Date: 11/14/2003 03:43 AM
Fix your system clock!
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On Sat, 27 Aug 2005, Eric Wieling aka ManxPower wrote:
Date: 11/14/2003 03:43 AM
No no - that IS the date where he is living ;-)
Steve
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Hi, all. I am building a SER+asterisk PBX airming at around 10k
persons' usage. For authentication purpose I am in favor of ldap
storage, while I am not sure the current ldap module for
asterisk(0.9.9.2) is stable enough? sorry I do not master the proper
testing mechanisms to find out myself.
Is it practical to 'assume' that in your case mentioned above that
#1 is not going to occur again (since I assume when you say 'line'
you are referring to an outside pstn line), and, #2 is in a mode
of fine-tuning the training when in fact you'd really like it to
start the
Now I'm worried - we have exactly the same problem, but were going to
upgrade to 1.2. Now it seems as if CVS-HEAD has the same issue.
We have a TE405P, with 80 cisco7960 phones connected to a isdn30 pri.
The same issues ocurr - Busy on inbound calls, cannot place outbound,
nothing in the
Is it practical to 'assume' that in your case mentioned above that
#1 is not going to occur again (since I assume when you say 'line'
you are referring to an outside pstn line), and, #2 is in a mode
of fine-tuning the training when in fact you'd really like it to
start the coarse-training
Dan Austin wrote:
3. Should I consider the current features a release point, or
is there
something I missed that should be added before
packaging it?
I appreciate the feedback I have received since the last announcement,
and
apologize that I allowed work to get
ok then go ahead and try. but don't expect too much .the voices will be
out of time .
Sean Rima wrote:
On Fri, 14 Nov 2003 01:43:21 -0800
cxpcman [EMAIL PROTECTED] wrote:
Sean Rima wrote:
I live in a very rural area, BB access will never happen and the only
choice I have it
Matthew Boehm wrote:
I am currently working on a new updated SNMP module for Asterisk. It
is currently in testing stage as I am having a problem with channel
listings.
It will support read-only attributes associated with Asterisk such as
number channels, number of calls, all config file
On Fri, 14 Nov 2003 04:38:59 -0800
cxpcman [EMAIL PROTECTED] wrote:
You must have future vision :)
ok then go ahead and try. but don't expect too much .the voices will be
out of time .
Sadly I am aware of this now but ISDN is slow slow :)
Sean
--
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Gossamer Spider Web of Trust
On 8/6/05, Eric Wieling aka ManxPower [EMAIL PROTECTED] wrote:
Peter Svensson wrote: On Sat, 6 Aug 2005, Robert Goodyear wrote:Can you educate us all on the appropriate circumstances in which touse 'r'? Some devices (voip phones, softphones) do not generate in band progress
information when
I think invite deafen feature in conference may be of great help
On 8/27/05, John Fawcett [EMAIL PROTECTED] wrote:
Dan Austin wrote:
3. Should I consider the current features a release point, or
is there
something I missed that should be added before
Thanks Tom!
Wasn't aware of that. Installed Asterisk by hand to do some learning, but
getting a Cisco phone up and running is not something that I'd like to spend
more time than necessary to do.
I'll let you know how it works out.
Bjorn
-Opprinnelig melding-
Fra: [EMAIL PROTECTED]
On Sat, 27 Aug 2005 09:43:33 +0100
Sean Rima [EMAIL PROTECTED] wrote:
On Fri, 14 Nov 2003 04:38:59 -0800
cxpcman [EMAIL PROTECTED] wrote:
You must have future vision :)
ok then go ahead and try. but don't expect too much .the voices will be
out of time .
Sadly I am aware of
Hi:
Does anyone knows how to connect a Nextone and an Asterisk-Linux ?
I want to pass all incoming calls from syrinx to metroredout.
Some of the call ring and are passed, but I am getting the following error:
Aug 27 11:41:48 NOTICE[5936]: rtp.c:277 process_rfc3389: Comfort noice
support
I am using viatalk as my voip provider and they use dtmf=rfc2833, but
asterisk is not seeing any of the dtmf. I am using CVShead as of
8/26/05. Nothing in the logs indicates a dtmf is being seen. If I
use my pots line it sees it fine.
