Joseph,
You may want to try RPA-2E1S1O from www.broad-tel.com from China. It
provides real FXO port that registers with Asterisk.
David
On Sat, Dec 12, 2009 at 1:37 AM, Joseph syscon...@gmail.com wrote:
I'm looking for a reliable ATA FXO/FXS adapter.
Linksys 3102 - a lot of echo problem + two
'#' was pressed.
Thanks.
David
On 9/24/07, Atis Lezdins [EMAIL PROTECTED] wrote:
On Monday 24 September 2007 10:21:44 VoIP Newbie wrote:
I wonder why my call was transferred when I pressed '#' in a
conversation.
How can I disable this kind of call transfer?
Thanks.
David
Take a look
Hi all,
I wonder why my call was transferred when I pressed '#' in a conversation.
How can I disable this kind of call transfer?
Thanks.
David
___
Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/
--Bandwidth and
You need a FXS to FXO converter between ATA and the GSM box. You can get one from www.broad-tel.com
On 1/13/06, Ronald Voermans [EMAIL PROTECTED] wrote:
Hi All,
I have a GSM box, which needs to connect to a analogue phone line. I've plugged the GSM box to a Grandstream ATA (386). This ATA has
Hi all,
I am trying to get DTMF digits from X-pro, through a grandstream ATA, to a FXS to FXO converter for outgoingPSTN calls. I could hear second dial-tone from the phone line connecting to the converter. However, no PSTN dialing occured after DTMF digits was sent from X-pro.I tried while
HI all,
I am wondering if asterisk supports USB phones.
Thanks.
David
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
you want something really cheap. you got to visit www.broad-tel.com. It is even offering a WiFi phone at US$125 for its existing clients.
On 12/20/05, Dakota [EMAIL PROTECTED] wrote:
Are there any IP Phones that can work with Asterisk, that cost less than $60?if so, what's the model/manufacturer?
There are4 options for your consideration:
1. use 2 x 1-port FXO gateway
2. use 2-port FXS gateway with FXS to FXO converter
3. use a 4-port FXO gateway.
4. use 2 x X100P cards
You can get them from www.broad-tel.com
On 12/21/05, Waldo Rubinstein [EMAIL PROTECTED] wrote:
I'm looking for a
PA-122TI from www.broad-tel.com supports T.38 and Fax pass-thru.
On 9/15/05, Rosario Pingaro [EMAIL PROTECTED] wrote:
about spa-2100, the t38 stream is on UDPTL and so asterisk passthroughdoesn't work.- Original Message -
From: Nenad Radosavljevic [EMAIL PROTECTED]To:
Alex,
Context solved my problem. Thank you so much.
HappyDavid.
On 9/13/05, Alex Ongena [EMAIL PROTECTED] wrote:
On Mon, 2005-09-12 at 23:34 +0800, VoIP Newbie wrote: Below are what I have in extension.conf.
Is this the complete file ? exten = s,1,Goto(1234,1) ^^^is used to jump
Hi all,
How can I makea X100P ZAP channel not answering to any incoming calls?
Thanks.
Newbie
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
to differentiate calls from a particular channel.
2. I don't know how tomake a channel no Answer.
Please advise and help.
Many Thanks,
David
On 9/12/05, Alex Ongena [EMAIL PROTECTED] wrote:
Euh, what is your extensionf.conf part that answers it ?On Mon, 2005-09-12 at 18:25 +0800, VoIP Newbie wrote:
Hi all, How
Get a 8-port FXS gateway from www.broad-tel.com. That is the single
box you need.
On 8/16/05, Roland Zagler [EMAIL PROTECTED] wrote:
Hello everyone,
I want to build an Asterisk Box where i need 8 FXS interfaces
to connect 8 phones to. The problem is, that there is only one
PCI slot
I bought 3 from 3 different vendors. One of them has echo issue.
Another one has an issue regarding PCI master abort. Only one really
works fine for me. These 3 cards use AMBIENT chip but with different
layouts and SLICs.
On 8/4/05, Mark Burton [EMAIL PROTECTED] wrote:
X101P with Ambient md3200
I have 2 OEM X100P. The one from www.broad-tel.com works fine.However,
the other one has echo. Both use MD3200 chips. Any one knows why it is
so??
On 8/13/05, Madhawa Jayanath [EMAIL PROTECTED] wrote:
Carlos Trallero wrote:
Hello,
I have asterisk running on Fedora Core 3 with a x100p
different firmware versions
from the OEM.
