Hi all,
From the web, I can find a table scheme of sipusers for ARA using.
However, I can't find any meaning of each field, especially for the
field regserver which is new in the table. Can any tell me more
detail about the usage of each field?
CREATE TABLE `sip_buddies` (
`id` int(11) NOT
Hi all,
In version 1.2, there is a realtime function and it is very easy to
use with prefix.
exten = s,1,RealTime(table|name|peter|user_)
and we can easy get back the value as user_name, user_id, etc.
However, I found the the function will be depreciated in 1.4. There
is a replacement using
PROTECTED] wrote:
On 8/13/07, Rilawich Ango [EMAIL PROTECTED] wrote:
However, I found the the function will be depreciated in 1.4. There
is a replacement using application REALTIME. I found that it is very
troublesome to use it.
I'm afraid I don't have any advice with regard to your problem
Hi all,
How can I update a field as null in table using realtime function?
i.e. : Set(REALTIME(tbl,name,1234,prop)=null)
Above will update the field as a null word in table but I want to
update it as null. Can we do that using realtime fuction?
Ango
Hi all,
In default, we can use # to transfer the call. I want to know how I
can know the user presse # to transfer the call in dial plan.
ango
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Hi all,
I am looking for a prepaid application. I found that there are many
applications in the page
http://www.voip-info.org/wiki/view/Asterisk+Prepaid+Applications.
Anyone recommendation among them?
ango
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I am using an asterisk to call another asteisk (i.e
Dial([EMAIL PROTECTED]) in asteriskA). After that, the following error
message displayed and asterisk crashes at once. Anyone has such
experience and can help to fix it?
asterisk version: 1.4.11
zaptel 1.4.5.1
using RealTime
From the web site said: 3-way Calling: Normally implemented by the
phone. Can I do it in asterisk? How?
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What do you mean? I just want to know whether there is a way to do
the following.
1. A --calls -- B
2. A on hold, B --calls -- C
3. A, B and C connected to talk
On 9/28/07, Paul Hales [EMAIL PROTECTED] wrote:
How are you going to do it without a phone?
PaulH
That's easy if phone supports 3 ways call. However, phones in my
company only have 1 line without join function. Is it possible to
implement 3 ways call using Asterisk without phone support in my case?
On 9/28/07, Anthony Francis [EMAIL PROTECTED] wrote:
Rilawich Ango wrote:
What do you mean
In CDR, I found that there are 3 records after doing call transfer.
However, 3 of them are individual record that is very difficult to
identify they are related to call transfer. My question is how to
identify the call with a clear flow, from CDR or by other means, is a
call transfer.
Do u mean meetme? It is total different from my case.
In meetme, everybody need to know and dial the conference room number
to get into the conference room. In my case, party A,B,C may not know
the conference number. A only knows B numbers and B only knows C
numbers.
On 9/28/07, Pamela Weis
there is a variable ${BLINDTRANSFER} that will fill in a
value in blind transfer. However, I can't find any variable that will
fill in a attended transfer. Anyone can advise?
On 9/28/07, Alex Balashov [EMAIL PROTECTED] wrote:
On Fri, 28 Sep 2007, Rilawich Ango wrote:
In CDR, I found
Hi,
I have a question about the combine key sequence in feature.conf.
Say, I have a featuremap for atxfer.
atxfer = *1
So I press *1 to enable atxfer. I want to know how can I adjust the
timeout second between * and 1. I found they need to be pressed
within 0.5 second to make it work. Can I
HI all,
I got segfault in the system log that make asterisk crash. I still
have no idea what cause this segfault. Is it a bug? Anyone has
experience about it?
phsip01 kernel: asterisk[3412]: segfault at 2aabd10f2b40 rip
0037e806ea75 rsp 41d3cc70 error 6
version:
Hi all,
I want to have a 16 FXO in a PC. Is it possible to use 4 x TDM404
or 2 TDM808 to get 16 FXO? What is the difference (in performance and
control) in using 4 x TDM404 and 2 x TDM808 if possible?
ango
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What do you mean by interruption? Is it possible to better control to
prevent it? The options you provided is over my budget. That's why
I am looking for multiple TDM cards.
On 10/22/07, Gergo Csibra [EMAIL PROTECTED] wrote:
Monday, October 22, 2007, 9:47:49 AM, Rilawich wrote:
Hi all,
you can do it using iptables, port forwarding.
