Well, in fact I have compilation error, Unixodbc is installed.
But I get :
gcc -shared -Xlinker -x -o res_odbc.so res_odbc.o -lodbc
/usr/bin/ld: cannot find -lodbc
gmake[1]: *** [res_odbc.so] Error 1
gmake[1]: Leaving directory `/usr/local/src/asterisk-1.2.0-beta1/res'
gmake: *** [subdirs] Error
hi all,
I wanted to call my asterisk on the Zapchannel with mainnumber+DID number:
it's ok for calls from handy and from sip i get the right
extension(DID) on phone, but when i call from an analog telephone the
DID number is not mentioned by asterisk.
in my zapata.conf:
overlapdial = yes
Orlando would be great in November, and you can get cheap and frequent
flights there from just about anywhere in the country. But it's about a
far from a central location for people in the US as you can get.
Although South American attendees can get direct flights to Orlando so
it is easier for
how can i dial DID to asterisk from analog telephone?
Zapata does not report a DID for the incoming calls - as I mentioned
in a recent post about seperating incoming calls on a TDM02B (see the
archive).
Effectively you will need to point each port to the appropriate
dialplan context in your
On Saturday 17 Sep 2005 23:20, Kevin P. Fleming wrote:
Damon Estep wrote:
I do not have the r option in the MOH class, but the files are played
in an order I can figure out, they do not appear to be random either,
same pattern repeats.
Oh come on, its obvious :-)
Have you figured it out
Better is really not to use G723... ;)
Can't you use others alternatives?
Regards,
-Mensagem original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Em nome de Asterisk guy
Enviada: segunda-feira, 25 de Abril de 2005 3:17
Para: Asterisk Users Mailing List - Non-Commercial
Damon Estep wrote:
That is what I thought I read somewhere, but it is not so. I will check,
but I THINK that * reads the file names left to right top to bottom and
my FC4 box lists them with an ls top to bottom left to right!
Oh you're right... they are not sorted. My fault, I forgot about
Steven Sokol wrote:
Actually, my wife's company holds several events each year in Orlando
and the turn-out is always great -- people love getting a bit of
vacation with their business travel. (An Universal's Islands of
Adventure is just plain awesome!).
If you are looking for the maximum
If you are looking for the maximum number of cheap flights from around
the world, and plenty of convention and room space, the answer is Las
Vegas :-)
I would definitively agree!
Senad
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Viva Las Vegas !!!
Neil
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Senad J
Sent: 18 September 2005 14:45
To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users
Mailing List - Non-Commercial Discussion
Subject: RE:
Hi all,
I have bought an Aastra 480i phone.
In order to configure the phone for using a TFTP server, I had to enter
the TFTP ip address directly in the phone, and then reboot the phone
again.
Is it possible to configure a DHCP server so it sends a TFTP server
coordinate for the phone to use?
On Sunday 18 Sep 2005 15:15, Francois Meehan wrote:
Hi all,
I have bought an Aastra 480i phone.
In order to configure the phone for using a TFTP server, I had to enter
the TFTP ip address directly in the phone, and then reboot the phone
again.
Is it possible to configure a DHCP server so
When I receive voicemail notify via e-mail I would like receive not the
phone-number, but the sender name. Where can I configure this and how?
Is it possible to have some example?
Thank
Luca
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Ok I use Libiobc instead of unixodbc.
But Now asterisk doesn't find the database...
Olivier
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Olivier
Taylor
Envoyé : dimanche 18 septembre 2005 12:58
À : 'Asterisk Users Mailing List - Non-Commercial
When I receive a voicemail notify via e-mail I would like receive not
the sender phone-number, but the sender name. Where can I configure
this and how? Is it possible to have some example?
Thank
Luca
___
--Bandwidth and Colocation sponsored by
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Sunday, September 18, 2005 8:32 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Voicemail
When I receive a voicemail notify via e-mail I would like receive not
the
Actually, we have thought about that too. The problem is scheduling
so as not to conflict with any of the other shows that our major
sponsors are doing (remember, they help keep the cost of the
conference down).
Within roughly week of AstriCon on one side or ther other we have the
If you are looking for the maximum number of cheap flights from around
the world, and plenty of convention and room space, the answer is Las
Vegas :-)
quoted for truth.
