I just read:
Certain options to the Dial() statement require that Asterisk is in the
media path, and consequently Asterisk will not let go of it: /t/, ''T,
h, H, w, W or L (with multiple arguments). Probably there are
more.
I had in my memory that r, R, m would also prevent a reinvite. Can
in,
bye
Ronald Wiplinger
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I have some DIDs from NuFone (tollfree).
How can I switch them and to which provider? What is the cost for that?
What is the procedure for that?
bye
Ronald Wiplinger
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and try to transfer from
callee and called phone with #1601# to transfer a call to 601, but I
just hear at both phones the keys I press, but nothing happens. Same
with *2601#
What do I miss here?
bye
Ronald Wiplinger
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Ronald Wiplinger wrote:
I have not used astcc with pin codes so far, since I set-up the phone
number as card number.
Some of my users want now to dial in to the system and than use their
card, which is their phone number.
For that I would need a way of authentication, like a pin.
I want
. That avoids later update
problems!!!
ln -s astcc.agi astcc-disa.agi
2. instead of
if ($config{'pinstatus'} eq YES) {
(please help me to write it, but what I want is:)
if (my called name is 'astcc-disa' ) {
bye
Ronald Wiplinger
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to me.
Any ideas? Any starting points? (other than google or wiki) Has
somebody experience with such things?
bye
Ronald Wiplinger
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, the code
includes a #, I guess I should change blindxfer = ##, right?
2. transfer the call to the conference room: *2789 (conference room
789) and key in the conference room's password
3. all other participants log into the conference room (789)
Any thoughts, improvements for that ?
bye
Ronald
parallel, but of course it assumes still that only one card is in use.
bye
Ronald Wiplinger
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in 111
4. waiting for the prompt
5. say the users name: Peter
6. call Peter's number:2345678
Has anybody done something like that (partially) before?
bye
Ronald Wiplinger
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I am looking for a way of not to install a softphone, preferable as a
link on a web site to a webphone on MY SITE !!!
Has anybody an idea for that? AJAX?
bye
Ronald Wiplinger
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words several times, but I could not figure out which
program to use you are referring. Maybe you could try it simple with
http://.x.
Thanks!
bye
Ronald Wiplinger
--
Tom
On 4/26/06, *Jim Houser* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
I need the same exact
Bruce Reeves wrote:
The only one I have heard of is WebIAX
http://www.voip-info.org/wiki/view/WebIAX
This one looks ok, with the exception that it is only working with
Internet Explorer!!!
bye
Ronald Wiplinger
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company announcement is.
Do you know such numbers?
bye
Ronald Wiplinger
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What are you using as FXO ports for a few analog (remote) lines?
What is the price, where to buy, what is your experience?
bye
Ronald Wiplinger
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?
bye
Ronald Wiplinger
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and
in the dundi.conf?
bye
Ronald Wiplinger
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and
extension.conf snippets.
Thanks for your assistance in this matter.
~ron
Could you configure it? Can you tell me more about it?
bye
Ronald Wiplinger
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I tried to find this in asterisk wiki, but each link I found was broken.
How can I use my Snom 190 or 360 softphone as Intercom ?
bye
Ronald Wiplinger
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with this (probably
simple) problem for almost a whole week.
What is exactly your dial command?
bye
Ronald Wiplinger
Thanks for any help.
Ronald Wiplinger wrote:
Tiago Stein D`Agostini wrote:
Hi,
Ie been looking for some time how to use asterisk to initiate SIP
connections between 2 IP
depending on the bandwidth?
Thanks for thinking with me ;-)
bye
Ronald Wiplinger
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for different users)
To dial to a phone we use something like:
exten = 8,1,Dial(SIP/6001,20,tr)
Can we use something like:
exten = 8,4,DeadAGI(astcc.agi,${EXTEN},6001,${TARIFF},3)
and use TARIFF with all location as x cents ?
bye
Ronald Wiplinger
kevin ling wrote:
Hi,
Check the vm_general.inc file
Where should this file be?
bye
Ronald Wiplinger
Kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ronald
Wiplinger
Sent: Sunday, April 16, 2006 12:24 PM
To: Asterisk Users Mailing List
Ronald Wiplinger wrote:
How can I change this:
Asterisk PBX [EMAIL PROTECTED]
to:
London PBX [EMAIL PROTECTED] ??