On 10/29/07, Abdul [EMAIL PROTECTED] wrote:
Hi,
Is it possible to have multi listening bindport in asterisk?
Now days mostly ISPs are Blocking the standard 5060 port so we want to keep
option if 5060 is blocked we can ask our customers to use
Hi all,
I got the following message in the log yesterday. After that, no
more in/out bound call can be made. What is the meaning of the
message? ango
[Mar 12 09:26:17] ERROR[29565] chan_sip.c: We could NOT get the
channel lock for SIP/2367-d8062fb0!
[Mar 12 09:26:17] ERROR[29565] chan_sip.c:
]
[mailto:[EMAIL PROTECTED] On Behalf Of Rilawich Ango
Sent: Wednesday, March 12, 2008 6:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] asterisk out of service
Hi all,
I got the following message in the log yesterday. After that, no more
HI all,
I have set up a queue with 2 members (A B). 1st call is waiting
in the queue and a queue member A is ringing but don't take the call.
Member A keeps ringing. Then 2nd call is also get into the queue but
I found that queue member B doesn't ring. That's mean member B is
available to
to no
;to keep backward compatibility with the old behavior.
;
autofill = yes
On Thu, Apr 10, 2008 at 8:57 PM, Don Pobanz
[EMAIL PROTECTED] wrote:
Rilawich Ango Thursday, April 10, 2008 3:28 AM
I have set up a queue with 2 members (A B). 1st call is waiting
in the queue and a queue
Do you mean autofill works in 1.4.x? But it doesn't work even I set it.
On Fri, Apr 11, 2008 at 11:07 AM, BJ Weschke [EMAIL PROTECTED] wrote:
Rilawich Ango wrote:
Thanks. I have checked that the queue.conf. I keep the default
setting as autofill=yes in my tests. That's mean even
Anyone can update me about the queue sticking by a caller? Is it
solved in version 1.4.x? How?
On Sat, Apr 12, 2008 at 9:42 AM, Rilawich Ango [EMAIL PROTECTED] wrote:
Do you mean autofill works in 1.4.x? But it doesn't work even I set it.
On Fri, Apr 11, 2008 at 11:07 AM, BJ Weschke
) UNSIGNED DEFAULT 1;
For following this issue, see http://bugs.digium.com/view.php?id=12445
Regards,
Atis
On Sat, Apr 12, 2008 at 4:42 AM, Rilawich Ango [EMAIL PROTECTED] wrote:
Do you mean autofill works in 1.4.x? But it doesn't work even I set it.
On Fri, Apr 11, 2008
Hi all,
In my understanding, we can use mssql as a database of asterisk
thro' unixodbc. And we can easy using mysql (realtime) to do the
same. Now, I want to keep 2 connections, one is mysql and one is
mssql. Because both database have information that needed to be read
from asterisk. Can I
Hi all,
Recently, I experienced one way audio after call transfer.
incalling call (PSTN) A -- GXP2000 thro' zap --blind transfer-- destination B
Finally A and B reach each others, but there is only one way audio.
Anyone get the same experience before? How to solve the problem?
Asterisk
Do you mean the problem is solved using asterisk 1.4.18? Are you
using the setting as mine?
Below is my setting. One way audio is a result after A B connected.
PSTN (A)--1200P-- Asterisk -- GXP2000 --blind transfer -- Extension B
You can see that involve many parties in the blind transfer
Segmentation fault occurs after executing the following cmd.
Dial(SIP/[EMAIL PROTECTED]|35|Ttr)
Is it a bug and how to fix it?
Below is the core dump message converted by gdb.
#0 0x068be1ad in realtime_peer (newpeername=0x1b37844 10.20.0.1, sin=0x0)
at chan_sip.c:2547
#1 0x068becb3 in
HI,
Does asterisk will ignore the setting in files if realtime is
applied, say asterisk will ignore all the setting in sip.conf if
realtime table sip_buddies is applied?
ango
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But why take chances anyway? Move all the relevant
conf files from /etc/asterisk to some other place to
be safe.
cheers
- Ben.
--- Rilawich Ango [EMAIL PROTECTED] wrote:
HI,
Does asterisk will ignore the setting in files if
realtime is
applied, say asterisk will ignore all
Hi,
A makes call to B. B has connection problem with the server (say, the
lan cable is unplugged).
1: A --- server
2: A --- server
3: server B
In 2, server will send the ring to A and it will hear ringing tone.
In 3, server will try to connect B until timeout.