$79 flights from Dallas.
___
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I use a macro to dial most of my extensions and in the macro the
variable ${macro_exten} is used to send calls to the right voicemail
box.
I also use realtime MySQL for SIP, voicemail, and extensions.
When I put a voicemail box in a voicemail context other than default the
method of getting the
4th'ed!
No lack of things to do, cheap hotels (off the strip...or on depending
on your definition of cheap), cheap flights from just about
everywhere...
On 9/18/05, asterisk [EMAIL PROTECTED] wrote:
Viva Las Vegas !!!
Neil
-Original Message-
From: [EMAIL PROTECTED]
Atlanta is hub for Delta and Airtran
Dallas is hub for American
Chicago is hub for ATA
All good central locations with cheap non stop flights.
Atlanta is central for who? With all of the tornados, hurricanes, etc.
I would definately vote no for there. Dallas and Chicago
-Original Message-
From: Steven Sokol [mailto:[EMAIL PROTECTED]
Sent: Saturday, September 17, 2005 10:31 AM
To: Kurth Bemis
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Who is going to AstriCon (The
AsteriskConference)?
We've
Kevin P. Fleming wrote:
Steven Sokol wrote:
Actually, my wife's company holds several events each year in Orlando
and the turn-out is always great -- people love getting a bit of
vacation with their business travel. (An Universal's Islands of
Adventure is just plain awesome!).
If you are
canuck15 wrote:
-Original Message-
From: Steven Sokol [mailto:[EMAIL PROTECTED]
Sent: Saturday, September 17, 2005 10:31 AM
To: Kurth Bemis
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Who is going to AstriCon (The
We have a basic application that runs a SIP channel to pick up a call
and process it. We are using Broadvoice and it's been working great.
We recently rebooted the machine and now when a call comes in Asterisk
picks up the call but does not process it. Asterisk seems to send the
call back to
If you are looking for the maximum number of cheap flights from around
the world, and plenty of convention and room space, the answer is Las
Vegas :-)
quoted for truth.
$79 flights from Dallas.
Having just returned from LV yesterday, the cost of flights is very
inexpensive, but all
On Sun, 18 Sep 2005, Michael Stearne wrote:
Looking for 6092991xxx in from-broadvoice
Reliably Transmitting (no NAT) to 147.135.20.128:5060:
SIP/2.0 404 Not Found
So - go to your Asterisk CLI and type show dialplan from-broadvoice.
Examine the list of extensions shown to figure out why
The best place for Astri Con 2006 would definatly be
Omaha, Nebraska! ;) very central
...ah one could hope.
or Lincoln, better facilities ;)
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Asterisk-Users mailing list
Senad J wrote:
If you are looking for the maximum number of cheap flights from around
the world, and plenty of convention and room space, the answer is Las
Vegas :-)
I would definitively agree!
Yes, but what would one do there?
One who doesn't gamble, drink, or carouse, that is.
I am
Brian Capouch wrote:
Senad J wrote:
If you are looking for the maximum number of cheap flights from around
the world, and plenty of convention and room space, the answer is Las
Vegas :-)
I would definitively agree!
Yes, but what would one do there?
One who doesn't gamble, drink, or
[EMAIL PROTECTED] wrote:
Brian Capouch wrote:
Senad J wrote:
If you are looking for the maximum number of cheap flights from
around the world, and plenty of convention and room space, the
answer is Las Vegas :-)
I would definitively agree!
Yes, but what would one do there?
Paul wrote:
Yes, but what would one do there?
One who doesn't gamble, drink, or carouse, that is.
I am making my first trip to LV later this Fall, and I dread it. I
can't imagine what I'll be able to find to do when I'm not at the
conference.
Rent a car and drive out to Hoover Dam.
Senad J wrote:
[EMAIL PROTECTED] wrote:
Brian Capouch wrote:
Senad J wrote:
If you are looking for the maximum number of cheap flights from
around the world, and plenty of convention and room space, the
answer is Las Vegas :-)
I would definitively agree!
Yes, but what would one do
Hi! I've got two POTS lines coming in to my * box, but I only want the
primary of the two lines available for outbound dialing. I can't quite
figure out how to make that happen. Suggestions?
