On Tue, 15 Mar 2005 11:56:18 -0500
Fabian Borot [EMAIL PROTECTED] wrote:
Hello all
I have been learning * from almost 1 month now. It looks
really powerfull. I
have some problem trying to find previous post, or
solutions to common
problems, advice to newbies etc in this mailing list.
There is
On Tue, 15 Mar 2005 14:50:38 -0700
Daniel Webb [EMAIL PROTECTED] wrote:
On Fri, Mar 11, 2005 at 11:50:01PM +, Jay Milk
wrote:
Dude, where have you been? This has been discussed here
at length.
Everyone agrees that it's on LiveVOIP's end, but they're
shrugging their
shoulders and pointing
Good morning all,
I have been trying to research of to change the ring frequency for the
TDM400 FXS port. I have several newer phones that will start to ring and
then quit intermittently. I have tried boosting the voltage using
boostringer=1 and that has not helped. I did verify in dmesg that
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Lyle Giese
Sent: Sunday, March 20, 2005 11:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] wctdm fxs ring frequency
Robert Webb wrote
I have searched the list and the wiki and have seen references to
changing this in the wcfxs.c file but I am not using that. Likewise,
I
have not founf anything in by looking into the wctdm.c file. I am no
programmer but can somewhat follow the code.
This is one of the best kept
SNIP
SigMON, Signate's included PBX monitoring software,
helps keep the PBX running.
SigMON monitors about 20 different conditions on the PBX
and sends alerts if a
condition needs to be attended to. Monitored conditions
range from hardware
conditions such as available disk space and CPU
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of sf sf
Sent: Friday, March 25, 2005 8:14 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Does [EMAIL PROTECTED] 0.6 really work???
I downloaded it yesterday,first
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom
Sent: Saturday, March 26, 2005 3:17 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Poor pstn line quality
That is exactly what I said in my post, the first line says
that
On Mon, 28 Mar 2005 12:24:00 -0500
steve szmidt [EMAIL PROTECTED] wrote:
On Monday 28 March 2005 12:19, Jon Walsh wrote:
How does one downlaod the upgrade only is there the
ability to do so
from the software or do you need to re-burn an iso or is
the iso an
upgrade version or the whole install
On Tue, 29 Mar 2005 12:55:41 -0600
Jeffrey Sharpe [EMAIL PROTECTED] wrote:
Thank you!
Jeffrey
Please do a little searching of the list next time. I just
answered this same question about 4 days ago!!!
Robert
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Asterisk-Users mailing list
On Tue, 29 Mar 2005 12:30:31 -0800
Noah Silverman [EMAIL PROTECTED] wrote:
hi,
We are using PTSN lines connected through the Digium FXO
modules for our
incomming lines
When a caller calls in, the prompts play back at a
really high volume.
They are a bit distored and fuzzy since they are so
On Wed, 30 Mar 2005 08:29:39 -0500
Matt [EMAIL PROTECTED] wrote:
Hi,
What happened to asterisk @ home 0.7 that the
dialout-default macro no
longer works?
___
EVERYONE
This is NOT the [EMAIL PROTECTED] list group.
Please go to:
On Thu, 31 Mar 2005 10:27:24 -0800
hank smith [EMAIL PROTECTED] wrote:
isn't [EMAIL PROTECTED] included in 1.07? of asterisk? also
I checked the asterisk.org site and saw 1.06 but not the
latest when was it put up on asterisk.org?
Huh??? Last time I checked, [EMAIL PROTECTED] was an install
On Fri, 1 Apr 2005 16:42:54 -0500
Kellner, Peter [EMAIL PROTECTED] wrote:
I've got an asterisk server 1.07 with a Digium TMD400P
(2fxo;2fxs). I
have it configured to answer an incoming line and
transfer to one of the
2 fxs's and it works.
