On 8/9/07, Gleim, Jason [EMAIL PROTECTED] wrote:
[snip]
I thought that might be an issue too... and it was originally. When we
started out, I had the Sangoma card generating the timing for the span
but we could never get the d-channel to come up. Turns out that since we
were connected to the
On 8/9/07, Oscar Patricio [EMAIL PROTECTED] wrote:
Hello!
I have following scenario:
PBX - Asterisk - ISDN E1 line
The asterisk box relays calls from the E1 to the PBX and vice versa.
Additionally some outgoing calls of the PBX are being sent over VoIP
providers instead of using the E1
Hi,
It is possible to jump into a Macro (or some similar dialplan jump)
when a transfer causes a call to be re-bridged? I do not believe that
GOTO_ON_BLINDXFER will do the job, because we use SIP phones, and use
the handset's own transfer or blind-transfer facilities.
What I want to achieve is
Hi,
(cc. asterisk-users, hope that is not a big Faux Pas)
I've had trouble with the qozap driver for a LONG time now, where it
will not recognise and ignore a missing ISDN2 line on a quad card if
one of the 4 ports is unplugged or somehow faulty.
The symptom is that is correctly recognises the
On 8/15/07, OCOSA ListAcct [EMAIL PROTECTED] wrote:
Did not work either...Thank you!
Otis
Michiel van Baak wrote:
On 15:02, Sun 12 Aug 07, OCOSA ListAcct wrote:
exten=5,2,Dial(SIP/supportSIP/support2,2,tr)
Make this line read:
exten=5,2,Dial(SIP/supportSIP/support2,,tr)
On 8/16/07, Rory Campbell-Lange [EMAIL PROTECTED] wrote:
We are trying to configure a Sangoma A101 card to allow both incoming
and outgoing calls on a UK (BT) ISDN30e line with only 24 channels
enabled.
At present incoming calls work fine. We can't call out -- we get a
BUSY/CONGESTED error.
On 9/24/07, Craig Guy [EMAIL PROTECTED] wrote:
The only nasty thing I've found is that whenever the handsets resync they
reboot even if no settings have changed. When this occurs anything
connected to the phones second Ethernet port will drop connection for a few
seconds.
The phones can
On 9/24/07, Erik Anderson [EMAIL PROTECTED] wrote:
On 9/24/07, Steve Davies [EMAIL PROTECTED] wrote:
The phones can send a parameter to the provisioning server to indicate
that they want an Update if they do this, and you send no network or
other major config parameters, the phone does
I can look at adding a server-filter parameter to astmanproxy.users
(no promises on timescale though!) as I wrote the per-user filtering
in the first place.
My problem with astmanproxy at the moment is that I don't get any
responses from the maintainer (Dave at popvox?). I have a couple of
Most of thread snipped.
On 10/24/07, marcotasto [EMAIL PROTECTED] wrote:
Some days ago I've sent to David Rowe a little patch that preserves the echo
cancel
status between calls.
Surely this is only appropriate where you have a local analogue device
that is unchanging - If you retained the
Utterly untested, but here goes with the server-filtering parameter...
The attached patch should apply to version the 1.21 tarball cleanly,
and includes all my other changes which haven't made it into the main
astmanproxy code.
Please do feed-back on whether this works (it compiles :-) ).
On 10/29/07, Christian Stredicke [EMAIL PROTECTED] wrote:
What you can still to is setting the port on the phone to port 5060 - just as
a little dirty workaround until there is a better solution available.
CS
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED] [mailto:[EMAIL
Try the following in zapata.conf
internationalprefix=900
nationalprefix=90
Which should do this for you unless your provider is not supplying the
correct indicator.
Regards,
Steve
On 1/31/06, Phil Blundell [EMAIL PROTECTED] wrote:
When a call arrives on our PRI from a UK domestic number, the
On 2/13/06, Bob McDowell [EMAIL PROTECTED] wrote:
Does anyone on the list have a recommendation for a TAPI interface to
Asterisk? I have tried all of the ones that Google produced, but have
still not yet found a solution that I can move into production. My
favorite to date is AstTapi, but
On 2/13/06, Bob McDowell [EMAIL PROTECTED] wrote:
The issue appears to be something on the XP desktop side. I can end-task
and restore TAPI functionality about 75% of the time. Otherwise, a
reboot always clears it up.