I tried several settings in voicemail.conf, without success!
I found it!
I had to change the info in the /etc/passwd file
bye
Ronald Wiplinger
begin:vcard
fn:Ronald
be in vm_general.inc
I also would like to know, how you came to the idea that in
vm_general.inc could be a solution.
I cannot resist, maybe you could take a volunteer job in the church
or in the next fishing club ...
bye
Ronald Wiplinger
Steve Totaro wrote:
Ronald Wiplinger wrote:
Thanks for posting it back to the list
No problem, not sure why you would think I would like to correspond
with you directly. I am into the community thing. Why send me a
direct email with some crappy process to become a verified sender just
How can I change this:
Asterisk PBX [EMAIL PROTECTED]
to:
London PBX [EMAIL PROTECTED] ??
I tried several settings in voicemail.conf, without success!
bye
Ronald Wiplinger
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-Users mailing list
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--
Ronald Wiplinger (CEO of ELMIT)
http://www.elmit.com http://voip.elmit.com http://e-paper.elmit.com
Tel. (M) +886.939.775.516 (O) +886.2.2835.7765 (ENUM) or FWD 511208
I want to upgrade * this weekend.
What can I prepare? What will I have to change in the settings? Where
can I read about it?
I use now:
*CLI show version
Asterisk SVN-trunk-r8447M built by root @ on a x86_64 running Linux
on 2006-01-25 15:33:01 UTC
bye
Ronald
If one of my users call with its xlite client out via voipdiscount, I
see in the CLI following message:
Comfort noise support incomplete in Asterisk (RFC 3389). Please turn
off on client if possible.
How can I turn this off?
He also complain that than he hear nothing at all.
bye
Ronald
I have two phones (111 and 112) on a LAN, and I have on a users site a
phone 333.
phone 111 uses sip.conf, while 112 uses real-time set-up.
111 can call 333 AND the audio is working
112 can call 333 but audio is just white noise.
333 can call 111 or 112 and audio is working.
The phones are
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--
Ronald Wiplinger (CEO of ELMIT)
http://www.elmit.com http://voip.elmit.com http://e-paper.elmit.com
Tel. (M
I am using eyebeam and I am happy with it. However, it is boring just to
talk to my son in the other room.
Whenever I try to convince somebody to buy eyebeam, they are scared of
the price.
Is there a free video soft phone available, that will work with eyebeam
/ asterisk?
bye
Ronald
no
explanations of how to do it.
Does anyone care to give a pointer to any explanation about how to do it?
canreinvite=yes
and look at the options for dial()
Thanks in advance
bye
Ronald Wiplinger
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the phone. Just plug it in.
It's all done via http and php (yes, even grandstream).
Dan,
is there a way to configure softphones that way too, like x-lite?
bye
Ronald Wiplinger
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Ronald Wiplinger
begin:vcard
fn:Ronald Wiplinger
n:Wiplinger;Ronald
org:ELMIT Co., Ltd.
adr:Shilin District;;5F., No.8, Alley 2, Lane 92, Dexing W. Road;Taipei;;11158;Taiwan
email;internet:[EMAIL PROTECTED]
title:CEO
tel;work:+886.2.2835.7765
tel;cell:+886.939.775.516
x-mozilla-html:TRUE
url:http
Rudolf Ladyzhenskii wrote:
What is the SIP server you specified?
Rudolf
On 4/10/06, Ronald Wiplinger [EMAIL PROTECTED] wrote:
I am still trying to figure out how to overcome this problem.
I use for International calls a, for USA calls b, ...
