*Hi pal.*
*
*
*Maybe you forgot to enclose the near parenthesis found? 'Goto(7124,1' in
extension file, Firstly correct this syntax then try again.*
*
*
*BRs*
*/***
*Nguyen Trung (Mr)*
*Trust me I'm an Engineer!*
* *
* VEGA CORPORATIONhttp://vega.com.vn/Default.aspx?AppID
Hi All,
Currently i'm facing with a cdr issue, When i originate a call (outbound
call) to uncorrect/unregistration user, asterisk inform me that call was
failed but in mysl-cdr (cdr-csv also) records.
Here are 2 samples
the
call, A would get C's voicemail, as expected.
Thanks,
Hai Nguyen.
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Dear All
I've setup in lab a model include
a *handse*t (nokia 6021 in supported
listhttp://www.voip-info.org/wiki/view/chan_mobile)
connect to an *bluetooth dongle* (Cambridge Silicon Radio, Ltd Bluetooth
Dongle (HCI mode)) attached to an *PC *(install Asterisk 1.6.2.19 and Bluez
3.7 with
= s,n,Wait(2) ;; THIS
exten = s,n,SendDTMF(1) ;; AND THIS ARE NEEDED
exten = s,n,Background(tnttspWelcome)
exten = s,n,Background(CurrentAnnouncement)
exten = s,n,Goto(0,1)
-- Vinh
On Tue, Oct 26, 2010 at 7:07 PM, Vinh Nguyen vinhdi...@gmail.com wrote:
Can anyone reproduce this with their google
Dear all,
First off, I am very new to asterisk so forgive me if any of my
comments or questions seem trivial. Thanks to [this
post](http://blog.polybeacon.com/2010/10/17/asterisk-1-8-and-google-voice/)
and [this post](http://www.davidvossel.com/?p=28), I have GV set up on
asterisk through
Hello,
i have Cisco Unified Communications Manager with 10 ip phone,i dont buy
license IVR of Cisco Unified Communications Manager. Can i use feature IVR
on Asterisk connect with Cisco Unified Communications Manager.
Sorry my English.Thank you.
--
Wrong context for incoming,
you can read
http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf
and
http://www.voip-info.org/wiki/view/IPKall
2009/12/25 --[ UxBoD ]-- ux...@splatnix.net
- Qurba Joog qurbaj...@gmail.com wrote:
| Hello,
|
| Please forgive me if I'm repeating this
You only need to install ztdummy.
It's usually straightforward if you have Linux kernel 2.6.
-quan
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pezhman Lali
Sent: September-11-08 5:59 AM
To: asterisk
Subject: [asterisk-users] meetme without zaptel
Dear,
I have some limitations
PROTECTED] wrote:
On Monday 08 September 2008 12:43:53 Nguyen wrote:
Hi listers,
We want to implement one call center with asterisk. The idea is it should
be scalable, with openser as an dispatcher and bunch of asterisk servers
to
do ACD, Queues, Agents things... Easy to say :(
Look closely
G'day listers,
Well, I just found this one:
http://svn.digium.com/view/asterisk/team/russell/events/doc/distributed_devstate.txt?view=markup
Time to read.
On Tue, Sep 9, 2008 at 2:29 PM, Nguyen [EMAIL PROTECTED] wrote:
Dear Tilghman,
Thanks for your feedback. Scratching my head to see how
to
start add clustering capabilities to asterisk?
Your replies are much appreciated,
--
With best regards,
Nguyen
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Hi,
I've tried the following toy dialplan:
[sipcalls]
exten = _X.,1,NoOp()
exten = _X.,n,Dial(SIP/${EXTEN},,g)
exten = _X.,n,Playback(good-bye)
exten = _X.,n,Hangup()
With the above dialplan, when the callee hangs up, Asterisk does play
good-bye to the caller. However, when the caller hangs
Hello,
I tried to configure a very simple case of Asterisk using SIP
userA --- Asterisk server userB
sip.conf
[userA]
type=friend
username=userA
host=dynamic
nat=no
context=test
[userB]
type=friend
username=userB
host=dynamic
nat=no
context=test
In extensions.conf
[test]
exten =
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Nguyen
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on protocol selection on both sides (Asterisk and HP). Can you help to point out, what protocol I should set on HP3750 and Asterisk?