Any assistance would be appreciated.
--
Your life is like
Hi,
just upgraded to 1.2.0 beta 1 on my 5.4 FreeBSD machine. The compilation
works fine but when starting * I receive errors when loading modules:
Undefined symbol ast_register_file_version for chan_modem.so,
res_musiconhold.so and probably other modules. Before building 1.2.0 I
removed all
Now I'm worried - we have exactly the same problem, but were going to
upgrade to 1.2. Now it seems as if CVS-HEAD has the same issue.
We have a TE405P, with 80 cisco7960 phones connected to a isdn30 pri. The
same issues ocurr - Busy on inbound calls, cannot place outbound, nothing
in the
Hello Michael and the list,
exten = 0,1,Dial(Zap/3/${EXTEN:1}) ; call to RTC FXO#3, digit #1 (0)
suppressed
I hope this could help some.
Best Regards,
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Michael
Felder
Envoyé :
I had the same problem, and have beenworking with Sherwood. he suggested that I try dtmf=inband, which has been working great for me.
On 8/27/05, John covici [EMAIL PROTECTED] wrote:
I am using viatalk as my voip provider and they use dtmf=rfc2833, butasterisk is not seeing any of the dtmf. I am
Alternatively you could do this
exten = _0.,1,Dial(Zap/3/${EXTEN:1})
This would allow you to dial the whole number including the zero without
waiting to confirm the dialtone
Mark
[EMAIL PROTECTED] wrote:
Hello Michael and the list,
exten = 0,1,Dial(Zap/3/${EXTEN:1}) ; call to RTC FXO#3,
Title: Message
I'm
assuming no apps/scriptsexist which completes this?
Can
someone please confirm thatif I use a FQDN in sip.conf for my external IP,
the FQDN is only resolved at the time of loading, therefore if my IP changes
after sip is loaded, I will have to manually reload
Try allow=all
Turn on sip debugging and see if you can see what the codec deadlock is.
- Original Message -
From: Patricio Ku [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Saturday, August 27, 2005 5:50 AM
Subject: [Asterisk-Users] Asterisk conection to Nextone, codec
Take
a look at www.generationd.com -
they have a free script for setting up ISA 2004 for SIP access (running on same
box as ISA). They have experience setting up ISA Asterisk for
customers (but charge for support since that's their
business)
From: Dean Collins [mailto:[EMAIL PROTECTED]
All right, installed Asterisk @ home and sat up DHCP as per described in the
handbook.
Defined a subnet with the Asterisk box at 192.168.1.1 as you can see in the
dhcpd.conf quoted below. Chmod'ed all files in the /tftpboot directory to
666, the directory itself to 777. However, still nothing but
On Friday 26 Aug 2005 14:54, Julian Lyndon-Smith wrote:
I cannot reach voip-info - is it just me or is the site not available ?
There is a bad route being propogated.
B
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So the only thing we have in common is the remote monitoring ...
Julian
[EMAIL PROTECTED] wrote:
Now I'm worried - we have exactly the same problem, but were going to
upgrade to 1.2. Now it seems as if CVS-HEAD has the same issue.
We have a TE405P, with 80 cisco7960 phones connected to a
Greetings all,
I just installed the beta now my SIP phone doesn't correctly hang up and
clear trunks once the call is answered.
First one is fine because it didn't answer on the far end. The second
one stayed connected.
-- Executing SetCallerID(SIP/100-5ab0, 516301) in new
stack
--
I was thinking of that, but I was hoping I would not have to do that
because I had a previous provider and all kinds of trouble going the
other way -- how's your outbound dtmf working?
on Saturday 08/27/2005 Justin Richards([EMAIL PROTECTED]) wrote
I had the same problem, and have been working
On 8/27/05, John covici [EMAIL PROTECTED] wrote:
I was thinking of that, but I was hoping I would not have to do thatbecause I had a previous provider and all kinds of trouble going the
other way -- how's your outbound dtmf working?
outbound so far, changing to inband made my company's VM system
Is it practical to 'assume' that in your case mentioned above that
#1 is not going to occur again (since I assume when you say 'line'
you are referring to an outside pstn line), and, #2 is in a mode
of fine-tuning the training when in fact you'd really like it to
start the
So the only thing we have in common is the remote monitoring ...