Most of the time I get: Forbidden
, br_OA_dvoice
I gone away from broadvoice, since they admitted to have troubles and I
had still to pay for NO phone call !!! (multiple lines)
bye
Ronald Wiplinger
type=peer
dynamic=yes
username=XX
fromuser=XX
authname=XX
user=phone
secret=SECRET
host
${EXTEN:[EMAIL PROTECTED])
;exten = _9011Z.,103,Dial(SIP/011${EXTEN:[EMAIL PROTECTED])
exten = _9011Z.,104,NoOp(${DIALSTATUS})
by hand I move the remark sign around!!!
How are you handling such situations?
bye
Ronald Wiplinger
begin:vcard
fn:Ronald Wiplinger
n:Wiplinger;Ronald
org:ELMIT Co., Ltd
internal LAN, without any audio
problem.
SOME of my phones of the LAN can call phones on the Interenet, which are
also on the a NAT. Usually the phone rings, but no audio.
SOME phones, however, CAN call without any audio problems.
Where and how to look at this problem?
bye
Ronald Wiplinger
])
;exten = _9011Z.,103,Dial(SIP/011${EXTEN:[EMAIL PROTECTED])
exten = _9011Z.,104,NoOp(${DIALSTATUS})
by hand I move the remark sign around!!!
How are you handling such situations?
bye
Ronald Wiplinger
I first try to dial provider-a, as you have, then do a gotoif on
${DIALSTATUS}, where
!)
bye
Ronald Wiplinger
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Kevin P. Fleming wrote:
Ronald Wiplinger wrote:
It does not go to the next provider. Is there a settings for timeout
to go to the next provider???
Uhh... yeah. That is why there is a timeout parameter for the Dial()
application.
___
After I posted
JP Carballo wrote:
Ronald Wiplinger wrote:
I tried now many places to put these lines in. The system still
announces This card number is in use.
Can you give me a place where to put it in?
It's not receiving a card number.
Find the following 3 lines:
#
# At this point we have a valid card
, ) in new stack
[Apr 5 09:22:37] == Spawn extension (default, h, 1) exited non-zero
on 'SIP/601-5039'
exten = _91Z.,103,Dial(SIP/00${EXTEN:[EMAIL PROTECTED])
exten = _91Z.,104,NoOp(Line 104 ${DIALSTATUS})
bye
Ronald Wiplinger
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JP Carballo wrote:
Ronald Wiplinger wrote:
Insert this in astcc.agi; anywhere after the calls for it to load
and connect to the db.
if ($phoneno eq RESET_INUSE) {
setinuse($carddata-{number}, 0);
exit(0);
}
Thanks!
I use it here:
elsif ($phoneno eq BALANCE) {
setinuse
JP Carballo wrote:
Ronald Wiplinger wrote:
Insert this in astcc.agi; anywhere after the calls for it to load
and connect to the db.
I tried now many places to put these lines in. The system still
announces This card number is in use.
Can you give me a place where to put it in?
bye
, it is the first time I hear positive about Sphinx.
Do you have a menu for the installation you did?
bye
Ronald Wiplinger
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JP Carballo wrote:
Ronald Wiplinger wrote:
I have some troubles with ASTCC. TO often the in-use flag
remains set.
I would like to find a solution, where astcc.agi checks automatically
if THIS user is in a call rather than to check the flag.
If that is not possible, I would like
guess what you have in your extensions.conf,
sip.conf, but above was a good start, wasn't it?)
bye
Ronald Wiplinger
begin:vcard
fn:Ronald Wiplinger
n:Wiplinger;Ronald
org:ELMIT Co., Ltd.
adr:Shilin District;;5F., No.8, Alley 2, Lane 92, Dexing W. Road;Taipei;;11158;Taiwan
email;internet
a copy
You are kidding, are you? You do not really expect that, do you?
bye
Ronald Wiplinger
of the data from the police, eventually, that way.
Until then, however,
we're out of luck.