Mark
VoIP Newbie wrote:
I have 2 OEM X100P. The one from www.broad-tel.com works fine.However,
the other one has echo. Both use MD3200 chips. Any one knows why it is
so??
On 8/13/05, Madhawa Jayanath [EMAIL PROTECTED] wrote:
Carlos Trallero
I got one from www.broad-tel.com. It works fine.
On 8/12/05, Douglas Logan [EMAIL PROTECTED] wrote:
Yes, but your results may vary. Apparently some people have problems
with clone cards (aka regular modems), dropping calls, and having
echos. (Then again some people have reported no problems at
You may want to contact www.broad-tel.com/index_en.php. They offer a
variety of FXO SIP gateways from 2-port to 16-port.
On 8/8/05, Chris Mason (Lists) [EMAIL PROTECTED] wrote:
Has anyone found a suitable but not exorbitant 4-6 port FXO = sip
gateway? I need something more compact than a
Tim,
Can I test it as well?
Best Regards,
Newbie
On 8/2/05, Vlasis Hatzistavrou - asterisk mailing list account
[EMAIL PROTECTED] wrote:
If anyone is interested I'm (slowly) developing a GPL'd Java applet that
works as an IAX softphone.
I should have a test version out at the end of
If it is expensive to get a separate LAN connection for analog phone
adapters, you can get one with 2 ethernet port and 1 FXS port such
that
it can connect your PC and analog phone over a single cable to the
network. It is not difficult to find such kind of analog adapters for
around US$50 or
Hi all,
When I was making calls from an IP phone, through a X100P, to PSTN,
the following error was encountered.
-- Executing Dial(SIP/25086937-aa6c, Zap/1/91713545) in new stack
-- Called 1/91713545
-- Zap/1-1 answered SIP/25086937-aa6c
Aug 6 00:12:53 WARNING[3983]: chan_zap.c:4717
http://www.broad-tel.com/products/wireless.php
On 7/7/05, Ola Lidholm [EMAIL PROTECTED] wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
IM.Nobody
Sent: Wednesday, 6 July 2005 11:51 PM
To: Asterisk-Users@lists.digium.com
Would it be a good replacement of expensive WiFi phones? How much is it??
On 7/6/05, IM.Nobody [EMAIL PROTECTED] wrote:
Hi all,
Just want to share with all of you a new hot DECT VoIP gateway
available from www.broad-tel.com/index_en.php.
The DECT VoIP gateway is capable of handling both
I wouldn't mind such a single message. It is really a new breed of
product that is not known to most of us. Correct me if I am wrong.
While there have been some discussions on HOP-ON's wifi phone of $39
that never came true, this may be a sound alternative for all of us.
On 7/7/05, Terry H.
Hi all,
Is there any AGI supported calls authenticated by IP address?
Many thanks.
Newbie
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
). Make sure, they
don't have to register - host=123.123.123.123 instead of
host=dynamic.
Julian.
On 7/5/05, VoIP Newbie [EMAIL PROTECTED] wrote:
Hi all,
Is there any AGI supported calls authenticated by IP address?
Many thanks.
Newbie
txgain/rxgain in zapata.conf.
On 7/3/05, wassim darwish [EMAIL PROTECTED] wrote:
i noticed that the sound volume of the zap(tdm400p)
was low ,so i tried to raise the sound volume but i
didnt know how please help me.
__
Do You Yahoo!?
Tired
The one that looks identical is selling at $180 from
www.broad-tel.com/index_en.php
On 7/1/05, Richard Malcolm-Smith [EMAIL PROTECTED] wrote:
If it does materialize, im up for 3 or 4 of them at that price.
Huddleston, Robert wrote:
Well poo - if I can use that word I'm one of those poor
Well, I found a WiFI phone that looks identical to the Hop-On one. It
is from www.broad-tel.com/index_en.php but is selling at $180/each for
every 20 units.
On 6/29/05, William Suffill [EMAIL PROTECTED] wrote:
Unfortunately no. Someone say the press release and bugged me about it
as well but I
My one from www.broad-tel.com works fine and is very cheap.
On 7/1/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Yes, I have :-)
3 of this cards running well on my personnal *
What price for your ?
