the X100P do
you get dialtone? The second port is hardwired to the first port, so if
you don't get dialtone on the second port then the phone line you have
plugged into the X100P is not working.
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS h
n either some of the traces on the board are broken or the line is
not working. I assume you've confirmed the line if fine by plugging an
analog phone directly into the line?
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled
> I was having the same problem as well as few other users that I know of.
> The registration issue seems corrected but we are having problems with
> our customers not receiving ring back tone when calling us.
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a rela
On Tue, 2004-05-18 at 10:13, Alberto Sato wrote:
> How I can do to X100P (FXO port) answer in the first Ring?
usecallerid-no
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be ded
ore
> dump, gnore?
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
--
calling"
As you can see at http://www.grandstream.com/Product_Spec.pdf the BT102D
is the only GS phone that supports 3-way calling.
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT
amp;lr=&ie=UTF-8&q=site%3Alists.digium.com+%22Getting+info+about+changes+in+CVS%22&btnG=Search
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of Windows
upgrade
e from the Caller*ID of the phone making the call
and use app_goto to jump to $AreaCode . $Exten or something like that.
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be
On Tue, 2004-05-18 at 23:46, Eric Wieling wrote:
> On Tue, 2004-05-18 at 23:26, Randy Bush wrote:
> > i have extensions in locations across a number of telco area codes.
> > when someone in seattle picks up and dials 91234567, it would be
> > nice to transform it to 9206123
list at http://lists.digium.com/
I asked why the rtp.c fixes for bad audio were not put in -stable and I
got this response:
http://lists.digium.com/pipermail/asterisk-users/2004-April/043296.html
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IR
ten to death already.
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss."
___
Asterisk-Users maili
On Wed, 2004-05-19 at 11:37, Paul Tyreman wrote:
> The problem that I have now is that when a call comes in on the
> external phone line and it is answered by a standard phone (not
> connected to Asterisk), the IP Phones that are connected to it ring
> for another two rings.
> This is very irritati
and it stil doesn't work. Is there something somewhere else that needs to
> be set to make this work properly, like may in the sip device itself?
SIP devices do not support ignorepat. The PHONE is providing dialtone,
not Asterisk for SIP devices. It's one of the dirty little secret
ures then you need to buy a Digium card (the X100P will
work and it's around $100).
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be dedu
X100P will
> > work and it's around $100).
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk
XXX,1,Dial(${PSTN-1}/${EXTEN})
> exten => _NXXNXX,1,Dial({PSTN-1}/1${EXTEN})
You're missing a $ on the second dial line on {PSTN-1}
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of W
guay (S. America)
> Phone:595-21-440104
> Fax: 595-21-440270
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update optio
be taken more seriously.
>
> ?
>
> -reed
>
>
>
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
> http:
CE[1232452528]: channel.c:1508
> ast_set_read_format: Unable to find a path from G729A to ULAW
>
> May 20 18:20:47 NOTICE[1232452528]: channel.c:1478
> ast_set_write_format: Unable to find a path from SLINR to G729A
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"
s/2003-November/027739.html
http://lists.digium.com/pipermail/asterisk-users/2003-November/027806.html
http://lists.digium.com/pipermail/asterisk-users/2004-February/035638.html
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that
mp;ie=UTF-8&q=site%3Alists.digium.com+uip200&btnG=Google+Search
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss."
_
On Thu, 2004-05-20 at 23:10, John Vogel wrote:
>
> Has anybody used these with Asterisk?
http://www.google.com/search?hl=en&ie=UTF-8&q=site%3Alists.digium.com+SwissVoice&btnG=Google+Search
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a rel
ce the rtp.c hack and all was back to normal again.
The fix in CVS -stable was about 10 HOURS ago.