I have noticed that on incoming calls I get
SNIP
If you look at a 'iax2 debug' log you will see things like:
Tx-Frame Retry[000] -- OSeqno: 006 ISeqno: 007 Type: DTMF
Subclass: 6
Timestamp: 15832ms SCall: 2 DCall: 00167
[217.160.244.186:4569]
which seem to indicate the codes are making to my local asterisk
box,
or
Good morning all..
I was following a discussion on this list about the
TDM400P revisions. It is my understanding that the current
revision that one should have is the Rev. H and not the
E/F. I have not yet been able to verify the rev stamped on
the board, but zaptel is reporting that I have
Sorry for the initial no subject line. Was in a hurry when
I typed this and somehow missed putting it in.
Please accept my apologies
On Mon, 11 Apr 2005 10:54:30 -0400
Robert Webb [EMAIL PROTECTED] wrote:
Good morning all..
I was following a discussion on this list about the
TDM400P
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
dean collins
Sent: Monday, April 11, 2005 5:35 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE: [Asterisk-Users] RE: Ebay listing selling
Asterisk @
I was following a discussion on this list about the TDM400P
revisions.
It is my understanding that the current revision that one
should have
is the Rev. H and not the E/F. I have not yet been able to
verify the
rev stamped on the board, but zaptel is reporting that I
have the Rev.
Robert Webb wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of dean
collins
Sent: Monday, April 11, 2005 5:35 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE: [Asterisk-Users] RE: Ebay
On Tue, 12 Apr 2005 15:04:26 -0400
David Brodbeck [EMAIL PROTECTED] wrote:
-Original Message-
From: Jean-Michel Hiver [mailto:[EMAIL PROTECTED]
I don't think the GPL obliges you to give credit to
anybody.
In fact, I think that's a key difference between the GPL
and the BSD
license.
On Tue, 12 Apr 2005 14:05:06 -0500
mr. barker [EMAIL PROTECTED] wrote:
I am using [EMAIL PROTECTED]
When I manually add anything to the
extensions_additional.conf file it gets
rewritten when I add an extension using the web
interface
I am trying to include the monitor function .. I got
that
On Mon, 11 Apr 2005 10:54:30 -0400
Robert Webb [EMAIL PROTECTED] wrote:
Good morning all..
I was following a discussion on this list about the TDM400P
revisions.
It is my understanding that the current revision that one
should have
is the Rev. H and not the E/F. I have not yet
On Thu, 14 Apr 2005 08:14:37 -0600
Rich Adamson [EMAIL PROTECTED] wrote:
I was following a discussion on this list about the
TDM400P
revisions.
It is my understanding that the current revision that
one
should have
is the Rev. H and not the E/F. I have not yet been
able to
verify
On Thu, 14 Apr 2005 07:19:50 -0700
Sean Kennedy [EMAIL PROTECTED] wrote:
G.Marshall wrote:
Hello,
I can not find anything on this, so it may not be
possible.
I would like to dial one number which then rings at least
two extensions
at the same time. Not a hunt group, but ringing at the
On Thu, 14 Apr 2005 10:59:11 -0600
Rich Adamson [EMAIL PROTECTED] wrote:
I was following a discussion on this list about
the
TDM400P
revisions.
It is my understanding that the current revision
that
one
should have
is the Rev. H and not the E/F. I have not yet been
able to
On Thu, 14 Apr 2005 12:45:44 -0400
Ian Pattison [EMAIL PROTECTED] wrote:
My specific issue has to do with ringing on my FXS
ports.
A Northen Telecom Harmony phone (circa 1983) rings
normally but when I connect my newer GE 2.4GHz cordless I
never get more than 1/2 ring (it lights up and works
On Thu, 14 Apr 2005 11:42:34 -0700
Sean Kennedy [EMAIL PROTECTED] wrote:
Hi all
With the recent thread on line presence in asterisk, can
anybody tell me if there is a phone out there that
supports this? Say I have 20 extensions: Is there any
way, hardware based, for me to see the activity on
Test. Please ignore.
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Sorry for all the tests. Please excuse.
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Hi all,
I am new to * and really hate having to ask questions. I have read and
read but still cannot get the concept down for what I want to do.
