HT-488 FXO port
Submitted by kenknight on Wed, 03/14/2007 - 12:01pm.
Hi All,
Just got in an HT-488 to play with a bit and naturally have run into some problems. I followed the guide over on Voip-Info regarding the setup and while that "sorta" works, it is currently less than ideal.
Grandstream has a beta firmware available that supposed supports One Stage Dialing which would be more in line with normal sip gateways. Has anyone been able to get this working yet and if so, how? Also, the echo was out of this world!
Also, I made a couple of calls out and in and the thing seems to just disconnect you after 10 or 15 seconds??? Anybody else experienced this?
Thanks,
Ken
Re: HT-488 FXO port
Tue, 02/06/2007 - 2:59pm
Finally got the FXO to work correctly as outbound trunk..
Ok I finally got it to work.. FXO
I use FreePBX
1. Create extension: Match up the login details for the FXO account on the HT-488
secret =MYPASSWORD
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
type=friend
nat=yes
port=5062
qualify=yes
dial=SIP/THEEXTENSIONNUMER
2. Create Custom Trunk: This was my main pain..
1 Channel Then:
Custom Dial String: SIP/,,,w$OUTNUM$@MYEXTENSIONNUMBER
3. Set outbound call route :
to SIP/,w$OUTNUM$@MYEXTENSIONNUMBER (Duh)
Then on the adapter settings:
on FXO page
Send DTMF:RTP(RFC2833)
(So you don't here the number being dialed)
Then fill in matching account info for the regular extension you created.
Allow outgoing call without Registration: yes
PSTN AC Termination: set to american 600 ohm
I did update to the newer firmware but I don't think that helped it work.. HT488-496-386_1.0.3.70
All in all it seems like this could have been alot less of a pain..
Re: HT-488 FXO port
Thu, 02/22/2007 - 5:59am
I upgraded to the firmware mentioned above and selected ONE STEP dialing and defined a TRUNK as follows:
canreinvite=no
context=from-pstn
dtmfmode=rfc2833
host=dynamic
qualify=yes
secret=YOURSECRET
type=friend
Define the trunk in an OUTBOUND ROUTE and it worked! In this configuration I have the FXO port registered to Trixbox as a TRUNK, and the FXS port as an extension.
The only thing that is still annoying is that an incoming call on the FXO port has to ring at least once on the FXS port before forwarding the call to VoIP, if Gransdstream could program the device for direct forwarding, it would be a functional replacement for the SPA-3000.
8-)
Re: HT-488 FXO port
Wed, 02/28/2007 - 2:19pm
Hi eihoward,
can you elaborate on the steps you took to get it functioning.
I created a sip trunk and pasted the info from your post in the peer details section modifying the password.
Then I logged into the phone which had already been setup as my previous post and I'm not seeing the unit being registered on the fxo side.
I do a sip show registry and theres nothing there.
Thanks for any info.
Cheers!
Ken
Re: HT-488 FXO port
Tue, 03/06/2007 - 7:24am
On the HT-488 FXO Tab I configured:
SIP SERVER: your.trix.box.ip
SIP USER ID: sip_trunk_name
AUTHENTICATE PASSWORD: your_secret
AUTHENTICATE ID: sip_trunk_name
on FreePBX I added a new SIP Trunk
Trunk Name: sip_trunk_name
Peer Details:
canreinvite=no
context=from-pstn
dtmfmode=rfc2833
host=dynamic
qualify=yes
secret=your_secret
type=friend
If every thing is configured correctly you should get your FXO to register.
To dial through this trunk I did change on the HT-488, Basic Settings Tab FXO One Stage Dialing for One Step dialing: YES.
I created an Outbound Route, and selected the Trunk, and it worked.
For incoming calls:
on the HT-488 I changed, Basic Settings tab:
NUMBER OF RINGS: 1 (number of phone rings before a PSTN incoming call is forwarded, default 4) :-( ,
FORWARD TO VoIP: xxxXXXX@your.trix.box.ip
on freePBX:
I created a new inbound route with DID: xxxXXXX
this allowed me to select what to do with incoming calls from the HT-488
One annoying thing is that the FXS port will always ring once before the incoming call is forwarded to Trixbox, but it works OK.