I'm unfamiliar with astmanproxy. I'll look it up.
I removed siptapi
at this stage?
Many thanks in anticipation.
Regards,
Steve
On 12/20/05, Steve Davies [EMAIL PROTECTED] wrote:
http://www.google.com/search?q=cache:3AXi4YvnS80J:www.sangoma.com/company/news_releases/octasic.htmhl=en
it seems that there will soon be an A102d, A104d and A108d available
on the market
If the two phones attempt to refer to each other using their external
(NAT) IP addresses rather that their internal addresses, then it will
all go horribly wrong. You do not provide enough information about
asterisk IP addresses or firewalls for a possible solution, but
assuming you are using SIP
On 20/03/2008, Johansson Olle E [EMAIL PROTECTED] wrote:
20 mar 2008 kl. 09.32 skrev Stefan Schmidt:
hello,
i am trying to set up a asterisk server (version 1.2.26 by now) with
realtime configuration but the user shouldnt register directly to the
server, instead i have set up a
On 20/03/2008, Loic Didelot [EMAIL PROTECTED] wrote:
Hello,
I am having some troubles with Snom phones and maybe someone can help
me.
Let me say this: BLF and pickup works great with Polycomes and
Grandstream etc... So I think my problem might not be Asterisk related
but I am not 100%
Alternatively...
On 24/03/2008, Tilghman Lesher [EMAIL PROTECTED] wrote:
On Monday 24 March 2008 04:02, mark morreny wrote:
Dear friends,
I am having problem with running a sample php and I can't figure out why.
I can run the sample.php using CLI but when I run it inside the dialplan
On 25/03/2008, Gordon Henderson [EMAIL PROTECTED] wrote:
On Tue, 25 Mar 2008, Vieri wrote:
How can I force soft hangup (if that makes sense)?
show channels reveals a stale sip channel. It's of
an analog phone behind a Grandstream ATA which was
communicating with another SIP
On Tue, 25 Mar 2008 14:58 +0200, Yehavi Bourvine +972-8-9489444
[EMAIL PROTECTED] wrote:
[snip]
LOL. Very creative :) Thank you for the suggestion. I can work with that!
Steve
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
On 25/03/2008, Tilghman Lesher [EMAIL PROTECTED] wrote:
On Tuesday 25 March 2008 10:17:54 Vieri wrote:
--- Steve Davies [EMAIL PROTECTED] wrote:
Using rtptimeout and rtpholdtimeout settings in
sip.conf
I set
rtptimeout=10
rtpholdtimeout=30
(just for testing; I know
Hi,
Could you explain for the benefit of the list what you have changed in
the snom image that will benefit this ticket? I am already receiving
your current beta images, through our distributor, up-to about
2008-13-19, and am not aware of any changes that affect BLF behaviour
or short-dials...
On 27/03/2008, David Nedved [EMAIL PROTECTED] wrote:
So now it seems 1.4.18 is doing the same as 1.2.27 -- working for the
most part but completely ignoring DTMF on incoming SIP calls.
Perhaps you now need to delve deeper. Capture a UDP trace between your
VoIP provider and Asterisk, and
Hi,
The twist? We actually have far-end hangup detection working fine, and
that seems to be where the problem lies for most people. Our problem
seems to be with requesting a hangup from our end reliably.
If we originate the call, we can hang it up. This suggests to me that
the Sangoma A200D is
On 31/03/2008, Mike Dent [EMAIL PROTECTED] wrote:
On 31/03/2008, Steve Davies [EMAIL PROTECTED] wrote:
The twist? We actually have far-end hangup detection working fine, and
that seems to be where the problem lies for most people. Our problem
seems to be with requesting a hangup from our
On 31/03/2008, David Boyd [EMAIL PROTECTED] wrote:
You should ask for ground start signaling. This will resolve your
issues.
