Hi matt
I use Asterisk release 1.6.1. So if it's my version which is source of
problem could you suggest an application like web-meetme to manage asterisk
conferences ?
Regards
Harry
2009/9/3 Matt Riddell li...@venturevoip.com
On 4/09/09 3:24 AM, harry R wrote:
Hi everybody
I have
Hi everybody
I have a problem and want to know if anyone has already seen it before :
I try to use web-meetme.3.1.0 and follow these instructions
http://sourceforge.net/docman/display_doc.php?docid=48924group_id=164788
1) when i do make command in cbmysql folder, errors happened
Hi
I have a question about application order when a do a call to a terminal
Do I need to use application in this order ?
Ringing()
Answer()
Dial()
or in this order
Ringing()
Dial()
Answer()
Suddenly I have some doubt. Personally, I usually do it like the first one.
Regards.
Harry
If you're using Asterisk to bridge an incoming call to a device (eg. a
SIP phone), then you just need
Dial(...
No need to Answer (as that then starts to cost the caller money if calling
via the PSTN or some PSTN to VoIP bridge to get the call into your Asterik
box in the first place),
No idea. I've never had a use for it...
Some unscrupulous (IMO) operators (in the UK) will answer a call then
present another ringing tone to the caller while their equipment then
places them in a call queue or forwards the call on, etc. They do this as
they're terminating a revenue
Yeah I know that flag, it's 'r' option but If i remember, that's usefull
only if your terminal don't generate ring tone.
I don't think that answer() is necessary with that option.
Harry
I just test ringing() application
it's just play to calling party a ring tone even if the called party
Hi
I want to create a feature to on/off a kind of followme option.
Is it possible ?
I success to use default feature blindxfer and atxfer defined in
features.conf in my dialplan by writting something like this :
exten = 111,1,Set(DYNAMIC_FEATURES=atxferblindxfer)
exten = 111,n,Dial(SIP/111,,Tt)
Hope this helps. Again, bear in mind that we are new to this so if
someone suggests a better way, they are probably right :-) - John
Thank you John for this example.
I'll try to implement it and give you a backup if I have any questions or
suggests.
Harry.
Read the UPGRADE.txt
Solution is to use functions instead:
Set(CALLERID(name));
Set(CALLERID(num));
Set(CHANNEL(language));
etc
Thanks again for solution
Harry.
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Hi
A few day ago, I notice that some applications missed in asterisk 1.6.1
release even if *.so file which normally create them were compiled during
Asterisk install.
SetCallerID(), SetCIDNum(), SetCIDName(), SetLanguage() ... and maybe so
more.
anyone already notice that to ?
If it's not
I have Asterisk 1.6.1.4 and GUI 2.0 (Latest). I never had such problem (my
main Asterisk server is at 1.6.0.6 with latest GUI as well).
I would reinstall the GUI. But I can tell you it *SHOULD* work.
Oh! While I'm writing this mail. I just looked at one of my client's
Asterisk: 1.6.1.1
Hi
Someone may give me an example of followme() application using in a dialplan
(including what to configure in followme.conf) ?
I use asterisk 1.6.1 so if your example can match to that release it's will
be wonderfull.
Thank in advance.
Harry.
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Hi
Has anyone already use asterisk 1.6.1.1 with asterisk GUI last release ?
I'm trying it but I have this problem :
Just after I logged, I have system status main page but no other links where
I can click to go to other pages (remember left panel!)
Regards
Harry
2009/8/21 Ishfaq Malik i...@pack-net.co.uk
I have to disagree with you there, we use 1.4.17 and sip prune realtime
works fine
After a few test, I notice these events when I use
asterisk+mysql+realtime+sip
1) after a sip prune realtime peername, peername will not be reachable
by another
Have you set the qualify column in the sip table?
yes and default set to yes
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.
regards.
Harry.
2009/8/19 Steve Howes st...@geekinter.net
On 19 Aug 2009, at 16:37, harry R wrote:
mysql -uasterisk -pasterisk asteriskdb
When I do that in a linux terminal it works.
But I always have this err 2002.
I'd try and tcpdump it if you can find a way. Might be something odd
Hi
I have some question about mysql realtime.
1) Anyone know exactly if there is a specific order to declare sip table
column for realtime ? In which file can I find that order ?
2) In my extconfig.conf, [settings] are :
sipusers = mysql,general,siptable
sippeers = mysql,general,siptable
so
When I reload chan_sip.so, it seems that connected terminals are no
longer
detected by Asterisk because when I tape CLI command sip show peers,
there is no results displayed. Any reflexions about that ?
They won't be found in the CLI command until Asterisk receives another
packet
from
All generic parameters are still taken from sip.conf and you must set
rtcachefriends=yes
If you change anything in your mysql sip table you do not need to reload
the modue, what you need to do is
sip prune realtime peername
from the CLI
As stated previously, you should never have to
I try CLI command sip prune realtime peer name and my peer infos was
perfectly updated when I do sip show peer name but have you any idea
of how I can do that automatically ?
How are you updating your sip table? Are you doing it manually or have
you built an interface for it? If you have
Well, it shouldn't. The Contact header should identify how to contact the
peer, and that is currently saved into the database on register (unless you
have updates turned off).
