I have a Dish 301 receiver that will not display CallerID when connected
to FXS module on TDM400. Uniden phone connected to the same FXS module
does display CallerID.
When Dish 301 receiver is connected to IAXy CallerID is displayed
properly.
Any suggestions on getting the CallerID to display on
http://snakesonaplane.varitalk.com
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Does the CLEC do it based on the CIDNum that you send?
On Tue, 2006-03-14 at 14:54 -0600, Peder @ NetworkOblivion wrote:
> Not to be a smarta**, but you have to ask them to do it. We do the same
> thing and it works for us. Depending on the CLEC, they may do it or
> they may say no. If they
Central business location has a PRI with a CLEC. Remote offices access
the PRI for all voice traffic via VoIP.
How does one get the telco to report the address of a remote office to
the 911 call center when the call is made from that respective location?
_
Anyone have any luck RMAing a Sipura phone since the Cisco take over?
Sipura only has support via email or fax to end users and I haven't
gotten a response to either for over 2 months.
Linksys Support will jump you through all their scripted hoops to
resolve your problem (they hope if they speak w
This worked:
exten => NXXNXX,2,Set(CALLERID(num)=88)
>
> CLI*> show function CALLERID
>
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It worked, but...
SetCIDName() & SetCIDNum() are depreciated;
I figured SetCallerID() is on the way out, too.
I'd rather just touch part than have to mess with the whole.
On Thu, 2005-12-01 at 13:35 -0500, Tom Hayden wrote:
> Why aren't you using the SetCallerID() cmd?
Didn't work.
On Thu, 2005-12-01 at 10:36 -0800, Innocent Evil wrote:
> Set(CALLERIDNAME="Innocent Evil")
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What do I need to do to alter incoming CallerID? The below isn't
working...
Running Asterisk 1.2 CVS HEAD
exten => NXXNXX,1,Wait(1)
exten => NXXNXX,2,Set(CALLERID(name) = "Fred")
exten => NXXNXX,3,NoOp(${CALLERID(name)})
-- Executing Wait("IAX2/A-9", "1") in new stack
-- Executing
Ending last year I used\sold several hundred of product#: FM-INL92SW.
Google for it...you'll find some for cheap.
On Tue, 2005-11-15 at 09:53 -0900, Mojo with Horan & Company, LLC wrote:
> From the wiki, X100P (and clone) cards are "Single port FXO card based
> on an Intel V.92 537 or MD3200 s
A Linksys WRT54G runs for about $60. I think it supports QoS out of the
box, but flash it with 3rd party firmware (i.e. Sveasoft) to get a bunch
of extra features.
Note: The latest version, version 5, of the WRT54G only has half the
memory of the previous versions and there is no 3rd party firmw
A call comes in I set the distinctive ring by setting variable
ALERT_INFO then dial a SIP channel. The channel is answered, but then
the user forwards the call to another SIP channel. ALERT_INFO is still
set.
How can I clear the ALERT_INFO variable after the SIP channel is
answered so that whe
That worked. The following line also got rid of "asterisk" without
entering any custom info:
callerid=
Thank you,
Hugh
On Thu, 2005-08-11 at 03:19 +0100, Tony Hoyle wrote:
> In the [default] section of sip.conf put:
>
> callerid=unavailable
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Ast
I am using Sipura 841 phones and Asterisk CVS-v1-0-06/14/05.
Whenever a call comes in with blank CIDName or CIDNum the phone reports
the respective variable as "asterisk".
I can manually set the variables to whatever I want: CIDName
(alpha-numeric) & CIDNum (Numeric). But if I try to make them b
> I don't know if I'd recommend them. There is a lot of personal
> preference involved and the 841 certainly seems to be able to provoke
> strong dislike for people.
I switched my business from our old PBX to Asterisk 3-4 months ago.
Some folks hate "Asterisk"...I've attempted to convince them
Time to find a new ISP. Why did they block the ports?
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I'm having sound quality problems on the remote side with anything
higher than 3.1.2(d).
3.1.3(a) oscillates and is just too quiet.
the pre-release of 3.1.4(a) is "staticy" according to multiple folks
that I called.
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Aste
Anyone out there tried going from cell to Motorola 02527MOT to FXO with
Asterisk?
What other kinds of fixed mobile service are available for use with
Asterisk?
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The Sipura 841 has the capability of downloading ringtones on the fly
via tftp. Unfortunately they only have a windoze program (downloadbale
from their support page) to create their .srt ringtones.
Does anyone know how to create these .srt files using sox or some other
linux program?
_
Twas an issue with the card. I tried a different TDM20B and it worked
perfectly. RMA time.
> Whenever I load wcfxs I get "ProSLIC 3210 version 2 is too old."
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I have several toll free numbers that get forwarded to a single local
number assigned to a trunkgroup. I've asked the telco to not forward
those toll free numbers but to assign them as DIDs to the trunkgroup, so
that I can differentiate via DNID.
They said that they can't do that. That toll free
Look in modules.conf and you will see that wctdm is just an alias for
wcfxs.
On Thu, 2005-06-02 at 11:17 -0600, Rich Adamson wrote:
> The TDM card uses wctdm (not wcfxs).
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I have been using an old DELL Dimension XPS as my asterisk server. It
has been using a Sangoma A101 for PRI.
I tried adding a TDM20B. Whenever I load wcfxs I get "ProSLIC 3210
version 2 is too old."
Is this because my board is PCI 2.1, or a DELL, or interrupt issues, or
all of the above?
Gener
Lookup "negated character class" in your library of RegEx books.
> Not sure which RegEx book you read but ^ (caret) stands for "line beginning"
> not "don't match".
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Does ^ work as a NOT in an expression for extensions?
Are the following equivalent?
exten => _58[^389],1,dial(${${EXTEN}},${RINGLONG},tr)
exten => _58[0124567],1,dial(${${EXTEN}},${RINGLONG},tr)
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Probably not authenicated. Checked debugging to see if IAX was
authenicated or not.
On Mon, 2005-05-16 at 20:00 -0400, John L. Magee wrote:
> Trunk between two Asterisk systems does not pass Callerid. Both
> systems use Voicepulse Connect and do pass callerid over that
> connection.
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Works great for me also.
On Mon, 2005-05-09 at 14:33 +0200, Altus Snyman wrote:
> How well does the sangoma cards work with fedora core 3
> Im doing the research on what hardware/os I need to use
> Please help and advice
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I agree.
> The Sipura SPA-841 is a nice "compromise" between the Ciscos and the
> Grandstreams.
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