Re: [Asterisk-Users] grandstream handytone 488 fxo
Soner Tari escreveu: I use HT488, and I can make and receive FXO calls. It's actually quite simple, you create a SIP acount in sip.conf. On the FXO section of HT488 web admin page you enter these registration values. When you reboot the HT488 you should see it registering on Asterisk CLI. What's left is a dialplan line in extensions.conf like this: exten = 9,1,Dial(SIP/sip acount name,10) I've tried your example shown here. When I dial 9 I get dial tone from the HT488 but I when I try to dial, nothing happens (i.e. I keep hearing dial tone even though I'm dialing). Any ideas? Keith Yoder ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] grandstream handytone 488 fxo
I use HT488, and I can make and receive FXO calls. It's actually quite simple, you create a SIP acount in sip.conf. On the FXO section of HT488 web admin page you enter these registration values. When you reboot the HT488 you should see it registering on Asterisk CLI. What's left is a dialplan line in extensions.conf like this: exten = 9,1,Dial(SIP/sip acount name,10) I've tried your example shown here. When I dial 9 I get dial tone from the HT488 but I when I try to dial, nothing happens (i.e. I keep hearing dial tone even though I'm dialing). Any ideas? That may be related with the dtmfmode. Can you try inband? I believe rfc2833 should work too, but once you have it working with inband, you can test the rest. Also I think you'd like to use PCMU codec on HT488, other codecs may cause DTMF detection problems (iLBC seems fine though). In short, I would play with DTMF and codec parameters on both sides. Hope this helps, Soner ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] grandstream handytone 488 fxo
On Wed, 2005-08-31 at 09:54 -0300, Keith Yoder wrote: Soner Tari escreveu: I use HT488, and I can make and receive FXO calls. It's actually quite simple, you create a SIP acount in sip.conf. On the FXO section of HT488 web admin page you enter these registration values. When you reboot the HT488 you should see it registering on Asterisk CLI. What's left is a dialplan line in extensions.conf like this: exten = 9,1,Dial(SIP/sip acount name,10) I've tried your example shown here. When I dial 9 I get dial tone from the HT488 but I when I try to dial, nothing happens (i.e. I keep hearing dial tone even though I'm dialing). Any ideas? What are your DTMF settings? I had all sorts of weird problems with a differant manufacturers ATA because of this. -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] grandstream handytone 488 fxo
nope, i havent :\ Keith Yoder wrote: Casey Boone escreveu: can someone who has a grandstream handytone 488 working with making outgoing calls through the fxo port please post the parts of their config that deal with this port? i cant quite seem to get it to make outgoing calls despite having tried several completely different ways of making that happen. I have one but I too haven't been able to make it work. I've been looking at the config pages for the 488 and trying to make sense of the Route to PSTN configuration. Have you found any documentation for this? Keith Yoder ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] grandstream handytone 488 fxo
I use HT488, and I can make and receive FXO calls. It's actually quite simple, you create a SIP acount in sip.conf. On the FXO section of HT488 web admin page you enter these registration values. When you reboot the HT488 you should see it registering on Asterisk CLI. What's left is a dialplan line in extensions.conf like this: exten = 9,1,Dial(SIP/sip acount name,10) That's for making outbound calls. Once you've done this, you can direct incoming calls to a context like this: exten = 50,1,Goto(MainMenu,s,1) You should enter 50 to Forward to VoIP box at the bottom of HT488 config page also. (Choose an extension as you like instead of 50) But beware, hangup detection method of HT488 was too simple for my needs. Incoming calls may leave the port open indefinetly. (In combination with the FXS port of a HT486, it works, but that's it.) Hope this helps, Soner nope, i havent :\ can someone who has a grandstream handytone 488 working with making outgoing calls through the fxo port please post the parts of their config that deal with this port? i cant quite seem to get it to make outgoing calls despite having tried several completely different ways of making that happen. I have one but I too haven't been able to make it work. I've been looking at the config pages for the 488 and trying to make sense of the Route to PSTN configuration. Have you found any documentation for this? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] grandstream handytone 488 fxo
On Tue, 2005-08-30 at 17:11 +0300, Soner Tari wrote:
I use HT488, and I can make and receive FXO calls. It's actually quite
simple, you create a SIP acount in sip.conf. On the FXO section of HT488 web
admin page you enter these registration values. When you reboot the HT488
you should see it registering on Asterisk CLI.