I currently have an incoming IAX from VoicePulse Connect that is
working. What I am trying to accomplish, is to have that incoming
connection
I have recently started seeing the following message a lot: We have hit
out IOCTL
Can someone please explain what this means and/or how to fix it? It just
recently started appearing and seem to mainly come from when I hang up
an analog extension on a TDM400 card FXS port.
Robert
Try looking in your extensions.conf file. If you are using ports 0 and 2
then you should see somewhere in there something like zap/1 and zap/4
and those should tied to the dial commands.
Hopefully these have been configure in the globals section of the
extensions.conf.
I am kind of a newb at
Have you tried http://www.livevoip.com or http://connect.voicepulse.com ??
I am not sure if they have Las Vegas numbers but you can
look real quick and see. I have one through Voicepulse and it works pretty well.
Have a another through LiveVoip but have not gotten it setup on my end yet. So I
For all the peoples that wanted to test my windows IAX2
phone, I've put it up on a server where it can be downloaded.
I like this phone better than any of the others I have tested so far.
Great work.
The phone can be used mostly with the keyboard :
All comments (good or bad) are welcome
I am having issues that when I call into my * box via a POTS
line and dial an extension that is located on an IAX softphone, if the caller
hangs up before going to voicemail the dialer continues through the plan and
dumps to voicemail. It then records a dialtone.
If I call in through
Subject: Re: [Asterisk-Users] *
not hanging up when call from POTS to IAX phone
Hi
Have you enabled detect busy?
- Original Message -
From: Robert Webb
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Wednesday,
February 02, 2005 5:01 PM
on the POTS line has
hung up.
Guess I will need to do some more digging.
Thanks for the responses
Robert
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert
Webb
Sent: Wednesday, February 02, 2005 11:58 AM
To: Asterisk Users Mailing List - Non
I am having an issue right now where they cannot seem to get their
switch configured correctly. When I call the number I get either a fast
busy or a You're call cannot be completed as dialed message.
I got a response back that when they call from their switch board, they
get a woman's voice
Does the MWI feature work with IAX2? I have read where it
should but cannot get the indicator to work on any of the IAX softphones that I
have tried which have this feature. I even did an IAX debug and did not see
where and indication was sent to the phone when it registered.
IAX2
What we have been discussing with no ringback, is if you have a caller
call in through your DID line and say dials an extension, then after
using the dial command, the caller hears silence and no ringing tone.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Matt,
I thought that DIAX was an IAX based phone not
SIP based. If this is the case then you need to be putting your configs in the
iax.conf not sip.conf file. I have several iax soft phones I have been testing
and have them registering with asterisk. If you want, I can email you the config
I
I have had the same issue when receiving a call from an IAX provider.
Here is what I did to solve it.
[from-proxy]
exten = .,1,Goto(voicemail-direct,s,1)
[voicemail-direct]
exten = s,1,Answer
exten = s,2,VoiceMail2(${EXTEN:1})
exten = 3,3,Hangup
Not sure it does not pattern match like it
I have a regular telephone that when hooked to a standard POTS line, the
incoming callerid signal sets the time on the phone. I do this because
the phone has no internal battery and any little power blip causes the
time to reset to 1/1/98.
Is there a way to either pass the date and time along
I am trying to figure out call parking. It is my understanding that it
is built into *. I have edited the features.conf like I want it but am
unsure where to add the include statement. Right now if I am on a call
from the FXO bridged to the FXS port and I hit the # key, nothing
happens.
I have
On Mon, 14 Feb 2005 14:11:15 +
Bob Goddard [EMAIL PROTECTED] wrote:
On Monday 14 February 2005 13:00, Brett, Gary wrote:
Thanks Mark
I am definitely interested in the budgetone 102 but am a
little concerned
about the 10mbit only Ethernet ports !! From what I have
read, these are
relatively
And middle posting is almost as bad. :-)
But.. To the point...