Are you using:
1) Realtime (and if so, with mysql, odbc, etc?)
2) Logging CDR records? (and if so, how)
This post looks like it could pertain to the same problem:
Hello,
Can we store voice messages in a database instead of
files.
Regards
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On Friday 26 Aug 2005 14:54, Julian Lyndon-Smith wrote:
I cannot reach voip-info - is it just me or is the site not available ?
There is a bad route being propogated.
Its working from the US. Lasts steps in traceroute are:
1181 ms80 ms81 ms sl-internap-95-0.sprintlink.net
Hi, i have one Asterisk with a Digium E1 card, and a Meridian Nortel
Release 11.
I need to connect both of them. We are using MFC/R2 for this..
The Diagram:
[ NORTEL ] ( AMI )
(DIGIUM) [ ASTERISK]
we have green light at the digium card, and at
I guess you need to use DeadAGI(), please check it out on voip-info.org
best regards
On 8/25/05, Patrick Tracanelli [EMAIL PROTECTED] wrote:
Patrick Tracanelli wrote:
Hello List,
This is my first message herein. I was playing around with System() and
AGI() and found out something I
My friend has an NEC Aspire S and he would like to integrate Asterisk with
the existing PBX. What he needs to do for now is to have remote extension
register to Asterisk and be able to place calls between NEC Aspire and
Asterisk extension transparently. I have read doing something similar using
a
And this has to do with Asterisk exactly how?? This is not a MS
Windows support group.
On Fri, Aug 26, 2005 at 06:42:33AM -0700, Ellafi Fituri said:
Hi,
I have 2 CDs that would like to make a backup of , I am having a hard time
doing. I have tried NERO ver.6 but it does not work
Hi!
I haven't done this myself (yet), but look at app_devstate that comes
with the bristuff patches to toggle one of the SNOM LEDs as needed.
http://www.voip-info.org/tiki-
index.php?page=Asterisk+cmd+BristuffDevstate
Other - more clumsy - ideas would include a) making Asterisk call itself
On the Snom phones you can use a SIP MESSAGE to overwrite the idle screen text
with a given text message. Maybe that is helpfull for your scenario.
Regards
Nils Ohlmeier
On Friday 26 August 2005 01:09, Franklin Webb wrote:
I talked to Digium about this and they are saying the best thing may
Hi,
sometimes it is not available.
Be patient, wait 10 minutes and try again.
Giorgio
Julian Lyndon-Smith wrote:
I cannot reach voip-info - is it just me or is the site not available ?
Julian
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I use Asterisk Realtime in an enterprise ITSP environment,
and while it's not stable, this is because of the bugs found in the cvs head,
the mysql portion has been rock solid as far as we can tell. However, my lead
engineer couldn't get realtime extensions.conf to work
properly.
I love it
Well, I changed to inband and that seemed to help.
Thanks.
on Saturday 08/27/2005 Justin Richards([EMAIL PROTECTED]) wrote
On 8/27/05, John covici [EMAIL PROTECTED] wrote:
I was thinking of that, but I was hoping I would not have to do that
because I had a previous provider and all
Hi Gonzalo.
You maybe need a couple of internal extensions of your NEC PBX
connected directly to a FXO porst in the asterisk box , then you can
register your remote extensions with SIP and put calls to the PBX or
to the PSTN directly.
Regards.
Athiel
Mexico.
2005/8/27, Gonzalo Gonzalez
It is NOT working from the UK.
There is some loop in a routing table somewhere which goes round and
round. I have been trying for 2 days to reach it but don't know who to
complain to to get it resolved.
Rgds
Tim
Rich Adamson wrote:
On Friday 26 Aug 2005 14:54, Julian Lyndon-Smith wrote:
Hi list,
I'm in central-europe and signed yesterday a broadvoice account. My
Asterisk box is CVS 2005-08-25.
Problem I face is:
Failed to authenticate on REGISTER to '[EMAIL PROTECTED]'
(Tries 2) then
Registration for '[EMAIL PROTECTED]' timed out and finaly
Giving up forever to register
we used sipp, the opeh source benchmarking software sponsored by HP. We can
send you our benchmark, if you like.