My question is:
A will still
I have a queue with the following setting.
total queue member =30, autofill=1, timeout=25, monitor_format=wav49
asterisk 1.4.18
In busy hour, the loading of CPU reaches over 300%. At that moment,
all members are occupied and many calls are waiting in the queue.
There will be choppy and line cut
Hi all,
There is a setting called autopause in queue.conf to pause a queue
member if they fail to answer a call.
The autopause setting will pause the agent even when they are on the
line. I want to know if it is possible to pause the queue member only
when they don't answer after timeout?
ango
Hi,
I want to send some text to the phone such that the phone can
display the text on its display. I have tried to use SendText but it
doesn't work. Does the phone need to support when asterisk issues the
SendText application?
ango
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HI,
I got a one way audio when an ip phone dial to another ip phone in
the same network. What I find is TCP UDP run different legs. Below
is my configuration.
asterisk (192.168.1.10)
ipphone-A (192.168.1.111)
ipphone-B (192.168.1.101)
router (192.168.1.1) external IP (116.48.138.83)
When A
Hi all,
I would like to know how can I immunize the background noise in my
case. Anyone can help? I have adjusted txgain rxgain in different
value but the result is the same.
ango
Below is my configuration.
asterisk1.4.21.1
zaptel1.4.11
addon1.4.7
TDM400 (FXOx4)
There is a very large
I have a realtime queue and the state of the queue member change as
below. Not-in-use (no call)- Unknown (ringing)- Not-in-use
(answered). The state shown in show queues does not really reflect
the state of the phone. I have searched the net and also the
UPGRADE.TXT by the warning message
Hi all,
I have the following queue and members. I found that there is a
call stuck in the queue so other call can't enter the queue. I want
to know whether we can remove the call (by CLI) to free the queue.
ango
2700 has 1 calls (max unlimited) in 'rrmemory' strategy (35s
holdtime),
hangup Local.
Andy
On 8/27/08, Rilawich Ango [EMAIL PROTECTED] wrote:
Hi all,
I have the following queue and members. I found that there is a
call stuck in the queue so other call can't enter the queue. I want
to know whether we can remove the call (by CLI) to free the queue.
ango
:
Try CLI soft hangup Local.
On 28 Aug 2008, at 09:01, Rilawich Ango wrote:
Hi ,
Actually, there are 3 queues in the server. Only one queue (2700)
has problem. I want to reset or remove the caller only in 2700
without affecting other queues or calls. Does it work for my case?
On Thu
Yup I just copy and paste to it but it shown not a known channel.
On Thu, Aug 28, 2008 at 6:47 PM, Steven Howes [EMAIL PROTECTED] wrote:
Did you tab complete it to make sure it was right?
On 28 Aug 2008, at 11:39, Rilawich Ango wrote:
I got the message below after I issue the soft
HI all,
I have a queue say 5000 and there are 10 member in the queue. When
there is a call to the queue, the members will ring according to the
defined strategy. In day end, I have to create a report about the
queue and its member. But I found that it is very difficult to find
the relation
Where can I get the meaning of each field in queue_log?
On 4/15/07, Darryl Dunkin [EMAIL PROTECTED] wrote:
You will probably find what you are looking for here:
/var/log/asterisk/queue_log
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rilawich
Ango
hi,
I have 2 asterisks with the following configuration.
asterisk server 1 (S1) has an user 9002
asterisk server 2 (S2) has an user 9003
Both users can make call to each other without problem.
Now I add both users to both servers, i.e.
asterisk server 1 (S1) has users 9002,9003
asterisk server 2
[EMAIL PROTECTED] wrote:
ango,
can you provide some sip.conf and extens.conf info?
daveC
Rilawich Ango wrote:
hi,
I have 2 asterisks with the following configuration.
asterisk server 1 (S1) has an user 9002
asterisk server 2 (S2) has an user 9003
Both users can make call to each other without
Hi,
Recently, I got the following error from the system and it caused
the asterisk down.
Apr 25 13:48:22 WARNING[27460] rtp.c: Unable to allocate socket: Too
many open files
Apr 25 13:48:22 WARNING[27460] acl.c: Cannot create socket
The output of ulimit is unlimited.
I have searched about the
-info.org/wiki/index.php?page=file%20descriptors
Justin
On 4/25/07, Rilawich Ango [EMAIL PROTECTED] wrote:
Hi,
Recently, I got the following error from the system and it caused
the asterisk down.