I've had a couple of offers of hosting (I put it on
voip-info.org) but
for the moment, I've signed up
phones try to
register all the time, .
bye
Ronald Wiplinger
begin:vcard
fn:Ronald Wiplinger
n:Wiplinger;Ronald
org:ELMIT Co., Ltd.
adr:Shilin District;;5F., No.8, Alley 2, Lane 92, Dexing W. Road;Taipei;;11158;Taiwan
email;internet:[EMAIL PROTECTED]
title:CEO
tel;work:+886.2.2835.7765
tel;cell
We are using several providers and I would like to know if and how many
concurrent calls you can place for a voipstunt account?
Has anybody tried multiple concurrent calls?
bye
Ronald Wiplinger
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not find one. I compiled stun and would like to run it. The docs says
you need two IP addresses. I could alias a second IP address to the
public Ethernet port I have on one machine.
What IP address should than be in the dns? The primary or the
secondary
bye
Ronald Wiplinger
in use (call)
How can I get this info as CLI comand and as a jump criteria in the
dialplan
bye
Ronald Wiplinger
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hang up, reset the flag!
The in-use flag remains set, if the caller hang up before the gateway
gets the call.
bye
Ronald Wiplinger
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I tried the example I found:
exten = 123, 1, Answer
exten = 123, 2, SendText(hello world)
exten = 123, 3, HangUp
However there was nothing on the display!
Any hints?
bye
Ronald Wiplinger
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Jerry Jones wrote:
Show channels?
Yes, on Linux
bye
Ronald Wiplinger
On Mar 31, 2006, at 2:09 AM, Ronald Wiplinger wrote:
In the past I used SetGroup and CheckGroup to figure out if my
allowed providers lines are all used or not.
Since most of my provider have given me a single
as well, without trying to call.
bye
Ronald Wiplinger
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, given
DHCP IP address, phone number and the password.
Nat traversaldisabled (but I tried also enabled and stun with
nat addr stun01.sipphone.com)
Any ideas?
bye
Ronald Wiplinger
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I have an ZAP extension number 222 which is connected instead to a phone
to a FXS/FXO converter and from there to a CDMA gateway.
To dial my mobile phone I use:
222 (wait 2 seconds) 09123456789
I cannot figure out how to write this into the dialplan as a default number!
222 as above I will
I have an ZAP extension number 222 which is connected instead to a phone
to a FXS/FXO converter and from there to a CDMA gateway.
To dial my mobile phone I use:
222 (wait 2 seconds) 09123456789
I cannot figure out how to write this into the dialplan as a default number!
222 as above I will
two possibilities:
1. on demand. Dial another extension number after the call, what
executes a system command
2. automatically. Add in the dialplan the system command after hanging up.
(just to start somewhere)
bye
Ronald Wiplinger
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and restarted asterisk
What do I miss?
bye
Ronald Wiplinger
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Florian Overkamp wrote:
Hi Ronald,
Ronald Wiplinger wrote:
voipbuster/ 194.221.62.201 5060
UNREACHABLE
voipstunt/x 194.120.0.200 5060
a reload shows than:
voipbuster/ 80.239.235.200 5060
/x 194.120.0.200 5060 UNREACHABLE
bye
Ronald Wiplinger
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Olle E Johansson wrote:
Chris A. Icide wrote:
Ronald Wiplinger wrote:
snip
601, 602, 605, 606, 608, 609, 610, 615 and 616 are in sip.conf
621 and 626 are in Real-time sip_buddies
621 and 626 changes username back from name to number (name) in the
database, and never shows it in sip show
[EMAIL PROTECTED] wrote:
I'm still trying to learn some parts of Asterisk, so sorry in advance for the
dumb question!
How do I set up an extension to dial out to the PSTN through my ZAP interfaces?