We did run into a problem, though. The benchmark suite core dumps on us at
about 5100 simultaneous SIP streams.
Regards,
Paul
Paul Mahler
www.signate.com
Perfect timing, 1.2beta1 is released and the bug tracker is
broken!
Try to submit a bug and get error 1303, invalid field value.
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What was the name of the web app that let you run an asterisk
extension from your web page so people could call you via your web page?
Cheers,
Dean
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Damon Estep wrote:
ast_expr2f.c:1784: warning: no previous prototype for âast_yyget_columnâ
ast_expr2f.c:1860: warning: no previous prototype for âast_yyset_columnâ
ast_expr2f.c:1259: warning: âyyunputâ defined but not used
These are not errors, that's why they are called 'warnings'.
I have had uniformly bad experiences with soft phones when there are network
issues. Hardware phones seem to work much better if there are network problems.
For example, I have been able to make fine calls over a wireless link I use
with a cisco 7960, but NO softphone works over the same link.
If its NOT working for you, please send a traceroute to:
[EMAIL PROTECTED]
Thanks.
Jim
[EMAIL PROTECTED]
- Original Message -
From: Bob Goddard [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, August 27,
I am new to asterisk and need to dig up some info on how to set it all
up. It looks a bit daunting especially all the options available in the
.conf files.
I have 2 SIP phones, GXP2000 and a budgettone 100.
phone1 - 192.168.0.160/24 extension 1000
phone2 - 192.168.0.161/24 extension 1001
I'm hoping for help on installing a second TE410P card in a system. I'm
confused about 2 things.
1. The rotary switch on the first card should be set to 0 and second card 1?
2. When adding the second card to zaptel.conf - is the new card span=5 or
span=1:
example:
span=1,0,0,d4,ami or
Yes. Look in the apps/Makefile for USE_ODBC_STORAGE and read in the docs/
for a table structure.
Right now it is ODBC only.
-Matthew
From: harry gaillac [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Sat, 27 Aug 2005
Hello
There is a loopback between these IP adress
63.216.31.38ge3-1.br01.atl01.pccwbtn.net
63.216.31.174 wvfiber.ge2-2.br01.atl01.pccwbtn.net
63.216.31.173 ge2-2.br01.atl01.pccwbtn.net
63.216.31.174 wvfiber.ge2-2.br01.atl01.pccwbtn.net
63.216.31.173 ge2-2.br01.atl01.pccwbtn.net
Thanks for help.
Harry
--- Matthew Boehm [EMAIL PROTECTED] a écrit :
Yes. Look in the apps/Makefile for USE_ODBC_STORAGE
and read in the docs/
for a table structure.
Right now it is ODBC only.
-Matthew
From: harry gaillac [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List -
I am using Telappliant PSTN - VOIP Goegraphic number. They are
passing me the call to my * box by IAX.
I have configured spandsp, libtiff, etc. It will pick up the call,
start to talk to the fax machine but at the same point every time it
hangs on and then hangsup.
Here is my iax.conf:
Damon Estep wrote:
Try to submit a bug and get error 1303, invalid field value.
I just entered a test bug and it worked fine.
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I just pm'd you the error and bug report values, perhaps it is specific
to a value I have entered.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming
Sent: Saturday, August 27, 2005 11:39 AM
To: Asterisk Users Mailing
I have this error
Aug 27 13:31:46 NOTICE[2863] chan_sip.c: stale nonce received from
'656720189sip:[EMAIL PROTECTED];user=phone'
generated for two equipment hardwired in asterisk. Some friend can help me?
Thank's
João Carlos Moura
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-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming
Sent: Saturday, August 27, 2005 9:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] bug tracker bug?
Damon Estep wrote:
I have the same problem. I've built cvs head countless times, first time
I've seen this issue.
1.2 beta
Julian.
Damon Estep wrote:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming
Sent: Saturday, August 27, 2005 9:30
And is does not exist in current cvs head either :) -- that builds fine.
I have the same problem. I've built cvs head countless times, first time
I've seen this issue.
1.2 beta
Julian.