What's left is a dialplan line in extensions.conf like this:
exten = 9,1,Dial(SIP/sip acount name,10)
That's for making outbound calls.
This means that you have 2 stage dialing, 9 gives you an outside dial
tone. Won't it work with single stage?
_9.,1,Dial(${DIALOUTPSTN}/${EXTEN:1})
Once you've done this, you can direct incoming calls to a context like this:
exten = 50,1,Goto(MainMenu,s,1)
You should enter 50 to Forward to VoIP box at the bottom of HT488 config
page also. (Choose an extension as you like instead of 50)
Problem with this is no CallerID it'll always be 50.
--
Dave Cotton [EMAIL PROTECTED]
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Re: [Asterisk-Users] grandstream handytone 488 fxo
Of course... Those are the basics to get HT488 working for the OP. In this
thread I am not trying to show how to create dialplans.
On Tue, 2005-08-30 at 17:11 +0300, Soner Tari wrote:
I use HT488, and I can make and receive FXO calls. It's actually quite
simple, you create a SIP acount in sip.conf. On the FXO section of HT488
web
admin page you enter these registration values. When you reboot the HT488
you should see it registering on Asterisk CLI.
What's left is a dialplan line in extensions.conf like this:
exten = 9,1,Dial(SIP/sip acount name,10)
That's for making outbound calls.
This means that you have 2 stage dialing, 9 gives you an outside dial
tone. Won't it work with single stage?
_9.,1,Dial(${DIALOUTPSTN}/${EXTEN:1})
Once you've done this, you can direct incoming calls to a context like
this:
exten = 50,1,Goto(MainMenu,s,1)
You should enter 50 to Forward to VoIP box at the bottom of HT488
config
page also. (Choose an extension as you like instead of 50)
Problem with this is no CallerID it'll always be 50.
--
Dave Cotton [EMAIL PROTECTED]
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Re: [Asterisk-Users] grandstream handytone 488 fxo
i greatly appreciate the information and will be giving it a whirl later today :) Casey Soner Tari wrote: I use HT488, and I can make and receive FXO calls. It's actually quite simple, you create a SIP acount in sip.conf. On the FXO section of HT488 web admin page you enter these registration values. When you reboot the HT488 you should see it registering on Asterisk CLI. What's left is a dialplan line in extensions.conf like this: exten = 9,1,Dial(SIP/sip acount name,10) That's for making outbound calls. Once you've done this, you can direct incoming calls to a context like this: exten = 50,1,Goto(MainMenu,s,1) You should enter 50 to Forward to VoIP box at the bottom of HT488 config page also. (Choose an extension as you like instead of 50) But beware, hangup detection method of HT488 was too simple for my needs. Incoming calls may leave the port open indefinetly. (In combination with the FXS port of a HT486, it works, but that's it.) Hope this helps, Soner nope, i havent :\ can someone who has a grandstream handytone 488 working with making outgoing calls through the fxo port please post the parts of their config that deal with this port? i cant quite seem to get it to make outgoing calls despite having tried several completely different ways of making that happen. I have one but I too haven't been able to make it work. I've been looking at the config pages for the 488 and trying to make sense of the Route to PSTN configuration. Have you found any documentation for this? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] grandstream handytone 488 fxo
can someone who has a grandstream handytone 488 working with making outgoing calls through the fxo port please post the parts of their config that deal with this port? i cant quite seem to get it to make outgoing calls despite having tried several completely different ways of making that happen. i have been told that [EMAIL PROTECTED] has this built in to just a button hit, but i dont want to reinstall the box and would prefer to use asterisk directly Casey Boone ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] grandstream handytone 488 fxo
Casey Boone escreveu: can someone who has a grandstream handytone 488 working with making outgoing calls through the fxo port please post the parts of their config that deal with this port? i cant quite seem to get it to make outgoing calls despite having tried several completely different ways of making that happen. I have one but I too haven't been able to make it work. I've been looking at the config pages for the 488 and trying to make sense of the Route to PSTN configuration. Have you found any documentation for this? Keith Yoder ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream HandyTone-488, * - FXO problems