Could you point me at some reference material for how this differs
from KS, and what compatibility issues this might cause with other
equipment? Has anyone tried this
You can get much better results (close to 56k reliable connections
sometimes) by using a Xorcom FXO Channelbank - You need recent enough
drivers so that the Xorcom internal clock can be synced to Zaptel;
This removes/reduces jitter and frame slippage, and allows a modem to
operate much more
Anyone interested in this feature? I have a version 0.1 patch, which
is currently against 1.2.25-bristuffed, but which should port
trivially to almost any version. I am away until Tuesday 8th April,
but if there is enough interest, I will open a new-feature ticket
and upload the patch to the
On 03/04/2008, Steve Davies [EMAIL PROTECTED] wrote:
Anyone interested in this feature? I have a version 0.1 patch, which
is currently against 1.2.25-bristuffed, but which should port
trivially to almost any version. I am away until Tuesday 8th April,
but if there is enough interest, I
I believe that what you described should just work with the caveat
that dtmf=inband is rarely the right thing to do over SIP, and is
prone to all sorts of DTMF detection and debounce issues.
I assume you've tried calling a POTS endpoint and listening to see if
you get DTMF passed through?
1) You
On 08/04/2008, Steve Davies [EMAIL PROTECTED] wrote:
http://bugs.digium.com/view.php?id=12382
Patch has been attached. Currently only for asterisk 1.2.25, but if
no-one else provides a 1.4.x patch soon, I will probably need to do
that for myself anyway.
As a courtesy I have uploaded
On 10/04/2008, Stefan Reuter [EMAIL PROTECTED] wrote:
Adrian A wrote:
Is there any way of removing this line from showing on the console? I
have a script that logs in every few seconds to manager (...)
Maybe a better solution is to rethink your architecture. The Manager API
is well
On 14/04/2008, Gordon Henderson [EMAIL PROTECTED] wrote:
Not used it myself, (Microsoft? Outlook? What that then!) but a couple of
my clients are using Snap a number:
http://www.snapanumber.com/
Gordon
Oh, that _is_ nice :) Thanks for the pointer!
Steve
On 15/04/2008, Rudolf Ladyzhenskii [EMAIL PROTECTED] wrote:
Hi, all
I have SPA3000 (in Linksys reincarnation) and it has very annoying problem.
Sometimes, incoming PSTN call drops the moment one picks up analog
phone on FXO port.
Most of the times it works, other times phone on FXS
On 21/04/2008, Benjamin Jacob [EMAIL PROTECTED] wrote:
Hello ppl,
Any way to do a re-invite and make RTP bypass Asterisk, after call
establishment.
In other words, I would like to control when to do the bypass work for
peer-peer RTP flow.
The issue is that I need to send DTMFs after
On 18/04/2008, Rosa De Santis [EMAIL PROTECTED] wrote:
Hi all.
Please, how can I configure an Asterisk PBX using an outbound proxy (that
resolve NAT Traversal)
I'm trying using the outboundproxy and outboundproxyport values in sip.conf
but the PBX don't get registered on the outbound
On 20/04/2008, robert boardman [EMAIL PROTECTED] wrote:
Hi All
I'm having problems with outboud ISDN calls,
They setup OK , and ring the other end OK, but when the call is answered
I get a disconnect cuase 17 with an error message in the console of
[Apr 15 08:06:13] DEBUG[4361]
On 21/04/2008, Benjamin Jacob [EMAIL PROTECTED] wrote:
Hello ppl,
Any way to do a re-invite and make RTP bypass Asterisk, after call
establishment.
In other words, I would like to control when to do the bypass work for
peer-peer RTP flow.
The issue is that I need to send DTMFs after
Hi,
Does anyone have a clever method of doing a conditional include =
line in the dialplan?
I want to include a bunch of standard contexts, but in the middle of
the bunch have one or more conditionally included, a bit like:
[default]
include = start-here
include = then-here
if $[{COMPANY} = A]
2008/4/22 Benjamin Jacob [EMAIL PROTECTED]:
[snip]
So, my question : once the SDPs are exchanged, what will happen to the DTMFs
sent by Asterisk using sendDTMF or the D option in dial.
[snip]
As far as I can tell, the D() option will be executed before the
re-invite takes place, so Asterisk
2008/4/22 Philipp Kempgen [EMAIL PROTECTED]:
Steve Davies schrieb:
Does anyone have a clever method of doing a conditional include =
line in the dialplan?
I want to include a bunch of standard contexts, but in the middle of
the bunch have one or more conditionally included
2008/4/22 Tzafrir Cohen [EMAIL PROTECTED]:
On Tue, Apr 22, 2008 at 09:55:37AM +0100, Steve Davies wrote:
Hi,
Does anyone have a clever method of doing a conditional include =
line in the dialplan?