Damon Estep wrote:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
Can anyone shed some light on which of these packages are
required and what component requires them? I am in the habit of putting them
on, but in a few cases am not sure if they are still (or were ever) needed.
qt-devel
rpm-build
gcc
gcc-c++
redhat-rpm-config
gtk2-devel
y
I have splitters on 2 of the 3 PSTN lines. As I mentioned in my previous
posts, the echo performance of my system is not so bad, but does anybody
know if ADSL splitters may cause echo? After all, splitters have some
circuitry, and my wild guess is that that may influence the characteristics
of
I'm going to build an Asterisk system on a 1G Compact Flash card with
NFS mounts for /var/spool, /var/log, etc. Does anyone have information
on which packages are required for the CF card?
Also, I would like to set the CF card to read only. Does anyone have
information on which directories are
Anyone know what this means:
Aug 27 12:55:02 WARNING[7799]: chan_sip.c:959 __sip_xmit: sip_xmit of
0x94c5c80 (len 734) to 192.168.2.29 returned -1: Invalid argument
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--- Sean Rima [EMAIL PROTECTED] wrote:
On Sat, 27 Aug 2005 09:43:33 +0100
Sean Rima [EMAIL PROTECTED] wrote:
On Fri, 14 Nov 2003 04:38:59 -0800
cxpcman [EMAIL PROTECTED] wrote:
You must have future vision :)
ok then go ahead and try. but don't expect too much .the voices
On Sat, 27 Aug 2005 12:14:55 -0700 (PDT)
Julius Igugu [EMAIL PROTECTED] wrote:
ok then go ahead and try. but don't expect too much .the voices will be
out of time .
Sadly I am aware of this now but ISDN is slow slow :)
Thinking here and this can be bad :) At
I know of good way to solve this problem. I have been authorize by my company to try to a group of people and businesses to give donations to get Digium to fix this issue. We will start the pot at $200. Are there any takers?
On Sat, 2005-08-27 at 10:08 -0400, [EMAIL PROTECTED] wrote:
So
On Sat, 2005-08-27 at 12:14 -0700, Julius Igugu wrote:
I use a satellite connection and VOIP is ok!
It depends,mostly, on what you expect!
There's an inherent delay in the system usually about 700ms - 800ms! but this
is bearable.
That depends on how you define satellite and the
Is anybody using AG-468 or AG-268 from ATCom?
I can not register the unit with Asterisk.
--
#Joseph
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Already have that..
On 8/27/05, Eric Wieling aka ManxPower [EMAIL PROTECTED] wrote:
Eric Bishop wrote:
Hi all,
Our Asterisk box sends calls outbound via either SIP (through our VoIP
provider) or an E1 PRI (directly connected via a TE410P). When we dial
a number that is engaged via
Found the problem... We recently upgraded our IOS to 12.4 on our
1760 cisco router. This introduced the choppy ringing. Previously,
I had mentioned that this issue didn't occur when using ulaw. I was
incorrect. This bug effected both g729 and g711ulaw codecs. After
checking the cisco
RG384 - Gilat!
--- Sean Rima [EMAIL PROTECTED] wrote:
On Sat, 27 Aug 2005 12:14:55 -0700 (PDT)
Julius Igugu [EMAIL PROTECTED] wrote:
ok then go ahead and try. but don't expect too much .the voices will
be
out of time .
Sadly I am aware of this now but ISDN is
Hi all,
I am struggling with the following and I can't get it work:
In the Netherlands where I live it is possible to use special codes
(aka vertical service codes) to set special 'behaviour' of phonecalls.
So e.g. when I want to dial out with a normal phone and I dial
*31*phonenumber to dial
Hello Julius,
Saturday, August 27, 2005, 10:38:18 PM, you wrote:
RG384 - Gilat!
That is the one I was thinking of going for :)
Sean
--
+---+
|VOIP= FreeWorldDial: 689482 VOIPBUSTER: thecivvie |
|GPG Key
My phone still says Not-Registered. I have a Polycom SoundPoint 600 SIP
phone.
Here is my sip.conf file:
;
; SIP Configuration
;
[general]
context=default; Default context for incoming calls
port=5060 ;added
bindport=5060; UDP Port to bind to (SIP standard port is 5060)
BTW, any password that is filled in is a test password and doesn't
actually exist :)
Joshua Abbott wrote:
My phone still says Not-Registered. I have a Polycom SoundPoint 600
SIP phone.