I want the ability to have a ring group that will ring all of the phones in an
office and
Jolly M. Recto wrote:
Hi,
i have diffirent provider example(3 single account in deltathree, 4
account in packet8 and so on) . How this possible to make the three
individual sip account in deltathree act as trunk so that i cannot get
a busy call. If line one fail goto line 2 then line 3 or
1. I want to call somebody and, as soon (and not before) a playback
should be played. How can I do that?
2. How can I accept dtmf tones with such calls?
Example:
System calls all staff and ask them a question. The staff will answer
with a digit!
The playback should start when the staff picks
Ronald Wiplinger
Best regards
jan
Sharon wrote:
hello,
can someone help me with ser redirect to asterisk.
any help appreciated.
Thanks,
AA
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the record as:
*CLI sip show peers
Name/username HostDyn Nat ACL Port Status
621/621192.168.250.76 D N 5060 OK (65 ms)
4. I search for this record again and found it is set: name=621 and
username=621
Bug or feature?
bye
Ronald
account
Phone: 610
Username: 610
User Pwd:
However, that does also not work.
Can anybody give me a hint, how it could work?
bye
Ronald Wiplinger
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615 changed username Ronald office to 615, although no change in sip.conf
Did anybody else experienced that?
*CLI show version
Asterisk SVN-trunk-r8447M built by root @ vpbx on a x86_64 running Linux
on 2006-01-25 15:33:01 UTC
bye
Ronald Wiplinger
Nabeel Jafferali wrote:
I got some troubles with my wifi phone.
What phone is this?
pulver phone
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still be used as a good reference.
bye
Ronald Wiplinger
All in all not a very astute purchase. I should know; I've had 5 of them.
I use the UTStarcom F1000 currently. Much better but still not good.
Mark
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
Ronald Wiplinger wrote:
Nabeel
, they cannot call.
You also would have a nice statistic for each call,
bye
Ronald Wiplinger
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; office 602 Ronald
PHONE_603=ZAP/1r1; living room 603 cordless
For you this should work too:
exten = 2020,2,Dial(SIP/2005IAX/2010,25,tr)
bye
Ronald Wiplinger
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Pete Barnwell wrote:
On Sat, 2006-01-28 at 13:13 +0800, Ronald Wiplinger wrote:
How can I make it more readable?
Name/username
601/601
123456789/123456789
voipbuster/abcd
601 = hotline
123456789 = Peter Pan
only voipbuster/abcd is easy read/understandable!
I'm not entirely sure
Pete Barnwell wrote:
On Sat, 2006-01-28 at 22:32 +0800, Ronald Wiplinger wrote:
[snip]
asterisk*CLI sip show peers
Name/username HostDyn Nat ACL Port Status
PeteB/peteb(Unspecified)D 0UNKNOWN
The username is the username
Bartosz Piec wrote:
Ronald Wiplinger wrote:
exten = 600,1,Dial(${PHONE_LOCAL},60,tr)
Type this:
exten = 600,1,Dial(${PHONE_LOCAL},60,tTwWr)
dial at 600 and see if this helps. If so, change all commands in that
way (tT is for transfer, wW is for recording).
You must also have sox
How can I make it more readable?
Name/username
601/601
123456789/123456789
voipbuster/abcd
601 = hotline
123456789 = Peter Pan
only voipbuster/abcd is easy read/understandable!
bye
Ronald Wiplinger
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(!!) accounts at voipbuster, each one for a different user!
How do I handle that optimized?
bye
Ronald Wiplinger
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Bartosz Piec wrote:
Ronald Wiplinger wrote:
I tried to transfer a call, pickupcall and onetouch recording, but
have not got it to work!
You must uncomment the lines in feature.conf (remove the ; character
from the beggining).
[featuremap]
blindxfer = #1; Blind transfer
for that application. Does anybody remember where it was?
5. I want to record a conversation in a conference room, how to do?
6. I want to record missing prompts, I saw somewhere a simple
application to record 100 prompts easy, ... where was it? Number was 666
bye
Ronald Wiplinger
for recording has no effect!
All phones have in the dial command t as option.