Which column are concerned by ? my update is on because some column are
dynamically updated each time a terminal
Unfortunately not, I built ours myself.
ok thank for all these advices and sol
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Hi
I Have a problem with mysql/asterisk realtime interaction.
each time I try to connect a sip phone or to use this CLI command - realtime
mysql status - I obtain this error message :
Mysql Realtime: Failed to connect database server asteriskdb on localhost
(err 2002)
here a sample of my
mysql -uasterisk -pasterisk asteriskdb
When I do that in a linux terminal it works.
But I always have this err 2002.
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Hi
Is it any interest to use realtime applications if I use mysql to store my
sip terminal informations ?
Can I use some informations from mysql databese and sip.conf in same time ?
Anyone already use Asterisk with Active Directory for centralized database
for authentication information and
anyone already used the realtime driver for LDAP in order to interact
Astérisk with Active Directory ?
regards
Harry
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2009/8/14 Pascal Bruno tipas...@gmail.com
Did you get CDRTool to work with Asterisk or Areski's CDR Stats?
Hi finally use Areski but until now I dont try all features, just CDR Report
button. But I had a quick look on other button and it seems work.
Hi
I just solve my problem today. Just a package on redhat that I need install.
H.
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Hi
Someone may have an issue to my problem :
- I install mysql server and create a database named asteriskdb to store
some data of asterisk. OK
- I create a table cdr in order to replace de Master.csv . OK
- I set up Areski CDR (asterisk-stats) for my environnement thank to
I linked to Areski's CDR Stats (which I've used a few times):
http://www.areski.net/asterisk-stat-v2/about.php
Asterisk-Stat is a visualisation layer for Asterisk CDR statistics which
are pulled from a database. It provides graphs as well as allowing you
to get more information on
2009/8/13 Ishfaq Malik i...@pack-net.co.uk
Have you configured your /etc/asterisk/cdr_mysql.conf file?
Yeah
I configure it. Now everytime I do a call, a CDR line is added in my table
cdr.
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Hello
Anyone who have already use/configure Asterisk with CDRTool ?
Or maybe can suggest another CDR GUI ?
regards.
Harry
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Dear all,
I want to setup the incoming calls, that don't use authentication in
sip.conf file.
My configurations as follows,
[2000]
type=peer
host=dynamic
insecure=port,invite; (both)
context=Testing
But when I call '2000', I noticed the following message in Asterisk
CDRTool operates on CDRs generated by RADIUS servers into the standard
'radacct' schema, along with some custom attributes added by OpenSER.
CDRTool is designed for use with OpenSER, not Asterisk.
src http://www.voip-info.org/wiki/view/Asterisk+GUI
CDRTool
Anyone know how to use regcontext et regexten parameter from sip.conf
and can give an example ?
Sure... let's say I have a phone with the following configuration in
sip.conf:
[myphone]
type=friend
context=inside
host=dynamic ; phone will register w/ Asterisk
secret=mysecret
Hi
Anyone know how to use regcontext et regexten parameter from sip.conf and
can give an example ?
thx
regards
Harry
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2009/8/6 Alex Samad a...@samad.com.au
On Fri, Jul 24, 2009 at 08:28:48AM -0500, Danny Nicholas wrote:
Here's how I think your dialplan should look:
exten = 101,1,Ringing
exten = 101,2,Answer()
exten = 101,3,Dial(SIP/quentin,10)
exten = 101,n,VoiceMail(1...@default,u)
exten =
- what's the difference between a subscribe request et a register
request ?
A subscription in the SIP protocol is saying Hey, I'd like to be
notified when something happens. This is most often used when a phone
wants to subscribe to the state of another extension, or to the status
of a
exten = 101,1,Ringing
exten = 101,n,Answer()
exten = 101,n,Dial(SIP/quentin,10)
exten = 101,n,Goto(101-${DIALSTATUS},1)
exten = 101-NOANSWER,1,VoiceMail(1...@default,u)
exten = 101-NOANSWER,n,Playback(vm-goodbye)
exten = 101-NOANSWER,n,Hangup()
exten = 101-BUSY,1,Playback(busy)
Hello
I have few questions :
- what's the difference between a subscribe request et a register request ?
- in asterisk 1.6 allowguest=yes or no param does it work ? if yes, please
someone could explain how doest it work because I think i'm a little bit
confuse.
- if I configure a sip terminal in
Hi jonathan
thx for the tips but with your solution the user can only log from one
computer. .
I thinks your solution may be usefull if the user use a physical phone but
if he use a softphone he became dependent of the IP adress. :-(
but I may have another way : get informations from asterisk
Hi everybody
In advance sorry for my bad english and if my problem was already exposed (I
didn't find any tips in the mailing list archive. Bad luck)
I have some questions about asterisk 1.6 release :
1) how can I do a n+101 priority jumping if a SIP canal is busy ?
I read that the general
, but
here’s a find and grep that would tell you when a user’s box was full (based
on default params):
find /|grep msg0099.txt|grep INBOX
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*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *harry R
*Sent
Hi
I have a new question. Here the situation :
I use softphone on 2 computers (soft1 and soft2) located on the same
subnetwork.
When I register on asterisk server using soft1 with one user (e.g JOHN)
which I declared in sip.conf I can register again with this same user using
soft2.
Is it normal ?
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