I want to include a bunch of standard contexts, but in the middle of
the bunch
2008/4/22 Tzafrir Cohen [EMAIL PROTECTED]:
[snip]
A different approach:
[company-base](!)
; common settings
[company-A](company-base)
; specific for company A
[company-B](company-base)
; specific for company B
[company-C](company-base)
; specific for company C
Keep in
2008/4/23 Steve Edwards [EMAIL PROTECTED]:
[big snip]
Steve,
Fantastic examples. Many thanks for the feedback :)
Regards,
Steve
___
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asterisk-users mailing list
To UNSUBSCRIBE or
2008/4/24 Jared Smith [EMAIL PROTECTED]:
On Thu, 2008-04-24 at 17:50 +0200, harry wrote:
The PBX is connected to an ISDN-30 connection. Are there any modules
for playing MP3 files, so I can use them with commands like Play() and
Background()?
If I were you, I'd transcode the files to
2008/4/24 Ken Williams [EMAIL PROTECTED]:
Came upon a problem today that I thought I'd see if it's by design, if I'm
missing an option somewhere, or if my fix is the way to fix it.
We setup a remote location with a server, same as we've done with others,
but for some reason when they would
2008/4/29 Tony Mountifield [EMAIL PROTECTED]:
[snip]
What values do you have in zapata.conf for pridialplan, internationalprefix,
nationalprefix and localprefix?
Try each of the following two sets of parameters:
(A)
pridialplan=dynamic
internationalprefix=00
nationalprefix=0
Hi,
I read the WiKi, which implied there was a way of working around this,
but the HTML nature of the WiKi seems to have destroyed some of the
output so I cannot see the correct answer...
I would like to match a special case of a number dialled 0x, now
normally I would simply do:
exten
2008/5/13 Stefan Schmidt [EMAIL PROTECTED]:
Hello,
i´ve a problem i dont find the reason for. An incoming call coming over
iax is connected to a Sip phone. Until the phone picks up the call i
could hear moh without problems. Then the phone sets the call on hold
and opens another call to
2008/5/12 Steve Davies [EMAIL PROTECTED]:
Hi,
I read the WiKi, which implied there was a way of working around this,
but the HTML nature of the WiKi seems to have destroyed some of the
output so I cannot see the correct answer...
I would like to match a special case of a number dialled
You might want to check the date on that email...
2008/5/18 Andrea Cristofanini [EMAIL PROTECTED]:
Hi
I just saw this now !
does the microphone and speaker works ?
Can you use it like softphone for recive calls ?
Regards Andrea
C F ha scritto:
TODAY I have managed to hack the iPhone and
2008/5/19 Steven Howes [EMAIL PROTECTED]:
On 18 May 2008, at 23:42, Andrea Cristofanini wrote:
Hi
I just saw this now !
does the microphone and speaker works ?
Can you use it like softphone for recive calls ?
Regards Andrea
Since when is Asterisk a SIP client.
Check the date on the
2008/5/21 The Asterisk Development Team [EMAIL PROTECTED]:
The Asterisk.org development team has released Asterisk version 1.4.20.
[snip]
Does this mean that the fixed IAX security fix for 1.2.28 (1.2.28.1?)
will also be officially released now?
If it helps, I have given 1.2 trunk some light
2008/6/4 Brent Davidson [EMAIL PROTECTED]:
[snip]
We're currently using Asterisk 1.4.19, Zaptel 1.4.10,
Oslec SVN, Rhino R4FXO-EC cards, and Snom 300 Phones.
[snip]
Just a small aside...
You go to the trouble of building/using Oslec, and then use hardware
EC? Very odd. Does Oslec understand
2008/6/6 Ron Wellsted [EMAIL PROTECTED]:
Kevin Smith wrote:
Hi everyone,
Perhaps I am just mis-reading the documentation, but for call recording,
is it possible to record the conversation over a SIP channel? We have
call recording preformed on all of our ZAP connections, but I was
wondering
2008/6/9 Sherwood McGowan [EMAIL PROTECTED]:
Sherwood McGowan wrote:
Gentlemen,
I have a particularly strange problem, just started happening. One of
my clients is running Asterisk 1.2.28 and has mysql realtime queues.