Pardon my answering myself (and for the long post). But I do have it sort of working, and I come back with information on the GS HT-488, as well as questions related to SIP / DTMF issues. The GS HT-488 acts as a PSTN pass through device for 4 rings. If the phone attached to the FXS port hasn't picked up by 4 rings, it will by default answer, and you're at an internal (*) dial tone. You can also configure the HT-488 to dial a specific extention, which it will then do instead of dropping you at an internal dial tone. From there you can obviously do what ever you want with the call. (It would be nice if you could configure and/or disable the # rings before it switches over to VoIP. Maybe that will be something they will add to a firmware update someday.) For dialing out, you set up an extention for the FXO port, and dial that. It will ring once, and then present you with the PSTN line, dial tone and all. From there you (should be) are able to dial out. Now, here is my problem and question. Both the FXS and FXO ports are set up to use SIP INFO for DTMF. You would think that when you have dialed the FXO port, and are at the PSTN dial tone, the HT-488 will translate the SIP DTMF INFO passed through to the FXO port as audible DTMF on the PSTN line. This is not the case. So I really can't make outgoing calls yet. Now, I can change the FXS line to send DTMF in audio, which works, but I figure that sending DTMF in audio is not ideal. So I'm trying to translate the SIP DTMF INFO to DTMF in-audio. I've tried a few combinations of SipDTMFMode(inband) (trying to do a DTMF style translation, I guess), and Dial(SIP/gs1-FXO,10,D(PSTNnumber) ), but can't get it to work. Should I just suck it up and keep the FXS port using DTMF in-audio, or is there a way to get SIP DTMF INFO translated to DTMF tones in audio in the Dial settings for the FXO extension? Thanks! Dan Dan Perik wrote: I just got my shiny new Grandstream HandyTone-488 today. My goal is to use it to allow incoming/outgoing calls to PSTN using my normal ole' phone as usual. I will be switching over to using BroadVoice as my main phone #, but want that to be as seemless of a switchover as possible (for the wife and kids, and for people needing to call us). I've got the following working: FXS - * ( and then - BroadVoice ) ( BroadVoice - ) * - FXS FXO - * ( and then - FXS ) I don't have this working: ( FXS - ) * - FXO In other words, I can't seem to call out on my PSTN line from Asterisk. snip ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Grandstream HandyTone-488, * - FXO problems
Title: RE: [Asterisk-Users] Grandstream HandyTone-488, * - FXO problems
You can stop trying. They still have problem with the firmware concerning the FXO port. If you really want to make a call from * out the PSTN, I suggest you to get a x100p. They are selling it on ebay for $6.99, and I have 4 of those in my * box.
-Original Message-
From: Dan Perik [mailto:[EMAIL PROTECTED]]
Sent: Tuesday, April 05, 2005 10:55 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Grandstream HandyTone-488, * - FXO problems
I just got my shiny new Grandstream HandyTone-488 today. My goal is to
use it to allow incoming/outgoing calls to PSTN using my normal ole'
phone as usual. I will be switching over to using BroadVoice as my main
phone #, but want that to be as seemless of a switchover as possible
(for the wife and kids, and for people needing to call us).
I've got the following working:
FXS - * ( and then - BroadVoice )
( BroadVoice - ) * - FXS
FXO - * ( and then - FXS )
I don't have this working:
( FXS - ) * - FXO
In other words, I can't seem to call out on my PSTN line from Asterisk.
Here's a snippet from sip.conf:
[gs1-FXO]
type=friend
context=default
host=dynamic
username=gs1-FXO
secret=mysecret
nat=no
canreinvite=yes
dtmfmode=info
incominglimit=1
disallow=all
allow=ulaw
allow=alaw
allow=g723.1
allow=g729
Here's a snippet from extensions.conf:
[gs1-fxo-out]
exten = _8.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
So when I dial, say 85429411, I would expect it to dial 5429411 out on
the PSTN line. I end up not getting any tone or other audio out of the
handset. But, using another phone directly connected to the PSTN, I
find that the Grandstream has taken the line off hook, but not dialed
any digits. I get this in my * log when I dial 85429411.
