FC2, Asterisk 1.0.0, Zaptel 1.0.0
TDM400P Port 1 FXS Port 4 FXO
Standard analogue handset plugged in with pstn line.
Problem:
I have 2 analog phones that I use, when plugged directly into pstn line
both phones work perfectly, dialing no issues. When I plug the handsets
into the TDM400P, one
I get this warning on zap channels:
Sep 26 07:17:21 WARNING[1102232496]: chan_zap.c:5903 do_monitor: Whoa
I'm owned but found (23)...
Sep 26 07:45:53 WARNING[1102232496]: chan_zap.c:5903 do_monitor: Whoa
I'm owned but found (23)...
Sep 26 07:59:59 WARNING[1102232496]: chan_zap.c:5903
Greetings,
I'm running latest * from CVS on FreeBSD 4.10 box. We've just bought
several WellGate 3502A FXSes to play with till welltech guys fix the
3504a's registration bug.
So far everything is working as expected, except the fact only ulaw and
alaw codecs work with *. If I add allow=gsm or
Hello!
Is it possible for somebody to email me the patch for the UK callerid (for
the X100P cards). I know that the TDM100P patches are included but .. I
still use the X100Ps
Many thanks
Vassilis
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--On Sunday, September 26, 2004 02:29 +0200 Philipp von Klitzing
[EMAIL PROTECTED] wrote:
FAQ: Turn of silence suppression in X-Lite by setting Transmit silence
to YES (in AUDIO settings).
AHA! That's it! Works now. Not sure what silence suppression has to do
with all but it works now.
Tnx
Hello Everyone,
Ive been struggling with this issue for about two
days; hopefully its something trivial that has been overlooked.
Basically, I have a Cisco 7940 handset running the SIP 7.1
firmware, which can place outbound calls to any destination, however
It can not receive calls
On 26/09/2004 15:41 Vahan Yerkanian said the following:
I'm running latest * from CVS on FreeBSD 4.10 box. We've just bought
several WellGate 3502A FXSes to play with till welltech guys fix the
3504a's registration bug.
welltech still hasn't responded to my complaint to them regarding their SIP
Alfred Nurnberger schrieb:
Try sipgate.de.
They have free DIDs in many german citys and their rate into Germany is
very affordable (aprx. $0.02 / min.)
...
Hi,
I'm sure, they won't anymore. For it were Sipgate and Nikotel,
who got those letters from RegTP, forbidding to give DIDs to
non-local
Hello Dinesh,
my welltech does support g729 g723.1 and g711u and a.
It works like a charm here.
BUT; the registration problem still exists (even after using your rc)
--
Best regards,
Dannymailto:[EMAIL PROTECTED]
belGOnet.com a Euro-pictures division -
On Sat, 25 Sep 2004, SeshKanuri wrote:
Dear Valued OnDO users,
Huh?? I'm not an OnDO user!?
This is the ASTERISK list.
Steve
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have posted a while ago on issues of receiving faxes by an Asterisk host
using an x100p fxo interface attached to BT pstn
the asterisk installation is the cvs download as of 23/09/04
is anyone able to confirm that the rxfax / txfax application that seems to
be 'bundled' in thecvs download is the
Hi,
Have some welltech devices myself, but after about one year of trials I
decided to throw them out ...
Even when you succeed in making it register it freezes after a random
amount of time, no matter what SW u use on it.
Regards,
Kiss Karoly
On Sun, 26 Sep 2004, Danny Zak wrote:
Date: Sun,
Hi All,
I have successful compiled asterisk from CVS. But when I start it up, it
show an messages like this:
[chan_sip.so]Sep 26 13:29:35 WARNING[-1084308832]: loader.c:248
ast_load_resource: /usr/lib/asterisk/modules/chan_sip.so: undefined symbol:
__use_ast_pthread_create_instead__
Sep 26
Graham Turner a écrit :
have posted a while ago on issues of receiving faxes by an Asterisk host
using an x100p fxo interface attached to BT pstn
the asterisk installation is the cvs download as of 23/09/04
is anyone able to confirm that the rxfax / txfax application that seems to
be 'bundled' in
Hi all.
I've been reading through Wi-Ki and at the extensions.conf file
description (http://www.voip-info.org/wiki-Asterisk+config+extensions.conf)
The author says this:
One day, someone is going to write a proper scripting language for
Asterisk that can understand a simpler, easier (and more
I've setup the voicemail that auths against the mysql db. Now,
everything works ok, except voicemail() calls fail with
Sep 26 18:09:34 WARNING[157070336]: app_voicemail.c:1517
leave_voicemail: No entry in voicemail config file for ''
all my users are in 'sip' voicemail context, but adding
On Sat, 25 Sep 2004, Florin Andrei wrote:
On Fri, 2004-09-24 at 05:47, Greg Boehnlein wrote:
Anyone else having the problems that Gary is reporting?