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted a
[EMAIL PROTECTED] wrote:
I am still having this problem of only capturing part of the IP address, I
am currently checking into a possible hardware/software issue on the
client side but was wondering if there are any setting I need to set on
the asterisk server to allow an peer to peer call. I have
Adam Goryachev wrote:
Based on the following diff in CVS:
-" 't' -- allow the called user transfer the calling user\n"
-" 'T' -- to allow the calling user to transfer the call.\n"
+" 't' -- allow the called user transfer the calling user by
hitting #.\n"
+" 'T' -- allow the call
Scott Stingel wrote:
Hi-
I have an upcoming application that requires use of PRI channels that are
primarily used for high-volume incoming traffic, but that are to be used for
outbound calling as well. Of course, one option is to have dedicated
outbound channels reserved, but this is an inefficien
Bruce Komito wrote:
I have a PRI with a bunch of DID numbers on it. When I dial one of the
DID numbers from the outside, the call is correctly routed, either to the
auto-attendant or to the correct extension. However, all the caller hears
until the call is answered is silence, i.e., no ringing.
nking=yes (or trunk=yes) in iax.conf but do not have a
Zaptel/Digium card in the system. You cannot use IAX trunking without a
Zaptel/Digium device.
> how can I get rid of these error messages first?
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story,
gt; who's problem is that?
Unless, of course, the person is writing about NOTHING that has to do
with the original thread. That was the case here. The Subject: was
"Voicepulse", but his questions were about specific non-Voicepulse
Astersik issues
--
Eric Wieling * BTEL
I get:
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss."
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://
M})} = 4]?f,1)
exten => external,2,SetCIDNum(${ARG1},a)
exten => external,3,GoTo(f,1)
exten => f,1,NoOp(Variable CALLERIDNUM is equal to ${CALLERIDNUM})
exten => f,2,NoOp(Leaving Macro "callerid-fixup")
--
Eric
Fixed an error in the macro The line:
exten => external,1,GoToIf($[${LEN(${CALLERIDNUM})} = 4]?f,1)
should be changed to
exten => external,1,GoToIf($[${LEN(${CALLERIDNUM})} != 4]?f,1)
--Eric
On Sat, 2004-05-29 at 02:49, Eric Wieling wrote:
> Here is a macro I wrote to deal with s
isk/asterisk/cdr/cdr_pgsql.c: if (PQstatus(conn) !=
CONNECTION_BAD) {
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss."
___
on the LineJack
To search the Asterisk mailing list archive go to www.google.com and put
site:lists.digium.com in addition to your other query terms.
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of Windows
upgrades
On Mon, 2004-05-31 at 09:44, Jeff Sczpel wrote:
> [/etc/zaptel.conf]
> span=1,1,1,esf,b8zs
span=1,0,1,esf,b8zs
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as
[general] and
didn't realize that incoming SIP calls that didn't match anything would
be accepted and sent to the context= that was specified in [genera].
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recentl
%3Alists.digium.com+sipura+noise&btnG=Search
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss."
___
Aste
ery common.
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss."
___
Asterisk-Users mailing list
[EMAIL PR
oes that mean they can't provide Zaptel timing like the X100P can?
The TDM400P takes one IRQ. The modules for the TDM400P do not take up
any additional IRQs
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of
ing them with PRIs. One of the biggest reasons people use PRIs
is so they don't have this problem. I don't know what to suggest to
you, other than not to give up. This is a fixable issue with PRIs.
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related s
is configured to
use (and waiting for) Caller*ID. See the usecallerid= and callerid=
options.
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss."
___
On Wed, 2004-06-02 at 07:41, Sergio Serrano wrote:
> Hi all, I try to do next transfer:
> A person contact with me, I would like transfer to other person
> in next manner. I call to other person and when I say who wants talk
> with him I hangup phones an call is redirect automatically to othe
df
Other IP phones like the Cisco DO support 3-way calling and support
supervised/consultative transfers (which is the term for what you want
to do)
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of
up
> > -- Hungup 'Zap/2-1'
> >
> > What have I to change? Could I supress that?
> >
> > Thanks
> >
> > Felix Deierlein
> >
> > ___
> > Asterisk-Users mailing list
> > [EMA
o how does one obtain the drivers? Or is
> it a better pots adapter for the wastebasket?
http://www.asteriskpbx.org/index.php?menu=hardware
http://www.google.com/search?hl=en&ie=UTF-8&q=site%3Alists.digium.com+dialogic+drivers&btnG=Google+Search
--
Eric Wieling * BT
sed anyone can survive without this feature.
They don't use Grandstream phones or they work around the issue if they
do use Grandstream phones. In the analog world everything is handled by
the PBX. In the SIP world most things are handled by the phone.
--
Eric Wieling * BTEL Co
, but Now it doesn't work.