Apr 25 13:48:22 WARNING[27460] rtp.c: Unable to allocate socket: Too
many open files
Apr 25 13:48:22
Hi all,
I have 2 cards, they are x100p and TDM400p (2 FXO and 2 FXS), in a
server. Is it possible to control the call pass through those cards?
Any example for me to reference?
ango
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Thanks for your reply.
What I ready do is:
add ulimit -n 65535 in safe_asterisk
increase value to 203380 in /proc/sys/fs/file-max
Both actions don't help much for the file descriptor growing.
What I want to know is:
Do I need to reboot if I insert the following in /etc/security?
*
How about if both ServerA and ServerB houses extensions 500 throught
699. Such that users can dynamically register Server A or Server B.
Can we use DUNDi to implement such network?
On 5/9/07, Alex Robar [EMAIL PROTECTED] wrote:
Hi Ronaldo,
Yes, you can use DUNDi for this. DUNDi simply
It is quite interesting and I am looking for it. Could you give me
some more information or website how to set it up?
On 5/10/07, Remco Post [EMAIL PROTECTED] wrote:
Rilawich Ango wrote:
How about if both ServerA and ServerB houses extensions 500 throught
699. Such that users can
HI all,
Recently, I got the following message from CLI and finally the
asterisk will hang. Anyone can tell me how to fix the problem or why
it will happen.
Thanks.
Jun 17 14:18:02 DEBUG[24573] channel.c: Avoiding initial deadlock for
'SIP/1127-008d65f0'
Jun 17 14:22:45 ERROR[24696]:
1.2.10
On 6/19/07, Doug [EMAIL PROTECTED] wrote:
At 02:08 6/17/2007, Rilawich Ango wrote:
HI all,
Recently, I got the following message from CLI and finally the
asterisk will hang. Anyone can tell me how to fix the problem or why
it will happen.
Thanks.
Version?
Also
Recently, I got the following error messages in CLI periodically.
Jan 20 17:43:18 ERROR[8641]: chan_sip.c:11002
handle_request_subscribe: Got SUBSCRIBE for extension
[EMAIL PROTECTED] from 192.168.0.123, but there is no hint for that
extension
I have no idea what the error message tell me. I
Hi,
In the system, there are realtime and realtimeupdate to access data
in realtime model. Does it include realtimeinsert and realtimedelete
such that they can be used to manipulate the database more completely?
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Hi all,
I am using asterisk1.2.14,realtime and I find there is a strange
case in the receiver's display. I have a dial plan to route a call
to the destination. I haven't set the callerid(num) for the caller.
In the receive ends, it's display shows asterisk when I make a call
to the receiver.
Hi all,
I found the following error in CLI. Anyone can tell me what is the
meaning of this error? Is it related to the codec problem? I only
allow g729 and gsm in the system. Most of the client use g729 to
connect to the server. In location A, the clients can make call
without problem.
Hi all,
Does any can give me some example to setup call parking and call
transfer of a call?
In my understanding, call parking and call transfer should be like
something below. Am I right?
ango
Call parking:
caller A - callee B
callee B park her call
callee B get back her call in another
Noah,
Thanks for you reply. I have a problem in call parking as following.
scenario 1
1.Caller A - callee B
2.Callee B answered
3.callee B dial # to park the call and hear transfer
4.callee B dial 700 to park the call
5.callee B hang up and caller A hear 701
Why caller A hear the call parked
Hi, I have tried the regex function below with MACRO_EXTEN=5000*.
However, both of them return 0 instead 1 to me. How can I search the
character in the end of line?
${REGEX([*]$ ${MACRO_EXTEN})
returns 0
${REGEX(*$ ${MACRO_EXTEN})
returns 0 with error
ango
Hi all,
I have read many forums and discussion groups talking about fax
support in asterisk. Some of them conclude that asterisk doesn't
support fax. However, some of them conclude that there is no
relationship between fax and asterisk as asterisk will only pass the
fax signal to the fax
Hi,
We can use RealTime to query the database with one criteria. How
about if I want to query with 2 or more criteria?
ango
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To UNSUBSCRIBE or update options
Hi all,
I want to build a web page for user to input a phone number. Then,
the number will input to asterisk and it will makes call. At that
moment, asterisk will make another call to a internal ext. Finally
asterisk will bridge 2 calls together for conversion.
Does asterisk can do it?
you.