Um, well, not really. I'm rebuilding your package on Fedora 2 (kernel
2.6) and i had to add a linux 26 at the end of the make line,
Hello Kiss,
well we didn't have this kind of errors .. YET :)
--
Best regards,
Dannymailto:[EMAIL PROTECTED]
belGOnet.com a Euro-pictures division - internet solutions
place princesse elisabeth 9/11 - 1030 Brussels - Belgium
Tel : +32-(0)2-215.67.65
Hi,
I am trying to configure a basic Asterisk setup with Asterisk running
on Linux and a GS 102 phone and an analogue phone connected (through
IAXY adaptor) to Asterisk.
I am stuck at the step of provisioning the IAXy adaptor. I will
quickly give an overview of my setup.
I installed Redhat 9, got
I've installed spandsp-0.0.1k on a RHv9 box with CVS-HEAD-09/19/04 and
compiled the libraries just fine. Having a problem with patching the
asterisk/apps Makefile however. The patch attempt results in:
[EMAIL PROTECTED] apps]# patch Makefile.patch
patching file Makefile
Hunk #1 FAILED at 35.
Transmit silence = no tells the client not to send anything until it has
some audio to send. The other end is dropping the connection because it
thinks the client has lost it's connection.
Lyle
- Original Message -
From: Michael Loftis [EMAIL PROTECTED]
To: Asterisk Users Mailing List
On Sun, 26 Sep 2004 15:03:58 +0200, Rodolfo Grave [EMAIL PROTECTED] wrote:
but I'm sure with the help of the community and if there
are enough people needing/wating this, we can make a good language
specification. Personally I can and will make the compiler from the
higher level to current
I set up the pilot number to voicemail to be 777. When a user calls 777 the voicemail answers
and asks for mailbox, then password. Is
there a way for the Voicemail to read what extension they are calling from and
just ask for the password? I have a
person complaining because they have to
Hello,
On Sun, 26 Sep 2004 00:49:35 -0400, Robert Jackson
[EMAIL PROTECTED] wrote:
[snip]
4. If a caller empties a handled queue (active agents) with
no callers, the caller will still hear messages (you are
first in queue, etc.). This should not occur. Someone
posted a 2-line patch
exten = 777,1,VoicemailMain([EMAIL PROTECTED])
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Sun, 26 Sep 2004, Henry Devito wrote:
I set up the pilot number to voicemail to be 777. When a user calls 777 the
voicemail answers and asks for mailbox, then
That is because it is a required argument.
http://voip-info.org/wiki-Asterisk+cmd+VoiceMail
And you can see the difference from voicemailmain():
http://voip-info.org/wiki-Asterisk+cmd+VoicemailMain
quote who=Vahan Yerkanian
all my users are in 'sip' voicemail context, but adding context to it:
On Sun, 2004-09-26 at 17:18, Benjamin on Asterisk Mailing Lists wrote:
[snip]
So how about a discussion to give Asterisk an XML based configuration as a base.
For backwards compatibility, the existing config language could then
sit on top of the XML as one alternative amongst others.
I think
I have the same problem and
suppose that by some works on a new application similar to
parkandannounce app. it should be done.
- shabanip
- Original Message -
From: Alex Forrow [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, September 17, 2004 5:43 PM
Subject: [Asterisk-Users]
Hello All,
I have been looking for a commercial version of an IAX2 Softphone for
Windows but the ones I have came across (i.e. Iaxcomm, Iaxphone, Diax) do
not seem to have an updated version since April 2004 in some cases.
We looked at Firefly but we sent emails to Virbiage/Freshtel with
Depending on your needs I don't know if you will find 1 that used IAX2
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Sorry about that cut off . Like I was saying I'm not sure if you will
find once advanced enough using IAX2 currently. Firefly was the most
evolved when I too was looking but their oem terms weren't exactly
what I wanted to spend given the fact that I probably would be going
hardphones eventually.
NI-2 gives you the best set of available features (ie.e CNAM callerid)
and is my preferred choice for our PRI setups.
Alfred.
Christopher Jacob wrote:
I am having my colo set up 2 PRI's for my new asterisk implementation. They
asked the following...
##SNIP##
What type (NI2, NTI, 4ESS,
XML isn't the magic answer to all questions. The configuration
database for Asterisk is relatively flat while XML supports
hierarchical data much better. Asterisk's current config file are
better suited to the task than XML inherently could be.
If the original poster is still interested writing
Greg let me know where the new RPM is so I can update my system as it's
still down...
Thankfully it wasn't my production machine..
Thanks again..