Sorry List for long posting.Thanks and best regards,Nguyen-- With best regards,
Nguyen
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,Nguyen
On 5/26/06, Josué Conti [EMAIL PROTECTED] wrote:
Hi I work with equipment Siemens Hipath 3000 and HiPath 4000, I can help you, I do not have manuals technician to send, but if to want can help. Already I established connection asterisk(
1.0.9) with Hipath 3750 with a TE110P and a TMS2
a bit?Thank you and best
regards,Nguyen
On 5/26/06, Josué
Conti [EMAIL PROTECTED]
wrote:
Hi
I work with equipment Siemens Hipath 3000 and HiPath 4000, I can help you, I
do not have manuals technician to send, but if to want can help. Already I
established
Hi,Oh, I just want to get it to work. Caller Name is something luxurious for us .NguyenOn 6/13/06, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
Hi,As long for HiPath 4000 callerID name doesn't work, only number -Original Message- From: [EMAIL PROTECTED]
[mailto:asterisk-users- [EMAIL
%. The equipment says between sim.The asterisk uses HiPath 3750, for access the PSTN and when a linking is for a telephone of asterisk, the Hipath directs the digits for asterisk.
I wait to have helped.
Greetings
Josué
2006/5/25, Benchev [EMAIL PROTECTED]:
Hi Nguyen ,I haven't got the opportunity
Hello All I read in www.sineapps.com have Asterisk 2.0 rewritten C# and run on windows, any body could be mail or send to me URL to download.ThanksTin Trung NguyenTechnical TeamMobile: 084-91.365.4857website: www.daivietcontrol.net___
Hi all,
I have an TE11 card and I installed the zaptel driver from digium.
The zaptel.conf look like:
span=1,1,0,esf,b8sz,yellow
bchan=1-23
dchan=24
when I tried modprobe -v wcte11xp without any error message
and then ztttol
I received the error Red alarm
What would be the problem?
Thanks in
Hi,
I need to develop a project in which the user can phone a number, say
something and the voice will be output to a speaker, if the user want to
select other actions, he could just press a number on the keypad, e.g.:
press 1.
I did it with the following:
1. make a incoming context, looks like:
Hello All
Anybody had used ooH323 for asterisk
i using ooH323-0.7.2 and asterisk CVS may 2005. OpenH323 1.17.1 and pwlib 1.9.0 and GNUGK 2.0.2
audio is very good, better than SIP and IAX, but i have problem.
how to router call from openh323 to outside PSTN.
my h323.conf setting
; Objective
Hello All
any body have used SS7 run with asterisk. could you like tell me how to download driver of SS7 and how to use it.
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Hi All
I have problem with LIBMFCR2 for once Exchange
I using Sangoma card, the firstly. my ssystem run successful with MFCR2, connected to E10 (Acatel Exchange), after that, i move connection connect to EWSD (Siemens), my system don't work. error protocol R2.
my system:
Asterisk CVS 1.1.X
Hello All.
I'm using sangoma card A-101. tested successful with E10 (ACATEL) Exchange,connection with E1, CAS, (using unicall-0.0.3pre4).
my systemrun success,incoming call and call out are good.
when iswitch to EWSD (SIEMENS) R-15. my asterisk faill, cannot connect with EWSD.
(E10 and EWSD
Hello All.
I'm using sangoma card A-101. tested successful with E10 (ACATEL) Exchange, connection with E1, CAS, (using unicall-0.0.3pre4).
my system run success, incoming call and call out are good.
when i switch to EWSD (SIEMENS) R-15 . my asterisk faill, cannot connect with EWSD.