"show application dial" on the Asterisk console. Pay attention to the
"t" and "T" options to enable # transfers.
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that
27;ve seen elsewhere -
> 0x00460400, and 0x01a40400)
> NatTimer: 0x0054000a
This is the standard config we use for ATA-186s using v2.16 firmware:
http://www.fnords.org/~eric/asterisk/ata-186.shtml
We are slowly migrating to the 3.1 firmware, but the settings are very
similar.
--
Eric
need them rather than using a digital PBX]). I'll have
> to use them now I've spent so much time/money on them
>
> For myself I'd love to play with a Cisco but with the 7960 going for £600 a
> throw I'm not in the market for spending that much...
>
&g
On Thu, 2004-06-03 at 05:27, Tony Hoyle wrote:
> Eric Wieling wrote:
> > Why are you even looking at VoIP? Analog ports and phones are pretty
> > cheap. They are not "pretty", but they are cheap and all the smarts are
> > in the PBX.
>
> Free calls to the US
think it decreased latency on the call, but I'm not sure. Regardless
the option was removed some time ago.
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gamblin
nly
> answer? thanks.
"show application waitforring"
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss."
__
query terms.
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss."
___
Asterisk-Users mailing list
[
risk/ata-186.shtml
>
> Any ideas?
>
> _______
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options
The "CANCEL" sip messages seem to fail also, as they are
> > retransmitted many times. I'm running the ATA conf from:
> > http://www.fnords.org/~eric/asterisk/ata-186.shtml
> >
> > Any ideas?
> >
> > _______
essages you are seeing. Causes of interrupts being locked for too long
can be: graphics, frame buffer, IDE DMA. There might be other things
that cause this error, I don't know.
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ru
or 500
> Jun 3 02:41:11 NOTICE[11276]: PRI got event: 6 on span 1
> Jun 3 02:41:11 WARNING[12301]: Detected alarm on channel 1: Red Alarm
I answered this same question *today*:
From:
Eric Wieling <[EMAIL PROTECTED]>
Reply-To:
[EMAIL PROTECTED]
To:
[EMAIL PROTECTED
> > testing with allow of ulaw only, then try turning on other codecs. We
> > know
> > > that one works well.
> > >
> > >
> > >
> > >
> > > Timothy R. McKee
> &g
a month or so ago. I haven't tested the fix yet.
What IOS version contains the fix for QoS on Ethernet?
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be
X${RDNIS} != X]3009,4)
exten => 3009,2,VoicemailMain()
exten => 3009,3,Hangup
exten => 3009,4,VoiceMail(u${RDNIS})
exten => 3009,5,Hangup
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of
On Fri, 2004-06-04 at 18:16, Maveric wrote:
> What type of cisco phones? i'm using 7960's and i know they don't have a
> to voice mail button. That annoys me.
Cisco 7905G is the one I had for a while.
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
&qu
On Mon, 2004-06-07 at 07:14, Duane wrote:
> exten => _.,1,Dial(IAX2/username:[EMAIL PROTECTED]/number)
As far as I can tell Asterisk will not use iax.conf if you do it that
way. (specifically if you use @host rather than @iaxconfentry)
--Eric
--
Eric Wieling * BTEL Consulting
})
> exten => _NXXNXX,3,Congestion
>
> The above will attempt to dial out your Zap interface first. If that
> fails, it will dial out using "username" for the username and the
> password, IP address info for the IAX2 peer will be grabbed out of the
> iax2.con
_______
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
--
Eric Wieling * BTEL C
___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
--
Eric Wieling *
>
> > brian wrote:
> >
> > >*SMACK* no you don't just use the native sip transfer to park it. :)
> > >
> > >bkw
> > >
> > >
--
Eric Wieling * BTEL Consulting * 5
tp://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related s
Karl J. Vesterling wrote:
It would be nice to have an option where dialing two #'s would send a
single #.
I too have had difficulty with this.
My workaround is to use my cell when calling an IVR.
ANother workaround I have through of would be to use 99 for outgoing,
but with no transfer options i
Remove the evil "r" option from the Dial line in Asterisk.