Best Regards,
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL
The supplier is from www.broad-tel.com
On 6/14/05, Jian Hong GUAN [EMAIL PROTECTED] wrote:
That interests me. Can you send me the informations about products and
suppliers?
Best regards,
--Hong
___
Asterisk-Users mailing list
There is an USB softphone or MP3 softphone that you may find useful.
The USB softphone in size of USB flash disk comes with built-in sound
drive. It can embed a softphone such that it is portable anywhere even
an computer does not equip with a sound card. It is also a USB flash
disk that can be
Please visit www.broad-tel.com for details.
On 6/8/05, Wai-Sun Chia [EMAIL PROTECTED] wrote:
On 6/8/05, VoIP Newbie [EMAIL PROTECTED] wrote:
My 4-port FXO is only $300.
Which product/model are you using then?
/wai-sun
___
Asterisk-Users
I am using Asterisk CVS-HEAD-06/02/05-19:37:27. It seems that every
call I made was duplicate.
Jun 8 00:11:30 DEBUG[21733]: chan_h323.c:411 oh323_call: Placing
outgoing call to 87874586, 101
-- Called 87874586
Jun 8 00:11:31 DEBUG[21733]: rtp.c:472 ast_rtp_read: RTP NAT: Using
address
My 4-port FXO is only $300.
On 6/8/05, Adrian A [EMAIL PROTECTED] wrote:
From your experience, would you recommend purchasing 8 Sipura 3000 1
port FXO gateways or 1 Audiocodes 8 port FXO gateway?
The way I see it, the advantage of going to the Sipura solution is
that it is more scalable (ie.
Another alternative is to use H.323 FXS IAD in combination of H.323
channels. I bought a 4-port IAD of US$50 per port. It works for me!!!
Let me know if you will be interested in the product.
On 6/2/05, Waldo Rubinstein [EMAIL PROTECTED] wrote:
I'm looking for an inexpensive way to connect 20
, the asterisk server will be behind
another NAT.
Thanks,
- Waldo
On Jun 2, 2005, at 3:23 AM, VoIP Newbie wrote:
Another alternative is to use H.323 FXS IAD in combination of H.323
channels. I bought a 4-port IAD of US$50 per port. It works for me!!!
Let me know if you will be interested
Hi,
I am using chan_h323 from CVS. An incoming call from H323 phone caused
the following error:
Jun 2 20:07:08 WARNING[1166]: rtp.c:457 ast_rtp_read: RTP Read too short
Any idea for the above message??
Thanks.
Newbie
___
Asterisk-Users mailing list
Does it support pre-paid billing?
On 5/30/05, Darren Wiebe [EMAIL PROTECTED] wrote:
El Flynn wrote:
Darren Wiebe wrote:
Good Day,
I'm finally getting around to officially announcing ASTPP. For the last
6 months, I've been working on converting ASTCC into a decent billing
package
Read README file first. You will get a clue.
On 5/19/05, FaberK [EMAIL PROTECTED] wrote:
Hello Guys,
first of all, I'm very new with asterisk.
I'm trying to set it up. I've already compiled and installed Asterisk-1.0.7
Now I'm trying with asterisk-oh323
I've already installed pwlib, oh323
Can anyone give me a big hand here??
On 5/16/05, VoIP Newbie [EMAIL PROTECTED] wrote:
Hi all,
I am using chan_h323 from Asterisk CVS to interconnect with GNUGK
v2.2.2. Then I made call from a H323 EP, thru GNUGK, to SIP EP on
Asterisk. However, I only heard ringing when the call
Can anyone give me a big here?
On 5/13/05, VoIP Newbie [EMAIL PROTECTED] wrote:
I am using Asterisk-oh323 v0.7.1 with GNUGK. Please advise what must
be done to make FastStart work with SIP phones. Thanks.
On 5/12/05, VoIP Newbie [EMAIL PROTECTED] wrote:
Hi all,
When I enabled
You are wasting your time while you can get an OEM X100P for a few
dollars. Check it out at eBay or www.broad-tel.com.
On 5/18/05, ALIF Mohssine [EMAIL PROTECTED] wrote:
Hello Dave,
Could I know why please ?? Thanks !
Dave Cotton [EMAIL PROTECTED] a écrit:
On Wed, 2005-05-18 at 11:21 +0200,
CLI show module
and look for chan_oh323.so
If oh323 is loaded, oh323 show conf will provide more useful info.