(E10 and EWSD
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Hello
I have problem with transfer call if using ACD
When i using ACD with agent and queue setting, i cannot monitor call and transfer call. this's my setting
- i have 2 IAX phone (phone number as 201, 202), agent.conf
agent = 1001,4321,member 1agent = 1002,4321,member 2agent = 1003,4321,Tin
then,
Hello
how to calculator billing exactly when IAX accept the call, my configure
customer -- telco --- asterisk -- ACD -- IAX
at time, for example: 11:00 i dial to asterisk
11:01 asterisk answer channel and dial to IAX phone (11:02)
ring 20 second (at 11:22).
when IAX answer call (11:22) and talk 10
Hello All
I want to setting my asterisk with features as belows:
- When dial to system, caller hear music and message "welcome to asterisk Open PBX".
- then when extension (IAX, SIP) is available, ringing on channel.
- when channel answer call (offhook), at the time, i want to Host Exchange
Hello All
i need to transfer CDR data from linux to MS SQL Serever (on Windows). writing by Perl. I have download and install UnixODBC, DBI, DBD from CPAN,
when i tested isql -DSN -UID - PWD, that's successful, but when run by perl, message alert
could not loaded driver database,
anybody
Hello ALL
SS7 for asterisk release http://www.footnotess7.com/. but i not yet account to download.
any body have SS7. could you like send to me.
thanks
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Hello ALl
i need context to do:
record to wave file and receive DTMF when recording wave file.
for example:
exten = s,1,Record(test:wav)
exten = s,2,hangup
when recording, press # to hangup and i want to receive others DTMF (while recording), max DTMF to received as 7 and when received enough 7
Hello All
Any body have NVDialDetect module using for dialing out. (www.newmantelecom.com).
How to solve problem, when dial out, i want to remote call answer, then asterisk play wave file. current, dial out, play wave file when remote call not yet answer.
any advice ?
Thanks
Hello ALL
When i dial out to outside, how to detected remote call offhook.
i append in extensions.conf
[ext-callout];exten = s,1,NVLineDetect(60,d);exten = s,1,NVLineDetect;exten = s,1,NVBackgroundDetect(custom/aa_1);exten = s,1,MachineDetect(7000,2,2200);exten = s,1,NVFaxDetect(10)
exten =
Hello
Any body was tested LIBISUP. and price of LIBISUP packet ?. how much to purchased it from digium.
if posible, tell me where are LIBISUP beta release to test with asterisk and my postoffice of my country (vietnam).
Best regards.___
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Hello All
I'm using TxFAX and rxFax. this's work when my system connected with PABX. then when i connect my card with PSTN, my system don't work. it's don't send and receive any thing fax document.
Thanks
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Hello All
how to install functions allow called record current call by pressed any key to wave file. for examples.
the caller call to asterisk, then press 123 tone to switch to saler person. then two persons conversation, the saler want to save current call. saler may be press any key as *5 or
Hello All
How to detect remote called offhook.
i make a context as below
i created call file. copy to /var/spool/asterisk/outgoing.
Channel: vpb/g0/9050718MaxRetries: 1WaitTime: 10Context: ext-calloutExtension: sPriority: 1
then when i copy to /var/spool/asterisk/outgoing. the asterisk auto call
Hi All
I wan to get DTMF while voicemail recording sounds. DTMF received save to contents field of mail attach with wave sounds file.
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Hello All
i have 2 problems, please help me
1. How to implenment record call at called side.
i want to record the call by called press the DTMF key.
2. how to implement call out functions, for example: i create .call file and copy to /spool/outging, then when asterisk call out, i want that: when
Hello All
i have big problem for unicall.
my system work successful with sangoma card, E1 and CAS signalling (vietnam).
when at the some time. i have trouble then my system is half (CPU instructions = 100)
i tested for some case as belows:
- When i dial, then my system became answer, the caller
Hello All
I'm settup my asterisk as belows:
sangoma card, connected with E1, CAS Signalling.
I have two problem.