We log in a member, and then place a test call to the 0 queue but
since
2008/6/21 Gert-Jan de Boer [EMAIL PROTECTED]:
Hi All,
I am still working on an TAPI solution for my customer.
They are trying to connect Asterisk to Navision.
I am using the Activa TSP and an TAPI connector for Navision.
When a customer calls I use the following rule:
exten =
2008/6/23 Gert-Jan de Boer [EMAIL PROTECTED]:
Thanks.
I will ask on the ActiveTSP forum page.
I was hoping there were people who have experience with this kind of setup.
I use the LOCAL/ extensions because the direct call through SCCP did not
work. I tried executing it through a macro.
2008/7/1 Loic Didelot [EMAIL PROTECTED]:
Hello,
one of my customers complained about bad voice quality on several calls,
so I programmed a button on each phone which users can hit if they have
audio drops and echo.
I did this to check if there is a common recurrent problem to a given
2008/7/4 RoLaNd RoLaNd [EMAIL PROTECTED]:
hi all,
is there any way of removing this line from showing on the console?
my verbosity level is 3.
and this is the following output on cli 24/7 unless its interrupted by the
msgs tht really counts like connected sip and so on..
[snip]
Stop
2008/7/23 MFH [EMAIL PROTECTED]:
Noah Miller wrote:
Hi Daniel -
How can I made a 3-way conference betwwen IAX channels?
My current version is: 1.4.21.1
Anytime you need a call with more than 2 parties, you need to use some
kind of conferencing application. The default conference
2008/7/25 Olivier [EMAIL PROTECTED]:
Completing and reformulating previous questions :
4. Is there any tarball for 1.22fork ? I can see files but no tarball. Maybe
this comes from the fact 1.22fork is not stable enough. Is this correct ?
5. This
2008/7/25 Olivier [EMAIL PROTECTED]:
Completing and reformulating previous questions :
4. Is there any tarball for 1.22fork ? I can see files but no tarball. Maybe
this comes from the fact 1.22fork is not stable enough. Is this correct ?
5. This
2008/7/25 Olivier [EMAIL PROTECTED]:
It has not been Formally tagged for release, but I would suggest
downloading the version at the URL you gave in your original post.
Just click the download icon on the page at
http://github.com/davetroy/astmanproxy/tree/master.
I helped develop this
Hi,
Sorry this is so long, but I am reasonably desparate.
I am having real fun with hooking an Avaya system to Asterisk using
ISDN30. I have the ISDN signalling all sorted one way, and can pass
calls from the real world (ie. the telco and asterisk) TO the avaya
box, and it accepts that and sets
Good question, I'll check.
Regards,
Steve
2008/8/6 Tom Lynn [EMAIL PROTECTED]:
Steve, what kind of Avaya system is this? They make several.
On Tue, Aug 5, 2008 at 11:36 AM, Steve Davies [EMAIL PROTECTED] wrote:
Hi,
Sorry this is so long, but I am reasonably desparate.
I am having real
I am told it is an IP Office 400 series.
I have not been on site physically which does not help.
Regards,
Steve
2008/8/6 Tom Lynn [EMAIL PROTECTED]:
Steve, what kind of Avaya system is this? They make several.
On Tue, Aug 5, 2008 at 11:36 AM, Steve Davies [EMAIL PROTECTED] wrote:
Hi
2008/8/5 Steve Davies [EMAIL PROTECTED]:
Hi,
Sorry this is so long, but I am reasonably desparate.
I am having real fun with hooking an Avaya system to Asterisk using
ISDN30. I have the ISDN signalling all sorted one way, and can pass
calls from the real world (ie. the telco and asterisk
2008/9/3 Jim Boykin [EMAIL PROTECTED]:
Brent/Steve, Thanks for the answer. Point here is that asterisk
already knows about first leg and the codec so shouldn't it select the
best codec for second leg to match first leg. Instead asterisk is
selecting first codec in order.
To illustrate, if
2008/9/8 Max Alex [EMAIL PROTECTED]:
Hi all,
I have a trixbox2.6.1 on my one server,
i have configured sangoma A200/Remora FXO/FXS Analog AFT card on that
server,
from my zap line the incoming faxes are coming, i have setup the did for zap
channel.
my question is when i am getting any
2008/9/8 Rodrigo Pinto [EMAIL PROTECTED]:
Hello,
Someone has worked with the astmanproxy? I am stating the use
astmanproxy with the AutoFilter ON, but it does not filter all events, I
am doing the test and he is still receiving some events to other
channels.