Mathieu Nantel wrote:
Good day,
I've got around to installing an X100P card in my computer to try out
asterisk. I noticed (and people who were testing with me also noticed) that
when dialing from my SIP soft phone to the PSTN, the ringer
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://
-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, t
mb Cards - all seem to cause the problem) +
> i4l + outgoing DTMF working at all? What Version of Asterisk?
>
> So far - people who I've asked say "No"
http://www.google.com/search?hl=en&lr=&ie=UTF-8&q=site%3Alists.digium.com+i4l+dtmf+patch&btnG=Search
--
These are the three cheap SIP phones that I've used.
Grandstream BT10x $65/street
Number only LCD
Zultys ZIP 2 $100/retail
No LCD
Uniden UIP 200 $120/retail
PoE, built-in switch
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS ha
o make it work. Not many people use Dialogic cards with Asterisk so
you'll not get much support from the Asterisk community. Contact Digium
if you want to buy the drivers for the card.
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has rece
as to .if the number is busy or invalid there is no way we
> can tell this to the switch because it has already been told
> that the phone is ringing
>
> I am using asterisk Asterisk CVS-04/06/04-10:46:21 with T410
> PRI card connected to a Nor
Olson, Dana wrote:
Are there any cards that work with * that do the VoIP-to-TDM processing
on the cards, with multiple interfaces?
The QuickNet Internet LineJack meets the description I believe, but it
only has a single FXS or FXO. Are there any cards that have more than
one FXS?
It's been a
Andrew Kohlsmith wrote:
On January 20, 2005 02:15 pm, Steve Clark wrote:
I am dialing from one zap channel to a second zap channel. Is there a way
to keep the channel I am dialing to from generating a ringback tone.
exten => 1,Dial(Zap/1)
should not generate ringback...
exten => 1,Dial(Zap/1,,r
Michael Baird wrote:
It's only one guy who seems to attack each poster for not posting in a
manner of which he approves (there is one/two of these fellows on every
mailing list), don't let him ruin your day, this list is quite helpful
and many guys will give you a good answer without the extra atti
Olson, Dana wrote:
Actually, I do care, and I did search Google (albeit quickly) and I did look on
the hardware list as well as the VoIP wiki. Maybe one of the cards listed there
does what I need, but it wasn't listed like the QuickNet cards are. I thought
perhaps the feature list on the site wo
[EMAIL PROTECTED] wrote:
Hi, i cant make a three way call using grandstream phones (BT-100) and
asterisk using sip, is this supported or i need a zap interface?
The BT101 cannot to supervised transfers or 3-way calling.
___
Asterisk-Users mailing list
Ast
Paul Rodan wrote:
The BT100's do support conferencing, most SIP phones do. But how does your
Asterisk connect you to the PSTN? Through a Zap interface? If so, what kind;
or through a VoIP provider like BroadVoice, NuFone, LookieLoo, VoipJet,
VoicePulse?
You basically need to make sure your Asteri
MJ wrote:
I'm having problems using the following.
[sip]
exten => _*4.,1,macro(test,${EXTEN:2},${CALLERIDNUM})
[macro-test]
exten => s,1,Answer
exten => s,3,Flash
exten => s,3,Dial(SIP/${ARG2},30,t)
exten => s,4,Dial(SIP/${ARG1},30,t)
exten => s,t,Hangup
exten => s,i,Hangup
exten => s,h,Hangup
Kenneth Long wrote:
You really do not want to run Asterisk and X-Windows
on the same box.
That I understand... but this is not a production
machine. Loading is not an issue. I'm using icewm.
are there any other issues, besides loading, to not
run
x-windows at the same time?
Actually the issue seem
David Shaw wrote:
exten => 510,1,Dial(SIP/510,20)
exten => 510,2,Voicemail,510
exten => 8500,1,VoicemailMain
exten => _NXX,1,Dial(${TRUNKL4}/${EXTEN})
exten => _NXX,2,Dial(${TRUNKL2}/${EXTEN})
exten => _1NXXNXX,1,Dial(${TRUNKL4}/${EXTEN})
exten => _1NXXNXX,2,Dial(${TRUNKL2}/${EXTEN}
Me wrote:
Can someone give me a clue as to why I keep hearing DTMF type beeps on
my phone calls. It sounds exactly like someone on the other end is
pushing a key on their phone but they are not!