John B.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Greg Boehnlein
Sent: Sunday, September 26, 2004
Henry,
exten = 451,1,Answer
exten = 451,2,Wait(1)
exten = 451,3,VoiceMailMain([EMAIL PROTECTED])
exten = 451,4,Wait(1)
exten = 451,5,Hangup
is what you're looking for.
VoiceMailMain(${CALLERIDNUM}) might be enough if you're not storing your
voicemailboxes in mysql
regards,
Vahan
Henry Devito
Hi William,
Most my users will be Windows based.
My problem is that I am not a developer so I am unable to modify an open
source client. Iaxcomm and Iaxphone only seem to support GSM but it would
be nice to have support for iLBC as well.
All I really need is to have the color scheme of the
Yes We have one... Its going to be released in October... We are in middle
of Moving so once we move and get things settled we will be going online
Unlike the diax and iaxcom and iaxphone ours is using DirectSound and has
many more abilities.
We are releasing software for our services but we
At 19:19 26.09.2004, you wrote:
Yes We have one... Its going to be released in October... We are in middle
of Moving so once we move and get things settled we will be going online
Unlike the diax and iaxcom and iaxphone ours is using DirectSound and has
many more abilities.
We are releasing
Only the Asterisk IAX2 Part
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Atuc
Sent: Sunday, September 26, 2004 1:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Looking for a commercial version of
Hi guys,
I know this isn't strictly about Asterisk, but it is related...
I am looking to buy a few IP phones, but I don't have a huge budged (hence
why I love Asterisk, its amazing and free !), so I was wondering if anyone
knew where I could get some cheap IP Phones ?
Ideally they should be no
On Sun, 26 Sep 2004 19:04:38 +0100, Paul Tyreman [EMAIL PROTECTED] wrote:
Hi guys,
I know this isn't strictly about Asterisk, but it is related...
I am looking to buy a few IP phones, but I don't have a huge budged (hence
why I love Asterisk, its amazing and free !), so I was wondering if
I have a strange question. I am new to * and have it up and running for our
office phones. We run a small dialer to call clients and remind them of
ordership dates etc. I would like to have * take the calls from it and send
them through a voip connection.
We have a digium quad port t1 card.
I was somewhat concerned reading Mark's posting earlier today.
Obviously, things are very bad in the US at the moment. Their
Government even deported Cat Stevens the other day (check
http://news.bbc.co.uk/1/hi/england/london/3686992.stm ).
Clearly, given the fact that Digium contributes so
On Sun, 26 Sep 2004, Daniel Pocock wrote:
Therefore, it seems to be in the best interests of Asterisk's `security'
to have the mailing lists hosted by someone other than Digium and maybe
in a country that doesn't prohibit freedom of expression.
Amusing bit of stirring there.
But, PLEASE,
Steve, ? Daniel thanks for reply posts
the location i download from is as per technote on * installation;
export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot
prior to the last download i had to manually install the rxfax / txfax
applications from opencall.org
after latest download rxFAX /
Not the cheapest ($75-80) but they look interesting.
http://ipphone.eezeephone.com/
Jonathan
At 03:10 PM 9/26/2004 -0300, you wrote:
On Sun, 26 Sep 2004 19:04:38 +0100, Paul Tyreman [EMAIL PROTECTED] wrote:
Hi guys,
I know this isn't strictly about Asterisk, but it is related...
I am looking
Another solution would be to keep the discussions on topic and open up a
separate mailing list for people interested in open discussions.
Jonathan
At 07:17 PM 9/26/2004 +0100, you wrote:
I was somewhat concerned reading Mark's posting earlier today.
Obviously, things are very bad in the US at
Dear;
regarding the details that i found on the site about spandsp; is it
correct to assume the following ?
+ spandsp will only work with a card that is located in the * box?
therefore
will my welltech or any other voip fxo adapter support FAX ( i know it
does t.38)
--
My current situation
I understand that the iLBC codec supports a variety of operative modes incluing
a wideband mode. This could be useful in improving call quality over other
codecs, say GSM, but retaining the iLBC strength in packet loss recovery. How do
I control the codec to establish the compression settings on
I agree with the mailing list part, but things arent very bad in the USA.
Yusuf Islam was denied entry, not deported. I am sure there is more to this
story than is being told or possibly ever will be told. He is not a US
citizen and can be denied entry for any reason or suspicion. Please do
On Sat, 25 Sep 2004, David Troy wrote:
[deleted[
I had a need for a much simpler proxy than his op_server.pl; to meet my
need I re-worked and simplified his code. See below for this simplified
proxy:
http://www.popvox.com/simpleproxy.pl
Hehehehe.. I mentioned this in the Developer's
Hey group!
Could someone could help me configure a DIal plan in order that when i dial 9 at the beginning i receive DIAL TONE?
Do you Yahoo!?