On 5/17/05, Micko [EMAIL PROTECTED] wrote:
Hello!
How can I check if oh323 is loaded and working? Is there a quick test for
this?
Thank you.
Micko
Below happened when I am using Asterisk-oh323 0.7.1 with FastStart
enabled. I made calls from H323 EP to SIP EP. After a long ringing at
the orginating H323 EP, * was aborted as follows. Any help???
*CLI 2:40.912Housekeeper PWLib Assertion
fail: Function ::close failed,
Hi all,
I am using chan_h323 from Asterisk CVS to interconnect with GNUGK
v2.2.2. Then I made call from a H323 EP, thru GNUGK, to SIP EP on
Asterisk. However, I only heard ringing when the call was answered on
SIP side. Below is the debug from chan_h323. Any help is welcome.
Thanks.
*CLI ==
I am using Asterisk-oh323 v0.7.1 with GNUGK. Please advise what must
be done to make FastStart work with SIP phones. Thanks.
On 5/12/05, VoIP Newbie [EMAIL PROTECTED] wrote:
Hi all,
When I enabled faststart in oh323.conf, calls from H323 endpoint to
SIP phones could not complete
Hi all,
When I enabled faststart in oh323.conf, calls from H323 endpoint to
SIP phones could not complete. The originating phone kept ringing when
calls were answered by SIP phones.
fastStart=yes
h245Tunnelling =yes
h245inSetup=yes
Please can you advise.
Many Thanks.
Hi all,
I could register * to a provider. However, I failed to make outgoing
calls through the provider. Please help and advise how to get it work.
m2*CLI sip show registry
HostUsername Refresh State
sip_proxy:5060 abc105 Registered
Sorry, I just fixed it by myslef. It is an issue of incompatible
codec. I am wondering why option t in dial() is not able to make it
work.
Any advice??? Many Thanks.
On 5/6/05, VoIP Newbie [EMAIL PROTECTED] wrote:
Hi all,
I could register * to a provider. However, I failed to make outgoing
My one from www.broad-tel.com works perfectly.
On 4/11/05, Sahil Gupta [EMAIL PROTECTED] wrote:
I'm having similar issues using an X100P Ambient Chipset Clone Card
any ideas?
Regards,
Sahil Gupta
VoiceValley
On Mon, 11 Apr 2005, Dave Weis wrote:
I've got a X100P in a
You can get one at around US$6 from www.broad-tel.com, including
installation instruction.
On 4/5/05, Tore Hansen [EMAIL PROTECTED] wrote:
You do need a proper FXO card to connect your POTS line
However, that need not be expensive. A suitable card is
available by mail order in the U.S. from
] wrote:
Hi all,
Did I make my issue clear? Can any one give me a big hand?
Many thanks.
Newbie
On Apr 5, 2005 12:59 AM, VoIP Newbie [EMAIL PROTECTED] wrote:
Hi all,
When I made calls from SIP phones through a analog PSTN gateway to
PSTN phones, I could hear rings twice on my
Hi all,
Did I make my issue clear? Can any one give me a big hand?
Many thanks.
Newbie
On Apr 5, 2005 12:59 AM, VoIP Newbie [EMAIL PROTECTED] wrote:
Hi all,
When I made calls from SIP phones through a analog PSTN gateway to
PSTN phones, I could hear rings twice on my SIP phones. From my
Hi all,
When I made calls from SIP phones through a analog PSTN gateway to
PSTN phones, I could hear rings twice on my SIP phones. From my best
guess, the first one from * and the second one from analog PSTN line.
Am I right? Is it configuration related? Can the two rings be reduced
to a single
Hi all,
How can I configure chan_h323 or chan_oh323 to register to multiple GK
and route calls in-between?
Many thanks.
Newbie
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To
Hi all,
I am wondering if chan_oh323 or chan_h323 supports NAT traversal the
following setup:
H323 phone - Asterisk --- NAT router - H323 gateway - PSTN
I am trying to register a H323 gateway through a NAT to Asterisk for
outgoing calls to PSTN.
How can I achieve the above?
depends also on whether the router can do port
forwarding and whether the H323 Gateway supports NAT.
This is possible with Quintum for instance with some port forwarding rules on
router level.
Selon VoIP Newbie [EMAIL PROTECTED]:
Hi all,
I am wondering if chan_oh323 or chan_h323 supports
58 matches
Mail list logo