Thanks,
-Ken D'Ambrosio
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--Bandwidth and Colocation
Hi! I've got two POTS lines coming in to my * box, but I only want the
primary of the two lines available for outbound dialing. I can't quite
figure out how to make that happen. Suggestions?
How about something like this?
exten = _9XXX,1,Dial(Zap/2/${EXTEN})
which dials out on the
=OUT24883xXxXx-20050918-114511-1127058311.261) in new stack
Sep 18 11:45:11 VERBOSE[2223]: -- Executing Goto(Zap/47-1, s|14) in new
stack
Sep 18 11:45:11 VERBOSE[2223]: -- Goto (macro-record-enable,s,14)
Sep 18 11:45:11 DEBUG[2223]: Expression is '0'
Sep 18 11:45:11 VERBOSE[2223]: -- Executing GotoIf(Zap
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Brian Capouch
Sent: Sunday, September 18, 2005 12:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] AstriCon 2006 Location
Senad J
Hi
I have what seems like a similar problem. In the last few days, I stopped
receiving calls from
my Broadvoice number into Asterisk. The account activity page at Broadvoice
does not show the
calls as either missed or incoming. Unfortunately, Broadvoice's line is we
don't support
Asterisk. :-/
How to implement call limitation based on amount of time call per day.
I've implemented the Dial L parameter but in addition I would like to
limit an extension to certain amount of call time per day.
--
#Joseph
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Hello All
any body installed functions as below ?.
i have 5 agent, logged in system.
phone - telco - asterisk server -- queue (5 agents). when all agents busy. i want caller hear message from IVR and when any agent is available. ringback to agent then caller may be connected with that agent
Just installed * 1.0.9 on a FC4 (full
install)
I am using 2 X100P clones
I do not
remember what all steps I took to get everything installed.
Every time
I reboot, I have to modprobe zaptel modprobe wcfxo
before
asterisk will work. Did I miss a
step somewhere?
also,
I have
Xlite
John Novack wrote:
Paul wrote:
Yes, but what would one do there?
One who doesn't gamble, drink, or carouse, that is.
I am making my first trip to LV later this Fall, and I dread it. I
can't imagine what I'll be able to find to do when I'm not at the
conference.
Rent a car and
Hi,
Thanks for the advice on this.The Hicom can be set in Point-to-Point or Point-to-Multipoint mode (amongst others), I assume one is NT and the other is TE mode? If not, Icannot find any specific option to set NT and TE.
Anyway, I have a crossover cable, however, thearticle here:
Hi Simon,
Point-to-Point (P2P) ist set if you want to use DID
(In Germany called "Anlagenanschluss") Point-to-Multipoit (ptmp) is if you want
to use the asteriskwith singelpoint-entry (in Germany called
Mehrgeräteanschluss)(also possible to dial extensions after this, but look
at the
Hi All,
I have installed ChanSpy but its not compiling gives me the following
errorif anyone can help that will be really appreciated...
[EMAIL PROTECTED] asterisk-1.0.9]# contrib/scripts/astxs -install -autoload
apps/app_chanspy.c
make: *** No rule to make target `apps_env'. Stop.
-c
Is there any method to make difference between Hidden (Private) and
unknown (Out of area) incoming calls on ZAP/x101p? I want to block any
hidden call, and to allow unknow calls, but ZAP channel (X101P) always
delivering empty CALLERID= in both cases.
Hey Everybody,
Well, I knew better.
I Competely broke my faxing in the current CVS (09/18/2005) of Asterisk.
Needed to do a clean install to see if that would fix my caller id
problem and thought that I had backed up all directories (Missed the
modules directory).
Modified the make file,
Je serai absent(e) du 16/09/2005 au 01/10/2005.
Je suis actuellement absent mais vous pouvez adresser vos mails à
[EMAIL PROTECTED] qui vous répondra.___
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Asterisk-Users mailing list
Just installed * 1.0.9 on a FC4 (full install)
I am using 2 X100P clones
I do not remember what all steps I took to get everything installed.
Every time I reboot, I have to modprobe zaptel modprobe wcfxo
before asterisk will work. Did I miss a step somewhere?