Regards,
Amit Mehta
Cell: +91 9898340962
On Tue, Jan 6, 2009 at 11:41 AM, Rilawich Ango maillist...@gmail.com wrote:
Hi all,
I want to build a web page for user to input a phone number. Then,
the number will input to asterisk and it will makes call. At that
moment, asterisk
Hi,
I wonder how I can relate the CDR records for the case of call
transfer. I can't find their relationship in CDR. Any can advice?
ango
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To
the call.
Hope this might help you.
Regards,
Amit Mehta
Cell: +91 9898340962
On Tue, Jan 6, 2009 at 11:41 AM, Rilawich Ango maillist...@gmail.com
wrote:
Hi all,
I want to build a web page for user to input a phone number. Then,
the number will input to asterisk and it will makes call
I also experience that problem. Is it a bug?
On Wed, Feb 4, 2009 at 5:53 AM, Mark Michelson mmichel...@digium.com wrote:
Remco Barendse wrote:
1.4.23.1 is quite badly broken and there are no significant new
features
There are no new features at all, actually. What problems are you
Hi all,
Is it possible to install more than 1 asterisk in a single server?
If yes, what do I need to set and take care?
Rgds,
ango
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It seems better to install once with multiple instances. Do we need
to take care the port or IP of each instance?
On Wed, Feb 25, 2009 at 5:36 AM, Klaus Darilion
klaus.mailingli...@pernau.at wrote:
Klaus Darilion wrote:
Rilawich Ango wrote:
Hi all,
Is it possible to install more than 1
to
/usr/local/sbin/safe_asterisk2
Cheers
Geraint
You will also need to look at asterisk.conf in the new installation
directory and as a quickfix to get it running, use a different location for
astrundir
2009/2/24 Rilawich Ango maillist...@gmail.com
- Show quoted text -
Hi all
Hi all,
I enabled recording (mixmonitor) in queue and process started after
queue member pick the call. But recording will stop after picking up
by another extensions of call transfer/parking in the same call. Is
it possible to continue to record the call for call parking/transfer,
how?
Rgds,
Hi all,
I found that a new field lastms is used in 1.4.24. What is the
usage of that field and the datatype of it?
ango
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Tilghman,
Thanks. Can you elaborate the usage about it? What is the meaning
of each valid value in this field?
ango
On Mon, Mar 23, 2009 at 11:24 AM, Tilghman Lesher
tilgh...@mail.jeffandtilghman.com wrote:
On Sunday 22 March 2009 21:40:14 Rilawich Ango wrote:
Hi all,
I found
HI,
We are experiencing the hum noise when the conversion of 2 parties is
established. How can we eliminate that noise? ango
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My configuration is simple as below.
SIP phone - asterisk - CISCO - T1
Do you mean the hum noise is created by electric-magnetic field?
Asterisk can do nothing to eliminate it?
On Sun, Mar 29, 2009 at 2:47 AM, Steve Edwards
asterisk@sedwards.com wrote:
On Sat, 28 Mar 2009, Rilawich Ango
The CLI shows zap is necessary for conference recording. Can I enable
conference recording if using ztdummy or dahdi, how? ango
-- Executing [...@owt_meetme:4] MeetMe(SIP/3601-c80b4520,
5599|rcixMP) in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe
HI,
Recently, I found that asterisk fail to get the correct context of
the sip phone. Below is the configuration and the log message. In
the log message, asterisk fail to identify the calling party. As a
result, it use a default context instead of int. Anyone know why and
how to fix it?
6, 2009 at 5:00 AM, Rilawich Ango maillist...@gmail.com wrote:
HI,
Recently, I found that asterisk fail to get the correct context of
the sip phone. Below is the configuration and the log message. In
the log message, asterisk fail to identify the calling party. As a
result, it use
Hi all,
I wonder who has the same voice quality problem as what we have.
Below is our configuration.
Company --- asterisk 1.4.22 (g729) --- CISCO --- T1 --- customer
Sometimes, customers told me that they heard our voice not very clear,
like a call from far far away. I heard the recording is
Normally, there are 10 concurrent calls in peak. You are right that
usage g729 is due to bandwidth consideration.
On Thu, Apr 23, 2009 at 2:42 PM, Gordon Henderson
gordon+aster...@drogon.net wrote:
On Thu, 23 Apr 2009, Rilawich Ango wrote:
Hi all,
I wonder who has the same voice quality
Hi,
Feature originate can be used make call thro' the web. There is a
parameter ,Async, in it. I set it to true but there is no effect.