1. The asterisk don't received any DTMF when caller input to
2. when i dial to system, the caller hear bad sounds. monitor on console. asterisk show error.
Jun 11 12:15:45
, it is
possible to do so with asterisk.
Regards,
El jue, 09-06-2005 a las 00:48, Phuong Nguyen escribió:
1. Play a low background music when the user record his/her voice
You Want a Karaoke? lol
Regards,
--
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[EMAIL PROTECTED]
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Hi All
i used cangoma card, connected with E1, using unicall. asterisk 1.1.x. when i dial to asterisk server. asterisk show error as belows:
-- Unicall/9 extension '9' in context 'from-pstn' from '71811242' does not exist. RejectingcallJun 10 16:47:59 WARNING[28159]: chan_unicall.c:2655
Hi all,
Does Asterisk support multi thread? I mean:
Is it possible to do one of the 2 following scenarios:
1. Play a low background music when the user record his/her voice
2. If the first scenario is not possible, can we play two music stream at
the same time? i.e: using MP3Player to play a
Hello
I'm using H323 channel and client used ohPhone-1.4.1 (with gatekeeper). when at client side dial to asterisk server (dial , test mode). ohPhone don't hear any thing sounds (no audio). i dial between ohphone (with gatekeeper). sounds are good.
my current setting. Asterisk-1.1.x, GNUGK
Hi all,
I could use MP3Player to play local sound (e.g: /usr/sound/abc.mp3) but I
could not use it to run a remote stream, if I use mpg123 in command line, I
can hear the audio ( /usr/bin/mpg123 http://...), but the same remote mp3
file could not be replay with asterisk.
I would appreciate with
Hello All
How to measure the time when i click hold button on softphone. i need to save the time when user choice hold call
for example: when user answer the call, the time is: 13:37:35 PM, after that, user choice hold call (time: 13:37:37 PM), then release hold call button (13:18:00 PM).
the
Hello All
I'm using asterisk 1.1.X and MFCR2 lib version 0.03pre2. when i call to E1 (connected with asterisk), chan_unicall don't detected event incoming call and show error.
error messages:*CLI Warning, flexibel rate not heavily tested!Rx CAS bits 0x9 [ 1/ 0/ 0]Line unblocked -- R2 Channel
Hi All
i'm using sangoma card. connected to E1,
my wanpipe file as
## WANPIPE1 Configuration File### Date: Fri May 27 00:25:04 GMT+7 2005## Note: This file was generated automatically# by
Hello All
I need to use Asterisk with an E1 sangoma card with CAS R2 signalling for Vietnam
what is difference between libr2 of CVS and libmfc2 of soft-switch.org ?
how to compile chan_unicall.c on asterisk. asterisk update CVS-head- May 27 2005.
Hello all.
How to compile chan_unicall.c
i have problem when compile chan_unicall.c, error message
please help
gcc -c -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS
Hello All
Any body used sangoma card A101. I have problem with this card. My system are:
Linux Redhat 8.0. asterisk 1.07 and libpri, zaptel,...
i connected E1 and MF/R2 signalling.
i configure HDLC and TDM Voice. this is my configure as belows:
##
Hello All !
i'm purchased sangoma card A-101. i connect to E1 with MF/R2 signalling. but card don't work. negotiation with E1 fail.
please help me to correct it. i dont' know some parameters such as:
MTU, BAUDRATE
Thanks
Tin Trung
___
Hello all
i'm build success H322 in channel/H323 of asterisk. but don't know how to use it.
i run GNUGK on server and client using ohphone. when i dial to asterisk server. the connection accept and disconnect.
please help me to configure in H323.conf and extensions.conf.
Can anyone tell me how to reset the IAXy? I used I put
it the wrong ip config in the IAXy and it conflicts
with my network whenever I plug it in. Currently the
DHCP is disable. I need to re-enable it to change the
settings.
The hard reset button on the IAXy doesn't seem to work
: [Asterisk-Users] CNAM for Asterisk
Kevin Nguyen wrote:
Thank you for your help.