Has anyone ever heard of this before? It use to happen once in a while,
today it's been happening a
Paul Rodan wrote:
Using FireFly, all other codecs but G711 Ulaw is selected. But whenever
I place a call, I get:
Jan 24 16:07:06 WARNING[30654495]: dsp.c:1468 ast_dsp_process: Inband
DTMF is not supported on codec G.711 u-law. Use RFC2833
Umm, wtf? I thought Inband was ONLY supported o
Mark Phillips wrote:
Getting nowhere with Digium support. Trying to tell me that their
engineers are working on it and that it could be months.
Ask if you can ship them the box so they can actually reproduce the
problem.
___
Asterisk-Users mailing list
Daniel Nyström wrote:
Is it possible to turn off DTMF recognition (and all transfer services etc.)
pending on CallerID (or FXS channel)?
Some of the FXS channels I will setup soon, is going to work exactly like POTS.
It will be used by people not knowing their within Asterisk.
They will be pretty
Florian Overkamp wrote:
Hi,
-Original Message-
Is it possible to make the telco send an busy signal when an
incoming call are supposed to dial a group which has all lines busy?
Since I will get many public phonenumbers into my E1 (from
telco), it will be sliced up into a few groups. Th
Andrew Kohlsmith wrote:
On January 24, 2005 11:21 pm, Bobby Lacey wrote:
Yes I have it there. Here is my iax.conf
[NuFone]
type=user
secret=pass
context=inbound
That is a user entry. I'm looking for a peer entry, which is used when
placing outbound calls.
If he's dialing via Dial(IAX2/username
David Shaw wrote:
Out of the blue extension 100 will ring once. This will happen 3-4 times
a day. I have checked the logs and no incoming calls. I have extension
100 and 101 going to a SPA-2000 ATA. Any ideas on this??
As far as I can tell that means the phone lost it's connection to
Asterisk, and
Adam Robins wrote:
1. I rebooted the server. Still NG
2. I manually deleted the asterisk executable in /usr/sbin/. I then did
a make clean, make, make install. The executable was replaced, but
STILL shows version 1-0 12/21/04.
delete the .version file in the Asterisk source and rebuild/install
_
Adam Robins wrote:
The TE410P is a T1/E1 card. I need the card for POTS lines. Is there
also a TDM410P that does not appear on the Digium web site?
The TDM400P only works in standard PCI 2.2. Not PCI-X, not
PCI-Extreme, not PCI-64bit.
--Eric
___
Ast
Wilson Pickett wrote:
L( ) option is
applicable here? And, if your version of Asterisk doesn't have a Dial app
with the L( ) option,
will it be worth your while to upgrade to have the L( ) option?
It should be in 1.0.0 and later. 1.0.5 is the current verison.
_
Direct from the source (1.0.x)
Here is a list of what module to use with what hardware
Module Name Hardware
tor2T400P - Quad Span T1 Card
E400P - Quad Span E1 Card
wct4xxp TE405P - Quad Span T1/E1 Card (5v version)
TE410P - Quad Span T1/E1 Car
Gene Naden wrote:
When trying to start asterisk with a four-port FXS/FXO card I get the
following error: "Unable to Specify Channel 1 - no such device or
address" (Channel 1 should be the first FXO port). I looked in the
asterisk-users mailing list archive and found one person who had this
prob
Adam Robins wrote:
There is no .version file anywhere in the /usr/src/asterisk tree
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
Wieling
Sent: Tuesday, January 25, 2005 2:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject
I am getting to my wits end with these phones (and so is my boss). I am
getting an random echo on these phones and I have an issue opened with
Polycom and its been in their research and development department for
almost a month with no results.
The Polycom Admin manual says that the echocan on t
Vahan Yerkanian wrote:
Was wondering if there are any news on the native MoH patch for
1.0.3/1.0.5.. or this still works on CVS HEAD only?
Since 1.0.x is for bug fixes only and not new features, I doubt that
patch will ever be in 1.0.x.
___
Asterisk-Use
B. J. Bomar wrote:
While your solution may work now, it might not work on the next release as
both incominglimit and outgoinglimit are deprecated. Here is an idea on how
to use SetGroup and CheckGroup using your template as an example.
exten => 1051,1,SetGroup(${EXTEN})
exten => 1051,2,CheckGroup(
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