If you would have read what you were replying to, you
would have noticed they did mention why weren't they
100Mbits connections on the 102 models for daisy
chaining
to a PC.
Robert
SNIP
Yes but failing to trim is even worse. :-)
-A.
Roy Sigurd Karlsbakk [EMAIL PROTECTED] wrote:
I wouldn't recommend the grandstreams, I had very bad
experience using
the grandstream 102, It kep locking up on me. The
buttons are very bad
buttons. The sound quality is just as bad.
grandstream barbie^H^H^H^H^Hudgettone phones really
sucks.
On Thu, 17 Feb 2005 15:04:50 +0100
Stefan Gofferje [EMAIL PROTECTED] wrote:
Hi folks,
I'm registered with sipgate, a German SIP provider.
Configs works fine so far. Trouble is, after a while, it
seems, my registration is dropped by sipgate. How do I
tell * the interval for * registering with a
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Robert Goodyear
Sent: Thursday, February 17, 2005 8:36 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Anyone having trouble with
VoicePulse Connect?
I am running version 1.0.5.22 on my 101 and am not having
any problems.
Robert
On Fri, 18 Feb 2005 12:01:22 -0500
dean collins [EMAIL PROTECTED] wrote:
1.0.5.22 is available for downloading here
http://gs-firmware.gratissip.dk/
I don't know why these are available if Grandstream
don't update
On Fri, 18 Feb 2005 18:11:26 -
Brett, Gary [EMAIL PROTECTED] wrote:
Hello all
I am relatively new to asterisk and am sure this will be
a simple question
to answer. I have a TDM400p card and I am in the process
of creating my dial
plan, however I am a bit stuck on one thing. I have 2
n Fri, 18 Feb 2005 16:06:54 -0500
Bill Hamlin [EMAIL PROTECTED] wrote:
I'm using Dial to place a call to a PBX. But then I
want to wait a few
seconds and dial an extension. Dial doesn't return
until the call is
disconnected though.
I also want the caller to not hear any audio until the
DTMF
PROTECTED]
Behalf Of Robert Webb
Sent: Friday, February 18, 2005 4:18 PM
To: Asterisk Users Mailing List - Non-Commercial
Discussion;
[EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Sending DTMF after a call
is set up
n Fri, 18 Feb 2005 16:06:54 -0500
Bill Hamlin [EMAIL PROTECTED] wrote:
I'm
On Fri, 18 Feb 2005 16:15:24 -0600
Marco Castillo [EMAIL PROTECTED] wrote:
You don't need the zaptel library if you aren't going to
use any digium
cards.
Marco
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
Behalf Of
[EMAIL PROTECTED]
Sent: Thursday, February 17,
Hi all,
With the ability of an easy install using [EMAIL PROTECTED], I have
decided to give it a try. It is my understanding though, that one cannot
add zap fxs ports as extensions using AMP. Is there anyone using
[EMAIL PROTECTED] and have added any extensions as zap fxs channels? Would
be
Any chance that is a bad number??? I do not see anything that would
cause this unless there is a problem with the number you are trying to
dial.
Maybe do am iax debug to get more info??
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of david kwok
Sent:
a patch to amportal project at sourceforge, to
support Zap Extensions...
http://sourceforge.net/tracker/index.php?func=detailaid=1146433group_i
d=121515atid=690574
I'd appreciate some feedback ;)
Greetings
Julian J. M.
On Sun, 20 Feb 2005 22:00:44 -0500, Robert Webb [EMAIL PROTECTED]
wrote
it under their Connect plan.
Thanks,
Robert Webb
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Ok,
With all that has been going on with the list today I may be sticking
my head out of my gopher hole and find a 12 gauge at point blank. But I
am going to take the chance
I have just started using * about a month ago. I have a small unit setup
at home running all my PSTN and VoIP lines.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Hermann Wecke
Sent: Tuesday, February 22, 2005 11:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: FAX
Olaf Klein wrote:
Why not just kill
On Thu, 24 Feb 2005 12:55:31 -0600
Anton Krall [EMAIL PROTECTED] wrote:
Guys...