-- Executing Dial(SIP/gs1-FXS-9041, SIP/[EMAIL PROTECTED]) in new
stack
-- Called [EMAIL PROTECTED]
-- SIP/gs1-FXO-877b is ringing
-- SIP/gs1-FXO-877b answered SIP/gs1-FXS-9041
-- Attempting native bridge of SIP/gs1-FXS-9041 and SIP/gs1-FXO-877b
== Spawn extension (outgoing-ok, 85429411, 1) exited non-zero on
'SIP/gs1-FXS-9041'
I know the Handy-Tone 488 is a new device, so there may be some quirks
to it. But I would think it _should_ work.
Any suggestions?
Thanks!
Dan
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RE: [Asterisk-Users] Grandstream HandyTone-488, * - FXO problems
Title: RE: [Asterisk-Users] Grandstream HandyTone-488, * -> FXO problems
Well, the x100p is not always good either. If we forget
that it only support 600 ohm impedance, the proper example would bethe
problem i have and not being able to overcome is tremendous echo on the VOIP
phone when i make a call to pstn. after 2 months of trying i had to quit using
it.
The issue i have is that no matter what i do i never
receive the output from Asterisk saying somethig else, than "Echo Cancellation:
0 taps unless TDM bridged, currently OFF" in responce to the command "zap show
channel 1". this is the ONLY card in the pc, does not share IRQ or IO. It does
not matter what i put in config files what echo cancellation i use, it just
never ever goes to something like "currently ON". I've read a lot about echo
problem on the pstn - voip but none of the solutionare working for
me.
Sincerely,--Andyx6722"Outsourcing is akin to
making a skyscraper taller by taking material from its lower floors."--Byron
Katz
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wai
WuSent: Wednesday, April 06, 2005 9:47 AMTo: 'Asterisk
Users Mailing List - Non-Commercial Discussion'Subject: RE:
[Asterisk-Users] Grandstream HandyTone-488, * - FXO
problems
You can stop trying. They still have problem with the firmware
concerning the FXO port. If you really want to make a call from * out the
PSTN, I suggest you to get a x100p. They are selling it on ebay for $6.99, and
I have 4 of those in my * box.
-Original Message- From: Dan
Perik [mailto:[EMAIL PROTECTED]]
Sent: Tuesday, April 05, 2005 10:55 PM To: asterisk-users@lists.digium.com Subject:
[Asterisk-Users] Grandstream HandyTone-488, * - FXO problems
I just got my shiny new Grandstream HandyTone-488 today.
My goal is to use it to allow incoming/outgoing calls
to PSTN using my normal ole' phone as usual. I
will be switching over to using BroadVoice as my main phone #, but want that to be as seemless of a switchover as
possible (for the wife and kids, and for people
needing to call us).
I've got the following working:
FXS - * ( and then - BroadVoice ) ( BroadVoice - ) * - FXS FXO - * (
and then - FXS )
I don't have this working: ( FXS -
) * - FXO
In other words, I can't seem to call out on my PSTN line from
Asterisk.
Here's a snippet from sip.conf: [gs1-FXO] type=friend context=default host=dynamic username=gs1-FXO secret=mysecret
nat=no canreinvite=yes
dtmfmode=info incominglimit=1
disallow=all allow=ulaw
allow=alaw allow=g723.1
allow=g729
Here's a snippet from extensions.conf: [gs1-fxo-out] exten =
_8.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
So when I dial, say 85429411, I would expect it to dial
5429411 out on the PSTN line. I end up not getting any
tone or other audio out of the handset. But,
using another phone directly connected to the PSTN, I find that the Grandstream has taken the line off hook, but not
dialed any digits. I get this in my * log when I
dial 85429411.
-- Executing Dial("SIP/gs1-FXS-9041",
"SIP/[EMAIL PROTECTED]") in new stack -- Called [EMAIL PROTECTED] -- SIP/gs1-FXO-877b is ringing -- SIP/gs1-FXO-877b answered SIP/gs1-FXS-9041
-- Attempting native bridge of
SIP/gs1-FXS-9041 and SIP/gs1-FXO-877b == Spawn
extension (outgoing-ok, 85429411, 1) exited non-zero on 'SIP/gs1-FXS-9041'
I know the Handy-Tone 488 is a new device, so there may be
some quirks to it. But I would think it _should_
work.
Any suggestions?
Thanks! Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users
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