8-)
Re: HT-488 FXO port
Wed, 03/14/2007 - 12:01pm
After I made the setting above, I get the following:
-- Goto (macro-dialout-trunk,s,26)
-- Executing Set("SIP/2003-b7905b78", "the_num=2145971111") in new stack
-- Executing Dial("SIP/2003-b7905b78", "SIP/,,,,w2145971111@5733|300|m") in new stack
-- Called ,,,,w2145971111@5733
-- SIP/5733-08f93e18 is ringing
-- SIP/5733-08f93e18 answered SIP/2003-b7905b78
-- Attempting native bridge of SIP/2003-b7905b78 and SIP/5733-08f93e18
== Spawn extension (macro-dialout-trunk, s, 27) exited non-zero on 'SIP/2003-b7905b78' in macro 'dialout-trunk'
== Spawn extension (macro-hangupcall, s, 10) exited non-zero on 'SIP/2003-b7905b78' in macro 'hangupcall'
5733 is the extension of the FXO port, connected to a POTS line
2003 is the extension (SIP phone) I am dialing from
2145971111 is the number I am trying to dial out to
I have a few questions about this:
Where did the |300|m part of the dial sequence come from? (and what does it do)
It looks like the failure occured after Asterisk tried to "bridge" to the FXO "extension", where can I look (log file?) for more information about the exit code?
thanks in advance!
Brian
Beta Firmware
Fri, 03/16/2007 - 3:16pm
It is available at:
http://www.grandstream.com/BETATEST/HT488_496_386/
Provides the One Step Dialing...
Software Version: Program-- 1.0.3.70 Bootloader-- 1.1.0.1 HTML-- 1.0.3.70 VOC-- 1.0.0.13
Please try to configure as per my instructions and disregard all other changes... My instructions are detailed in my post of March 6th.
Thanks for the additional
Mon, 03/19/2007 - 8:08pm
Thanks for the additional info. I have the beta firmware loaded in the 488 now. I did a factory reset to make sure all my previous settings are gone. When I try to place an outgoing call, it acts like it connects for an instant, then disconnects. (if I monitor the FOP, I can see the extension I am dialing from go red, then the trunk goes red for a short time)
In the debug info, it looks like the error occured after it tried to bridge the two SIP devices:
-- Executing Dial("SIP/2003-083b2e00", "SIP/xxxxxxxxxx/yyyyyyyyyy|300|") in new stack
-- Called xxxxxxxxxx/yyyyyyyyyy
-- SIP/xxxxxxxxxx-083b8340 is ringing
-- SIP/xxxxxxxxxx-083b8340 answered SIP/2003-083b2e00
-- Attempting native bridge of SIP/2003-083b2e00 and SIP/xxxxxxxxxx-083b8340
== Spawn extension (macro-dialout-trunk, s, 20) exited non-zero on 'SIP/2003-083b2e00' in macro 'dialout-trunk'
== Spawn extension (macro-dialout-trunk, s, 20) exited non-zero on 'SIP/2003-083b2e00'
-- Executing Macro("SIP/2003-083b2e00", "hangupcall") in new stack
To me this means the 488 has answered the request, but somthing bad happens when they try to start talking....
Any suggestions?
Where should I look for more clues?
Works, almost
Wed, 07/11/2007 - 9:11pm
I followed your instructions. I think I have everything squared up. One thing is NOT working.
I have set the Inbound route for the FXO trunk to call the SIP extension used by the FXS port. This works. Just have to NOT pick up the phone on that 1st ring.
I can call from an extension with the dial prefix that gets to the Outbound route that goes to the FXO trunk.
But if I call from the FXS SIP extension, I get a busy. I actually see the SIP traces of ringing the FXO port, and it being answered:
-- Executing Dial("SIP/2202-0a0fb188", "SIP/249809/2482192059|300|") in new stack
-- Called 249809/2482192059
-- SIP/249809-0a1255d0 is ringing
-- SIP/249809-0a1255d0 answered SIP/2202-0a0fb188
-- Attempting native bridge of SIP/2202-0a0fb188 and SIP/249809-0a1255d0
== Spawn extension (macro-dialout-trunk, s, 20) exited non-zero on 'SIP/2202-0a0fb188' in macro 'dialout-trunk'
== Spawn extension (macro-dialout-trunk, s, 20) exited non-zero on 'SIP/2202-0a0fb188'
It dials, it rings, it is answered, it fails....