Ive been having problems with my callerid and I have no
more clues as to
what I could be.. dates and times stamped on voicemail
and info received on
the phones display are off by +6 hours and also the date
for
On Fri, 25 Feb 2005 12:09:16 -0800
Trevor Peirce [EMAIL PROTECTED] wrote:
I don't suppose anyone might know why I hear ringing
transposed over itself when I place a call out via PRI?
SIP to SIP is fine
SIP to IAX is fine
SIP to PRI is always transposed
I mean sometimes you don't notice it much
On Fri, 25 Feb 2005 14:42:09 +
[EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
OK,
After checking into this, I have found the following:
I can set it up so either incoming or outgoing sip calls
on this trunk work but NOT both. The sip show registry
command shows everything as it should be.
The
Me again... I have service with a company that does not allow for a BYOD
plan. They will not give out credential or server info either. Is it
possible to run the FXS port of the ATA to an FXO port in *?
The service I have is throug Broadvox Direct using the Mediatrix 2102. I
have tried this using
')
-- Hungup 'Zap/1-1'
Here is my incoming extensions.conf dialplan:
[globals]
FWDNUMBER=223611 ; your calling number
FWDCIDNAME=Robert Webb; your caller id
FWDPASSWORD=password ; your password
FWDRINGS=IAX2/rwebb ; the phone to ring
FWDVMBOX=2002 ; the VM box for this user
ANALOGPHONE=zap/2
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Guy C.
Guckenberger
Sent: Sunday, February 27, 2005 5:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Outbound call on TDM400P
ok so I put the wait in
This is getting VERY annoying.
Is there anyone in here that has access to the list administration to
delete the user below???
This is an automatically generated Delivery Status Notification.
Delivery to the following recipients failed.
[EMAIL PROTECTED]
This is an automatically
On Thu, 3 Mar 2005 16:20:54 -0500
Amit [EMAIL PROTECTED] wrote:
Hi Everyone,
I am student and I have to study about the source code
of Asterisk. I have
downloaded asterisk and was able to install it on Red
Hat Linux. My study is
to go into the source code of asterisk and see how it
works, how
On Fri, 04 Mar 2005 10:12:05 -0800
Ed Greenberg [EMAIL PROTECTED] wrote:
--On Friday, March 04, 2005 11:58 AM -0600 James Taylor
[EMAIL PROTECTED] wrote:
It would be nice if they told us what the problem with
Asterisk is...
There's probably enought great minds on this list, that
it could be
On Fri, 04 Mar 2005 11:35:55 -0700
Paul Fielding [EMAIL PROTECTED] wrote:
Ok, time for me to ask my own newbie question. :)
I've done some digging on ringback, and if I'm
understanding it correctly, it's the ring tone that the
caller hears when dialing another person.
What exactly is it
on the the
calling end.
Sorry for the incorrect info the first time, it had just
been quite a while since I had played with the Live
account.
Robert
- Original Message - From: Robert Webb
[EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
the time the line is picked up and the time
the * box starts dialing??
Something like:
exten = s,1,Dial(ZAP/g1/ww206391)
I have seen a couple of people that have had the same
issue as you and this fixed it for them.
Robert Webb
-Original Message-
Subject: [Asterisk-Users] Problems dialing
-Original Message-
From: Robert Webb [mailto:[EMAIL PROTECTED]
Sent: Saturday, March 05, 2005 5:24 PM
To: 'Asterisk Users Mailing List - Non-Commercial
Discussion'; 'leandro_tenorio'
Subject: RE: [Asterisk-Users] IAX2 (Variables)
-Original Message-
From: [EMAIL
On Sun, 6 Mar 2005 20:22:48 -0500
Steven Frazier [EMAIL PROTECTED] wrote:
I have about 10 DIDs, I had an issue that lasted a day
or so that was Level
3's issue, it took about 12 seconds before the calls
would come in. That
was resolved and I haven't had any issues at all. I
appreciate the
On Tue, 8 Mar 2005 14:17:23 -0300
Alejandro G [EMAIL PROTECTED] wrote:
I have 2 asterisk box in different locations. When I
received a call in one
location and want to transfer it to an extension in the
other location the
external call is hanged up when the person who is
transfering the call
On Tue, 08 Mar 2005 14:11:26 -0500
[EMAIL PROTECTED] wrote:
Christopher Jacob wrote:
Hey All,
I have a user whose Cisco 7960 died the other day. When
you disconnect the
Ethernet cable the phone boots (as far as it can w/o
network connectivity)
but as soon as you plug in the CAT5 it goes dead.