Actually, I want to do the following.
What I know the function originate is:
originate call --- party A
party A rings
party A answers call
party B rings, party
when they answer?
Wouldn't it just be better to play a message after party a answers and then
start ringing party b so that party a knows what's going on?
2009/4/24 Rilawich Ango maillist...@gmail.com
Hi,
Feature originate can be used make call thro' the web. There is a
parameter ,Async
Hi,
I follow the
web,http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf
- mohstream.sh , to configure music on hold to play using mms but
failed. Anyone can play using mms?
ango
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Rilawich Ango wrote:
Hi,
I follow the
web,http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf
- mohstream.sh , to configure music on hold to play using mms but
failed. Anyone can play using mms?
ango
Hi,
After following the messages to install flite, I can find the following files.
/usr/lib/asterisk/modules/app_flite.so
/etc/asterisk/flite.conf
That's mean flite is installed successfully. Then I restart asterisk
but nothing found for that module.
sip*CLI core show application flite
Your
Can you try to disable call waiting in your phone?
On Fri, May 15, 2009 at 6:44 AM, sean darcy seandar...@gmail.com wrote:
sean darcy wrote:
I have two internal analogue extensions off a TDM400P. If the first is
busy, I'd like to ring the second. So:
[incoming]
exten =s,1,Answer()
exten
HI,
I want to allow user to press 0 to the voicemail if the user don't
want to wait in the queue. Below is what I set but it doesn't work.
Anyone can help? ango
file: features.conf
[applicationmap]
opervm = 0,self/both,Macro,opervm
file: extensions.conf
...
exten =
Thanks all. I figure out to exit the queue by setting context in queue.conf.
On Thu, May 21, 2009 at 11:20 PM, Kevin P. Fleming kpflem...@digium.com wrote:
Mark Michelson wrote:
Not to undermine Kevin's requests to read what is documented, I can say that
what you want actually will not be
Hi all,
I download asterisk-addon 1.6.1 but the VoIP phone failed to
register to the system with the message below.
[May 26 15:45:11] WARNING[29665]: res_config_mysql.c:317
realtime_mysql: MySQL RealTime: Invalid database specified: asterisk
[May 26 15:45:11] WARNING[29665]:
, Tilghman Lesher
tilgh...@mail.jeffandtilghman.com wrote:
On Tuesday 26 May 2009 02:52:18 Rilawich Ango wrote:
Hi all,
I download asterisk-addon 1.6.1 but the VoIP phone failed to
register to the system with the message below.
[May 26 15:45:11] WARNING[29665]: res_config_mysql.c:317
realtime_mysql
Hi,
I use realtime and I found that changing accountcode needed to
restart asterisk to activate that code and shown in CDR. Does it has
a way to update accountcode without restart asterisk?
ango
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I am using 1.4.24 with realtime.
On Fri, May 29, 2009 at 5:21 PM, Rilawich Ango maillist...@gmail.com wrote:
Hi,
I use realtime and I found that changing accountcode needed to
restart asterisk to activate that code and shown in CDR. Does it has
a way to update accountcode without restart
Thanks. I wonder do I need to reload it if I am using
realtime/database? I have to change the accountcode during the call
so it is not possible to do it if reload is needed.
On Fri, May 29, 2009 at 9:35 PM, Tarek Sawah tareksa...@hotmail.com wrote:
accountcode is a setting you add to your SIP
Hi all,
Any good recommendation of IP phone in term of sound quality and
price (reasonable) using with asterisk?
ango
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Hi,
How can I get the codec using of the current call in dial plan? Is
it possible to do it?
Thanks.
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As I know, the voicemail will be sent using localhost smtp. I want to
use another smtp server for sending voicemail to the users. Is it
possible to set it, where to set it?
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asterisk-users
Hi all,
I noticed that when an user register asterisk, asterisk will update
the corresponding record in db user table. However, if the user
register failed, maybe wrong password. There is no record in
database. How can I log those register records, including successful
and failed login to
Hi all,
I get the following message in the CLI after enabling video
function. I have searched about the codec 126 but nothing found.
Anybody can tell me how to fix the problem?
Nov 30 15:54:27 NOTICE[16508]: rtp.c:576 ast_rtp_read: Unknown RTP
codec 126 received
Nov 30 15:54:27 NOTICE[16508]:
Hi all,
How can I redirect the CLI output to file without viewing it on
screen? Is it possible.
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