.
Looking forward to any good idea…