On Wed, 09 Mar 2005 13:06:24 -0800
Sean Kennedy [EMAIL PROTECTED] wrote:
Hi all, I'm running Asterisk 1.0.0. I am a customer (
and supporter ) of voicepulse. For me, it works
perfectly, but one of my customers noticed a small
problem: During a conversation, when the otherside isn't
On Thu, 10 Mar 2005 15:53:05 +0200
Turgut Abacioglu [EMAIL PROTECTED] wrote:
Hi
I am fairly new to Asterisk. I will have few questions
on one Asterisk
system:
Description: I looked at Asterisk Mall says that The
TrueLine SMB PBX is
perfect for the small office providing service for up to
On Thu, 10 Mar 2005 11:03:44 -0600
Henry Devito [EMAIL PROTECTED] wrote:
Hi does anyone know how to get festival to run on the
latest [EMAIL PROTECTED]
0.6. Is there a mailing list for [EMAIL PROTECTED]
THere is a forum section on SourceForge where you
downloaded the ISO from. Try there...
On Thu, 10 Mar 2005 21:10:54 +0100
Manuel Antonio Casal Hernández [EMAIL PROTECTED]
wrote:
Hi all,
We need help with our SuSe9.2 asterisk box
We have one QuadBRI and one TDM40B in an ASUS pundit R-2
barebone.
We have downloaded the bristuff (0.2.0-RC7j) and
installed it without problems.
once
On Fri, 11 Mar 2005 11:47:53 -0700
Wiley Siler [EMAIL PROTECTED] wrote:
Hello All,
I saw some coverage of this in the list archive but no
one seems to have
posted a resolution.
I am using [EMAIL PROTECTED] 0.06 and when I get a call from
LiveVoip over
IAX I dump it into my IVR.
From there the
On Fri, 11 Mar 2005 12:21:21 -0700
Wiley Siler [EMAIL PROTECTED] wrote:
Hello all,
For some reason I am not showing registration in SIP.
Can anyone give me an idea what can cause this?
asterisk1*CLI sip show registry
HostUsername Refresh
State
How about showing
On Fri, 11 Mar 2005 12:48:49 -0700
Wiley Siler [EMAIL PROTECTED] wrote:
All my SIP phones are still working and all my dialing
is still working,
so I did not think it relevent.
Sip reload...
asterisk1*CLI sip reload
Reloading SIP
== Parsing '/etc/asterisk/sip.conf': Found
== Parsing
On Fri, 11 Mar 2005 21:48:07 +0100 (CET)
Peter Svensson [EMAIL PROTECTED] wrote:
On Fri, 11 Mar 2005, Wiley Siler wrote:
I saw some coverage of this in the list archive but no
one seems to have
posted a resolution.
I am using [EMAIL PROTECTED] 0.06 and when I get a call from
LiveVoip over
IAX
Is anyone out there using any of the newer linksys phones since Cisco
took over? I am more specifically looking at the spa-941 942's. Just
curious about call quality, programability, and functionality with asterisk.
I have read through the literature, but would like some real world feedback.
I sent the below message out last Friday when the list
seemed to be having issues. Never got any responses and
not sure if it just no one knows or if it did not get
through.
Please don't flog me too bad for reposting... :-)
Hi
On Wed, 15 Feb 2006 08:59:22 -0800 (PST)
housi mueller [EMAIL PROTECTED] wrote:
Hi there,
I would like to connect an Aasterisk Server with a
Panasonic PBX (has E1extension).