What do I miss?
bye
Ronald Wiplinger
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connection.
Has anybody done already some parts of that?
bye
Ronald Wiplinger
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I am looking for a way to signup users and provide them with a file
which includes all settings, just to put somewhere.
Does something like that exist?
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To
[Jan 23 19:56:44] -- Got SIP response 300 Multiple choice back
from 194.120.0.201
[Jan 23 19:56:44] -- Now forwarding SIP/601-fc4d to
'Local/194.120.0.211:5060,sip:194.221.62.211:5060,sip:80.239.235.211:[EMAIL PROTECTED]'
(thanks to SIP/voipstunt-5c8c)
[Jan 23 19:56:44] NOTICE[3439]:
Technical Support wrote:
Check out www.generationd.com for their fax2mail and mail2fax scripts. It
might make life simpler
There is no description how to set-up!
The scripts are working with asterisk, but how?
bye
Ronald Wiplinger
-Original Message-
From: [EMAIL PROTECTED
I had:
exten = 695,2,SayDigits(${CALLERIDNUM}) ; Says your phone number
but it does not work anymore after upgrade. How should it be now?
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I cannot see it
make[1]: Leaving directory `/usr/local/src/svn-versions/asterisk/pbx'
/bin/sh: curl-config: command not found
make[1]: Entering directory `/usr/local/src/svn-versions/asterisk/apps'
Makefile:103: *** missing separator. Stop.
make[1]: Leaving directory
Ronald Wiplinger wrote:
I cannot see it
Found it!!! Tab and spaces are hard to see,
make[1]: Leaving directory `/usr/local/src/svn-versions/asterisk/pbx'
/bin/sh: curl-config: command not found
make[1]: Entering directory `/usr/local/src/svn-versions/asterisk/apps'
Makefile:103
How should be the macro rewritten?
[macro-faxreceive]
exten = s,1,Set(FAXFILE=/var/spool/asterisk-fax/${UNIQUEID}.tif)
exten = s,2,DBGet(EMAILADDR=extensionemail/${MACRO_EXTEN})
exten = s,3,rxfax(${FAXFILE})
exten = s,103,Set([EMAIL PROTECTED])
exten = s,104,Goto(3)
...
[Jan 23 10:43:38]
How to set the callerid? I had prior 1.2:
exten = _91NXXNXX,3,NoOp(SetCallerID(${username}))
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(SetCallerID(${username}))
exten =
_91NXXNXX,4,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN:${TRUNKMSD}},${TARIFF})
bye
Ronald Wiplinger
On 1/22/06, Ronald Wiplinger [EMAIL PROTECTED] wrote:
How to set the callerid? I had prior 1.2:
exten = _91NXXNXX,3,NoOp(SetCallerID(${username
I used:
cvs checkout zaptel libpri asterisk asterisk-addons asterisk-sounds
iaxyprov astcc
and in the same order I try to compile it.
Asterisk ends with the lines below. It complains of a newer libpri, but
I just did it a step before!
What do I miss?
chan_zap.c:62:2: #error You need newer
Dave Cotton wrote:
On Sun, 2006-01-22 at 09:14 +0800, Ronald Wiplinger wrote:
I used:
cvs checkout zaptel libpri asterisk asterisk-addons asterisk-sounds
iaxyprov astcc
and in the same order I try to compile it.
Asterisk ends with the lines below. It complains of a newer libpri, but
I
1. At the end of compiling asterisk I got a lot of warnings. How can I
solve that?
I used:
# svn checkout http://svn.digium.com/svn/zaptel/trunk zaptel
# svn checkout http://svn.digium.com/svn/libpri/trunk libpri
# svn checkout http://svn.digium.com/svn/asterisk /trunk asterisk
# svn checkout
Is there a solution for the problem that the card in use flag is set,
after the user hang up?
The flag remains set, if the user hang up, after the price for the call
will be announced.
It is bad (for the business), because this happens most of the time only
for NEW users!
Solutions?
1. Do
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