He managed to filter all
2008/9/11 Stefan Schmidt [EMAIL PROTECTED]:
Chris Bagnall schrieb:
snip
Hello,
first you have to use the lastest firmware for the spa962. When you have
this installed you will see a input field for pickup code in the webif
for the spa932
just put a # after the pickup code you want to
2008/9/11 Stefan Schmidt [EMAIL PROTECTED]:
Steve Davies schrieb:
Thanks for that excellent information - Now does anybody know the XML
to provision that field? Normally you take the text on the screen
Call Pickup Code and replace space with underscore
Call_Pickup_Code ua=na *8
2008/9/29 Olivier [EMAIL PROTECTED]:
Hi,
Reading http://www.asteriskguru.com/tutorials/bri.html , it seems PtP is
the way to connect businesses but if you read
http://public.swbell.net/ISDN/connect.html you would think the opposite.
Can anyone elaborate a bit PtP or PtmP respective
2008/9/29 Olivier [EMAIL PROTECTED]:
From http://public.swbell.net/ISDN/connect.html :
If you only intend to connect a single device/application to your ISDN
line, then you only need the point-to-point configuration. With the
point-to-point configuration you are assigned a single phone
2008/11/1 Rodolfo Alcazar Portillo [EMAIL PROTECTED]:
Hi. I have a SPA3102 updated with with Software Version: 5.1.7(GW).
I have this settings on Voice/Regional:
Interdigit Long Timer: 10
Interdigit Short Timer: 3
Anyway, when hooking up (without dialing anything), the timeout starts
Hi,
I would just like to know if any work was ever done on COLP or its
related cousins? The last evidence of it seems to be about 2 years old
when K.Flemming and Olle both showed some mild interest. I am not sure
how well that code would apply to today's Asterisk.
(I realise that this is sort
On 4/20/07, James FitzGibbon [EMAIL PROTECTED] wrote:
On 4/20/07, Olivier [EMAIL PROTECTED] wrote:
Are you sure eyeBeam config are binary ?
I thought it was just the case for XLite.
Having looked into it further, you're right. For some inexplicable reason
it's not putting the files where
On 4/25/07, Barton Fisher [EMAIL PROTECTED] wrote:
Michael Kamleitner wrote:
I'm currently trying to implement a simple voicebox-system.
for demonstration purposes, I've successfully connected my cellphone
via bluetooth using the current chan_cellphone-patch on the current
SVN-version of
This is very interesting. I am now getting this double-digit behaviour
occasionally, and only on IAX channels (so far). Did anyone come up
with a solution or a way to improve matters?
The scenario where I get this is:
PSTN - Provider - IAX - Gateway - IAX - Customer
So I will go and do some
in the
audio stream to be detectable by Asterisk, but it ALSO sends an
rfc2833 packet and both are detected and sent onwards!
I would still be interested in any ways to improve this!
Thanks,
Steve
On 5/3/07, Steve Davies [EMAIL PROTECTED] wrote:
This is very interesting. I am now getting
On 5/3/07, Ken Leland III [EMAIL PROTECTED] wrote:
When dtmfmode is set to inband for SIP, and i originate a call from sip
out to the PSTN, I can hear the DTMF digit twice in the audio stream.
Once very briefly and once for normal duration.
Our Theory: While Asterisk is parsing the DTMF, for a
As usual, it is worth searching the WiKi for answers to this sort of question:
http://www.voip-info.org/wiki/view/Asterisk+Fax+to+email
This is not the only answer.
Regards,
Steve
On 5/9/07, Josu Lazkano Lete [EMAIL PROTECTED] wrote:
Hello everybody,
I am receiving faxes and I don`t know
On 5/31/07, Carlos Chavez [EMAIL PROTECTED] wrote:
Sometimes I get the following error on the console:
[May 31 11:14:01] ERROR[23502]: cdr_addon_mysql.c:230 mysql_log:
mysql_cdr: Failed to insert into database: (1062) Duplicate entry
'1180628004.3214' for key 1-- Zap/38-1 answered
On 5/28/07, voip crazy [EMAIL PROTECTED] wrote:
Hello all,
Some of you are using astmanproxy with asttapi or activa TSP?