I only need 4 Lines. So I thought I could use an
Dignum TDM04 Card with 4 FXO or a Dignum TE110P E1/T1
card
Sorry, this is off topic to asterisk itself, but is about
the list server.
I had a power failure lastnight at home, where my email
server resides, and my network was down for about 20
minutes, that was after 45 minutes of uptime on UPS. Since
power was restored, around 9:45 PM EST on 2/16,
On Tue, 7 Mar 2006 09:12:25 -0700
Douglas Garstang [EMAIL PROTECTED] wrote:
I have a configuration where RTP traffic is going out
interface pub0, and coming back into through pub1.
I have bindaddr=0.0.0.0 in sip.conf, and a netstat -an
shows:
udp0788 0.0.0.0:5060
On Wed, 20 Jul 2005 18:00:24 +0200
Robert Rozman [EMAIL PROTECTED] wrote:
Hi,
I spot weird behaviour of latest Firefly 3rd party on my
laptop. Sometimes it comes to state that it won't start
(hangs on Initializing ) and it again works after
system restart... Didn't yet figured out how to
SNIP
.
And please note that in general members of the list
dislike List Police even more than they do off-topic
posters.
B.
Cool...
I will be sure to ask any question I have now and
expect not to get Policed by anyone on this list. Sounds
like this is the list for the support of
On Mon, 25 Jul 2005 15:44:07 +0200
Alexis F. [EMAIL PROTECTED] wrote:
Hi,
I would like to use a digum card to call an external
number through my PSTN. I think that I have a problem in
the configuration. Asterisk returns me app_dial.c:764
dial_exec: Unable to create channel of type 'Zap'
On Tue, 26 Jul 2005 10:24:20 -0400
Andrew Kohlsmith [EMAIL PROTECTED]
wrote:
On Tuesday 26 July 2005 09:43, chouck wrote:
I assure you I have read the asterisk handbook many
times. The
immediate=yes is for picking up a phone on an fxs and
having it immediately
dial an extension. I am
On Wed, 27 Jul 2005 18:07:23 +0200
Walid Azab [EMAIL PROTECTED] wrote:
Hi..
I am trying to do something but it is giving me some
hard time here. I have
an IAX2 trunk to FWD which is registered and working
just fine. I have =
011|. as my dial pattern to allow that. But if I want to
dial a
SNIP
Ok, I figured it out, * was not using the config under
the [router]
context in the config file. Once I enabled g729 in
[general] it worked.
So the question is why does * ignore this config for the
192.168.77.254
endpoint?
in sip.conf:
[router]
type=friend
context=default
On Wed, 24 Aug 2005 14:47:25 -0400
Araba, Michael [EMAIL PROTECTED] wrote:
Thanks John, You are my savior. This is such a great
relief. Apparently
realtime will not use either '127.0.0.1' or 'localhost'
to connect to the
database. I had to use the actual IP address attached to
the NIC before
On Wed, 24 Aug 2005 15:25:15 -0400
John Novack [EMAIL PROTECTED] wrote:
In my case, mysql is set to any host
So, yes, it does seem to be an Asterisk issue
And my buddy is pretty savvy with mysql, Linux and
databases on Unix/Linux, having worked for a large IT
company for some 20 years.
Senerio
multiple * boxes connecting to a central * box with T1 card via IAX2.
1box 1 abd 2 work fine all the time
box 3 - after approx 10-15 minutes with no calls - central box with T1
card
fails to deliver incoming calls to box 3.
Connectivity is good, * exten-2-exten good
Ok, I am
See inline responses...
On Mon, 18 Apr 2005 10:43:30 -0400
Ian Pattison [EMAIL PROTECTED] wrote:
I don't know how everyone else is doing but my woes are
continuing.
Hardware:
Digium TDM400P (REV G according to the silk screening on
the board) 2xFX0, 2xFXS purchased in August/September
2004
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