Take Yahoo! Mail with you! Get it on your mobile phone.___
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On 27/09/2004 00:50 Jay Milk said the following:
Eliminating the need to specify (and keep track of) priorities would
make changes to extensions.conf much easier to implement.
or perhaps allow non-consecutive priorities.
--
Regards, /\_/\ All dogs go to heaven.
[EMAIL
Graham Turner a écrit :
Steve, ? Daniel thanks for reply posts
the location i download from is as per technote on * installation;
export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot
This only for *, nothing todo with spandsp
prior to the last download i had to manually install the rxfax /
Dinesh Nair wrote:
On 27/09/2004 00:50 Jay Milk said the following:
Eliminating the need to specify (and keep track of) priorities would
make changes to extensions.conf much easier to implement.
or perhaps allow non-consecutive priorities.
After this topic was discussed a bit at the developer's
Danny Zak a écrit :
Dear;
regarding the details that i found on the site about spandsp; is it
correct to assume the following ?
+ spandsp will only work with a card that is located in the * box?
should also work with ztdummy if you don't have a card.
--
Daniel
I pray for an end to the priorities as well. The +101 could be easily solved
by a default label, or an option to the dial
for example:
exten = _7XX,1,Dial(yada,10)
exten = _7XX,2,Voicemail(unavail)
exten = _7XX,3,Hangup
exten = _7XX,102,Voicemail(Busy)
could be:
exten = Dial:_7XX,Dial(yada,10)
Why VMxyz, does every line end up at the VM when it's busy or
unavailable or unregistered btw we could then also add a rule for
the case the user agent has registered with * (bristuff addon n+201)
Best Regards,
Marc
Asterisk wrote:
I pray for an end to the priorities as well. The +101 could
I have hdlc networking and voice channels between two * boxes using a
T-1 P2P circuit. I have Digium T-1 cards on both systems.
I've loaded zaptel/libpri/asterisk 1.0 on one of the boxes. When I
start zaptel and run ztcfg I get Zaptel networking not supported by
this build.
Has anyone else
Brian Capouch wrote:
Of course, the problem of the hard-coded priority + 101 situation is
problematical. I say we think through what the perfect world would look
like in this respect and then see how hard it would be to implement. . .
XML will probably able to store much, probably more, of the
I like my first suggestion the best... And that is of course fully
subjective:
[context]
exten = _1xxx
:DigitTimeOut(10)
:ResponseTimeOut(20)
BackHere:Answer
:Read(callto,pls-entr-num-uwish2-call,10)
:Read(callfrom,enter-phone-number10,10)
:SetCIDNum(${callfrom})
Hello Asterisk,
is it possible to make an extensions that write a call file
(like a call back to the callerid) in
the outgoing directory WITHOUT using a perl AGI ?
--
Best regards,
Danny mailto:[EMAIL PROTECTED]
belGOnet.com a Euro-pictures
Things aren't bad in the US at the moment. In fact, I think they're
pretty good, because people actually seem concerned about PATENT and
COPYRIGHT LAWs which your initial post attempted to circumvent. There
was no issue with freedom of expression; there was an issue with
legality of posts based
Hi people,
My asterisk wont register with any sip providers, I have tried three
different but they all end up with:
Sep 26 17:36:36 NOTICE[114696]: chan_sip.c:4035 sip_reg_timeout:
Registration for '[EMAIL PROTECTED]' timed out, trying again
There is no firewall and my server has a public IP.
-Original Message-
From: Alex Forrow [mailto:[EMAIL PROTECTED]
Sent: Friday, September 17, 2004 9:13 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Transferring Calls
Both methods seem similar; you enter the number and it
transfers. The
problems arise when the phone
This is in the notes in the default extensions.conf - ignorepat
= 9
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Gonzalo Gasca Meza
Sent: Sunday, September 26, 2004
2:28 PM
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users]
Has anyone made spandsp to work with a digium tdm fxo card?
I finally got the rxfax and txfax modules to compile, the spandsp lib
installed (and in the libpath), and now receive:
-- Starting simple switch on 'Zap/1-1'
-- Executing RxFAX(Zap/1-1, /var/fax.tif) in new stack
-- Hungup
-Original Message-
From: Danny Zak [mailto:[EMAIL PROTECTED]
Sent: Sunday, September 26, 2004 5:29 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Dialplan question
Hello Asterisk,
is it possible to make an extensions that
I forgot to add a link to the system command:
http://www.voip-info.org/wiki-Asterisk+cmd+System
-Original Message-
From: Robert Jackson
Sent: Sunday, September 26, 2004 5:57 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Dialplan
- Original Message -
From: Jay Milk [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
[EMAIL PROTECTED]
Sent: Sunday, September 26, 2004 4:35 PM
Subject: RE: [Asterisk-Users] Digium and mailing lists
You're free to express your
discontent about G.729
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