How does you make to work?
Thanks
VoipCrazy
The latest 1.21 (?) version should work okay with Asttapi, but has a
flaw which stops Activa TSP from working. I did have a
Hi list,
When this question came up, I realised how little I know about ISDN
data calls (the sort used for ISDN video-conferencing etc), so I
thought I would solicit pointers here.
I have a requirement for an Asterisk-based system to connect to an
ISDN30 line (using Sangoma hardware), and to
On 6/29/07, Ed Nuñez [EMAIL PROTECTED] wrote:
What is a good solution for playing music on hold on the 1.2 branch. I do
not want to use mpg123 because last time I used it in a production server it
caused many problems. The MPG123 process was taking about 60% of my Xeon
CPU.
For minimum
Hi,
Just a quick question. Is there a way when making an IAX call to
transmit some additional call-data, perhaps in a variable? I could
overload callerid-name, but that is nasty and ugly :)
Thanks for any suggestions.
Regards,
Steve
___
--Bandwidth
I have not confirmed this independently, but I believe this is fixed
if you disable the Show message light when a call is missed feature
in the phone config. Alternatively, try pressing X to clear the
missed call indication before pressing Retrieve
Might work... Might not :)
Steve
On 1/17/07,
On 1/25/07, Peter Mitchell [EMAIL PROTECTED] wrote:
Has anyone seen this issue with asterisk running like a dog when the
internet is down ? Internal calls, incoming ISDN calls etc all seem to be
affected. There is a local DNS server that is always available so I'm not
sure why asterisk is so
On 1/26/07, Jon Schøpzinsky [EMAIL PROTECTED] wrote:
Hello List
I am having a rather big problem with a sangoma A104 card, I just installed
to replace a Digium TE410 card, that was acting up.
But now we have a problem with the sangoma card. It runs great after being
started, and calls proceed
Which asterisk versions etc etc?
On 1/26/07, Jon Schøpzinsky [EMAIL PROTECTED] wrote:
I am running the newest version, from the sangoma wiki.
Jon
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Davies
Sent: 26. januar 2007 10:56
To: Asterisk
I would be interested to know whether this
http://bugs.digium.com/view.php?id=8376
patch makes any difference. The problem is almost certainly not caused
by Centos (which is widely used with Asterisk) or EPIA (which I use
lots).
Regards,
Steve
On 1/29/07, Erick Perez [EMAIL PROTECTED] wrote:
On 1/29/07, Dominik Zalewski [EMAIL PROTECTED] wrote:
Hi All,
I'm using Asterisk 1.2.14 under openSuSE 10.2 with kernel 2.6.18. I have
Wildcard TDM400P card and D-Link DPH-120S and DPH-140S SIP phones. I would
like to be able to pickup ringing extention from any SIP phone using Pickup()
On 1/29/07, Dominik Zalewski [EMAIL PROTECTED] wrote:
Hi All,
I'm using Asterisk 1.2.14 under openSuSE 10.2 with kernel 2.6.18. I have
Wildcard TDM400P card and D-Link DPH-120S and DPH-140S SIP phones. I would
like to be able to pickup ringing extention from any SIP phone using Pickup()
that in SVN 51223 it was implemented, im running
51363. However I may be wrong. I will apply that patch and let you
know.
Thanks for the pointer.
should I leave asterisk as -march=i586? or 386?
On 1/29/07, Steve Davies [EMAIL PROTECTED] wrote:
I would be interested to know whether this
http
*ping*
I am interested in this too if anyone has any clues? I am looking to
do this on a Cisco 7941/7961.
Thanks,
Steve
On 1/26/07, Naija Man [EMAIL PROTECTED] wrote:
Hello,
We have an asterisk system with about 40 cisco 7940/7960 phones and a few
linksys SPA941. I recently analyzed our
Hi,
I have a problem understanding which 'h' (hangup) extension is used in
which case - It seems to vary depending on channel type. Assuming the
following simplified dialplan:
[macro-faxhere]
exten = s,1,rxfax(file)
exten = h,1,NoOp(Hangup in macro)
[fax]
exten = _X.,1,Macro(faxhere)
exten =
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