Kevin N.
I already replied to your first message with a great deal of
information; did you not receive it?
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I am working for Accudata Technologies. We
provide CNAM via http request or raw TCP/IP connection.
We would like to provide the same capability to Asterisk. I installed
Asterisk on Fedora 2.0
and did reading about AGI and AGI application at
I am working
for Accudata Technologies. Weprovide CNAM via http request or raw
TCP/IP connection.We would like to provide the same capability to
Asterisk. I installedAsterisk on Fedora 2.0and did reading about
AGI and AGI application
Dear all,
I just update my asterisk from v 1.0 to lasted CVS, I also used module
res_config_mysql in asterisk-addons. Every working fine, but i got
problem with IAX user can't make a call, when IAX user make a call, i
got message in console like this
Feb 21 06:20:07 NOTICE[365]: chan_iax2.c:6090
Hi all,
I am trying to setup the SMS of Asterisk. I have a Siemens SMS enable
fixed phone which connects to my Asterisk through PSTN. Currently, I
can use my fixed phone to edit and send messages to the Asterisk.
However, I cannot make my Asterisk to send messages to the fixed phone.
The SMS
Hi all,
I am trying to setup the SMS of Asterisk. I have a Siemens SMS enable
fixed phone which connects to my Asterisk through PSTN. Currently, I
can use my fixed phone to edit and send messages to the Asterisk.
However, I cannot make my Asterisk to send messages to the fixed phone.
The SMS
Hi all,
I am trying to setup the SMS of Asterisk. I have a Siemens SMS enable
fixed phone which connects to my Asterisk through PSTN. Currently, I
can use my fixed phone to edit and send messages to the Asterisk.
However, I cannot make my Asterisk to send messages to the fixed phone.
The SMS
Hello
I am using a TDM400P-4FXO to connect my Asterisk to
telephone line. However, this TDM400P uses RJ45 connection while our telephone
standard uses RJ11. How can I wire the cable for the connection?
Thanks
Hoa
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I implemented successfully with guidance from this document
http://www.voip-info.org/wiki-Asterisk+-+dual+servers
However, I had to make a small change to the sip.conf sample files:
From:
exten = _1XXX,1,Dial(IAX/asterisk:[EMAIL PROTECTED]/[EMAIL PROTECTED])
and
exten =
. But this is not
EG convenience, since i need to use g279 with another endpoint (working
EG ok).
EG Why this negotiation problem happens?
Try to add to cisco peer (not shown in your mail)
[cisco]
disallow=all
allow=alaw
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Nguyenmailto:[EMAIL
, but after looking at the Makefile, I am not sure
where I can get the source code for G723? Can you provide a hint?
TIA
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http
for example) won't
work.
My question is , why not take just buf[0]? why translate? my UA
always send something like d= (one digit) at a time.
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Nguyen mailto:[EMAIL PROTECTED]
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[EMAIL
regards,
Nguyenmailto:[EMAIL PROTECTED]
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are suitable for
production environments?
My needs is simple: standard switch (Call transfer, Parking...),
Voicemail, Voice menu (Autoattendand).
If someone have running Asrerisk on production system, please share the
yours version info.
best regards,
Nguyen
At 03:49 PM 8/12/2003 -0500, you
Hi listers,
is my question about stability of * a wrong question to ask here?
Nguyen
Date: Wed, 13 Aug 2003 10:39:43 +0700
To: [EMAIL PROTECTED]
From: Nguyen Nam [EMAIL PROTECTED]
Subject: Stable versions of Asterisk (Was: Re: Fair comparison (John Todd))
Hi,
It's really a problem for new Asterisk
recognized.
But compiling under Linux give me following error. The error come from
generating the dependency files.
All the Makefile(s) was leave untouched. I was not sure what's the problem.
Can some one help to point me to right way?
TIA
best regards,
Nguyen
[EMAIL PROTECTED] asterisk]# make
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