Here is my sip.conf file:
;
; SIP Configuration
;
[general]
context=default; Default context
I am currently using the Manager API to place a few calls. I have more than one VSP available.
I was wondering how to best tell a call failed to move on to the next VSP.
I see messages like this, which is an obvious failure, and I would then move on to the next VSP.
Event: Hangup
Cause: 3
I have seen talk of adding the capability to pass variables on an IAX2 call.
I would like to know if this is possible, yet.
Thanks!
Sincerely,
Trevor Hammonds
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Folks,
I am a newbie to the VOIP world and have a question (might as well
sound silly to some). I would like to set up a PC-to-Phone call from
my desktop to a regular PSTN number. Does the Asterisk PBX itself act
as a VOIP-PSTN gateway or do I have to subscribe to a VOIP provider
for this? Are
Hi, i have one Asterisk with a Digium E1 card, and a Meridian Nortel
Release 11.
I need to connect both of them. We are using MFC/R2 for this..
The Diagram:
[ NORTEL ] ( AMI )
(DIGIUM) [ ASTERISK]
we have green light at the digium card, and at
I have splitters on 2 of the 3 PSTN lines. As I mentioned in my previous
posts, the echo performance of my system is not so bad, but does anybody
know if ADSL splitters may cause echo? After all, splitters have some
circuitry, and my wild guess is that that may influence the
Does anyone know if gotoiftime can take any subset of 7 days
for the days of the week or only a contiguous range?
I want to use gotoiftime to change dialplan behavior on Monday,
wedneday and Friday
-- Executing
GotoIfTime(Zap/8-1, 09:00-20:00|MON WED FRI|?21) in new
stack
Aug 27
My boss is complaining that the delay between speaking and hearing in a
meetme conference is noticeable and doesn't want to roll out our system
until I can eliminate the delay.
Personally, I don't think the delay is significant, but I don't sign his
check.
The system consist of 3 1u's, each
I think you are looking for something like this (from my dial plan):
...
[incoming]
; First, let's do the holidays
exten = 888,1,GotoIfTime(*|*|1|jan?holiday,s,1)
exten = 888,2,GotoIfTime(*|*|1|jul?holiday,s,1)
exten = 888,3,GotoIfTime(*|*|1|aug?holiday,s,1)
exten =
Maybe both your asterisk box and Nortel ox are both set to cpe
signaling?
In most cases where you are using asterisk as a VoIP gateway from a pbx
your pbx would be set to cpe and asterisk to net
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On
hello,
i change my OS from solaris 9 to solaris 10, tried
running make to install asterisk but i'm getting the error below:
make -C editline libedit.amake[1]: Entering
directory `/export/home/fst/ice/cvs/asterisk/editline'/bin/sh makelist -h
common.c common.h/bin/sh makelist -h emacs.c
Hello Group,
Current Setup:
- Eicon Quad BRI ISDN Card.
- 4 ISDN BRI (Telco: Telstra) Onramp2 services.
- Mode: Point2Point.
- 100 Indial Number ranges. Full National Number (9 digit format): BAAXX
where: B (Area code): 2/3/7/8
A (Normal Numbers)
X (99 Indial extensions)
eg: BAA00
I am looking for a single soft phone application that is capable of a
minimum of 24 concurrent lines. Suffice to say that I have a somewhat
unique application here, and I would like all connections active all the
time. I want to be able to switch between them for monitoring purposes,
placing
On Wed, 2005-08-24 at 15:04 +1000, Michael Felder wrote:
Hello Craig,
Yes I would like to dial 0 to get an outside line and dial tone, then
dial the number.
I have Polycom IP600 and IP 500s.
Mike
Just wondering how people who use 0 to access an outside line deal with
the following
I have just bought several Sipura SPA-841 SIP phones, and after some testing I
have found out that the volume received by other parties when calling using
the handset is very low. I've been able to reproduce this problem in the 3
phones I've tested so far. I've tried tweaking several
Or is it a monitoring application that you need? for instance, do you
need the ability to monitor active channels on request? The
description below isn't clear around what you mean in regard to
'monitorin' and 'placing the others on hold'. Normally you 'place
someone on hold' after you have
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