Yes, the step
Why Asterisk showing (on SCCP and H323 phones) different CID related to
type of Incoming channel:
If incoming channel is SIP, on phone is displayed CALLERIDNUM
If incoming channel is ZAP, on phone is displayes CALLERIDNAME
It vas very frustrating! I lost couple hours of my time to find that my
On 9/19/05, Goran Dj. [EMAIL PROTECTED] wrote:
Why Asterisk showing (on SCCP and H323 phones) different CID related to
type of Incoming channel:
If incoming channel is SIP, on phone is displayed CALLERIDNUM
If incoming channel is ZAP, on phone is displayes CALLERIDNAME
It vas very
While we are on the subject.. how do you modify the TXT message that
gets send to the 'pager'... Is that hard coded.. or can that be
changed?No variables I change in voicemail.conf seem to change the
from address, etc.
On 9/18/05, Damon Estep [EMAIL PROTECTED] wrote:
-Original
While we are on the subject.. how do you modify the TXT message that
gets send to the 'pager'... Is that hard coded.. or can that be
changed?No variables I change in voicemail.conf seem to change the
from address, etc.
Not sure this answers your question, but in voicemail.conf, I have:
;
No, no, it's more SIMPLE than that. Try this:
[incoming]
exten = s,1,setcallerid(NAMENUMBER)
exten = s,2,dial()
That's all (after couple of hours of investigation).
If call origin from SIP, i see NUMBER on my phone, if call origin from
PSTN, i see NAME on my phone.
- Original Message
Heh,
Why don't we do a tele-conference :)
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Reid
Forrest
Sent: Sunday, September 18, 2005 2:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] AstriCon
I think the best place would be Louisville, Kentucky.
Its drivable from Cincinnati, Indianapolis, Tennesee, and a short
plane ride from Chicago, Atlanta, NYC. Best of all, its near me :)
On 9/18/05, Rich Adamson [EMAIL PROTECTED] wrote:
The best place for Astri Con 2006 would definatly be
Download cvs head and look at
/usr/src/asterisk/configs/voicemail.conf.sample
All of the variables for email, page, etc are listed in the sample
files, it is more comprehensive than many of the other samples.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL
Hello All,
I am using monitor with soxmix, however the quality seems somewhat low
after sox converts to mp3.
Does anyone know a way to get a higher quality file? Some of my lines
are coming in on isdn.
Regards,
Greg
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Hi, all
Did anyone tried TDM400 card on old main board (Intel 440LX chipset -- PII)?
The reason I am asking is because TDM400 needs PCI2.2 and main board is PCI2.1
I do not want to upgrade yet.
Thanks,
Rudolf
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Some of you may remember back in May the thread on using Asterisk as a
voicemail server for a Cisco Callmanager system.
My own Callmanager system is integrated into an Asterisk server for
voicemail (and other things). Back in May I was using H323 for
integration, but since I've upgraded to CCM
I am trying to configure E1 card (Digium) but not able to do that. The
green light doesn't come up when it starts.
What can be the problem. I have also changed the jumper settings of the
card from T1 to E1 but still no relief.
Thanks in advance
Manish
Hi,
First of all make sure you have inserted zaptel and wxfx* modules
and
Then execute ztcfg -vvv
It will show you the channels configured
and see its green signal now. :)
Regards
Gurminder
On 9/19/05, manish kumar [EMAIL PROTECTED] wrote:
I am trying to configure E1 card (Digium)
Hello,
I'm going to be starting implementing something like this in ASTPP. I
don't know of any very easy way to do it at present though. We did
implement a solution similar to this using ASTCC. If you're interested
email me off list.
Darren Wiebe
[EMAIL PROTECTED]
Joseph wrote:
How to
Some of you may remember back in May the thread on using Asterisk as a
voicemail server for a Cisco Callmanager system.
My own Callmanager system is integrated into an Asterisk server for
voicemail (and other things). Back in May I was using H323 for
integration, but since I've upgraded to
This is not much help but it is more of a suggestion than I've seen on
this list on this issue. Would you be able to try an older release of
asterisk say 1.00 or right around there. I really do believe it was
working back then but I've not played with that exact issue for a long time.
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