Of course I could just go directly out, but this should work....
And I think it is a bummer that the FXS port has to ring once before the VoIP forward starts. One more delay.
Trash your HT-488
Thu, 07/12/2007 - 10:28am
I have my HT-488 really working and it has **TWO** major defiencies:
PSTN forwared to VoIP (FXO to VoIP) does NOT send the caller ID info. From
http://forum.voxilla.com/grandstream-support-forum/confirmed-ht48...
Ok, Grandstream Tech Support confirmed by email that callerID forward to voip line will be not supported in HT488.
here a transcript of the email.... I'm really dissapointed with Grandstream cause in several adversitements, they note that these feature works.....
>>Ticket#: 2007022099000851] RE: Please, suppport with [...]
>>Dear Daniel Leonardo Albe - RIVO,
>>This is a known problem. As result from luck of resources we are unable
>>to transfer caller ID information to the VoIP network extension.
>>This feature will be supported in our new generation products,
>>will come soon.
>>Best Regards,
>>Eduard Rubinshtein
>>Grandstream Technical Support
That is go and get the HT-502, maybe.
Now the other is more subtle. I have not been able to work out a configuration for:
FXO -voip-> Asterisk -voip> FXS
or the reverse. I get a busy condition.
This is also a killer.
The last is probably a bug. Doics for firmware 85 said that you can now set teh ring number before PSTN forwared to VoIP to ZERO! I am running ver 86; if I set it to zero, it never forwards. Or I think it forwards after 4 rings, which is when that line's PSTN voicemail picks up....
So I am now shopping for alternatives. But it needs 2 FXS and the FXO, and I don't need the WAN/LAN portion....
HT-488 for free
Thu, 07/12/2007 - 10:37am
nice, quiet HT-488, just sits there, doesnt bother anyone and never does anything. Needs to go to good loving home where it will not be required to do anything there either...
Any takers...? I have all the papers for it. We're moving and our new apartment doesnt accept grandstream devices (even with a deposit)..arghhh...
[what a joke this thing is... ]
Freepbx and ATA problem
Sun, 01/27/2008 - 11:03am
i want to tie among 13 branches each one contains a Panasonic PBX by a SIP
Asterisk Server (Freepbx)by 2 ATA's in each branch..the problem is that there are
repeated extensions between branches
so i have to make Each branch's ATAs as a trunk with a specific prefix..but
i observed that in Grandstream ATA HT-386 i can call only the 1st FXS Port
and when another call comes it gives busy and didn't handle it on the 2nd
FXS Port...
So should i make a trunk for each FXS Port in the ATA
Please Give me an EXAMPLE....in details please
i am using Freepbx....what should i make SIP or custom trunk and what i
should write in PEER Details? and how can i make an outbound route for
each branch
Thanks Alot
Incoming calls
Fri, 02/08/2008 - 2:33am
Dear all,
I've just purcased an HT503 yesterday. (But is similar to HT488)
I've configured it to work with my trixbox as a PSTN gateway.
I've a question and i hope you can help me.
The question is about incoming calls from PSTN: when someone calls me in my PSTN public number the HT503 wait 4 rings (but i configured just 1) and then it forwards the call to the voip number (in trixbox) i configured. The BAD thing is that when HT503 forwards the call it ANSWER the call (pick up the line) and then it stars to dial in trixbox: the result is that the caller starts paying after 1 ring and it pays even if noone in my trixbox pickup the call.
Is it regoular? I hope i'm missing something because if this is a regoular behavoiur i really disappoint with Grandstream. If someone calls my PSTN number i don't want he has to pay even if noone in my house answers the call.
Please help me!
Thank you
Marco from Italy
need help configuring ht503 as trunk
Fri, 08/01/2008 - 7:25pm
i have tried the trunk approach listed above with my ht503 and just cannot the device to register. on the ht-503 status page, if i enter Asterisk extension credentials then FXO shows as registered. if i enter trunk details, as described above, then the status page continues to show "not registered".
pl help!! i have read everything that i could get my hands on, but no luck :(
Member Since:
2006-08-02