Displaying 20 results from an estimated 6000 matches similar to: "fastagi example"
2009 Sep 25
3
disable dtmf on SIP peer
Hello,
I have one problem and I need to disable dtmf (disable rfc2833, info and
inband) on one (other peers must support dtmf) SIP peer . Is it possible?
Workaround would be use g729 codec with dtmfmode=inband.
Maybe there is better solution?
Thanks for help.
--
Pagarbiai / Best Regards,
Giedrius Augys
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2008 Nov 17
1
asterisk conference
Hello,
I've asterisk 1.4.22. I need to that the first conference user hears
"You're the only conference user..." . When the second user joins (without
recording his name) , the first user only hears "new user have join" , when
the third user joins to conference, others hear "new user have join" and so
on. I'll try to do this with meetme, but it always
2008 Dec 16
2
starting call recording using AMI or other stuff
Hello,
Is it possible, that during the call one side , for examples clicks the
button on the web, and this call starts recording? It's possible with
asterisk feature automon and DTMF. So it is possible to start recording the
channel using AMI or ... ?
Thanks
--
Pagarbiai / Best Regards,
Giedrius Augys
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2008 Oct 05
5
asterisk, phpagi and singleton
Hello,
I've this situation: 300+ simultaneous calls and dialplan like this:
exten => _X.,1,Answer()
exten => _X.,2,DEADAGI(check_status.php)
exten => _X.,3,Dial(SIP/other/${NUMBER})
exten => _X.,4,Hangup
exten => h,1,DEADAGI(cdr.php)
When project is running , I had a lot of defunct php scripts (I've exceed
mysql connection limits and so on, deadagi help a bit). The
2009 Feb 27
1
change language and playback issue
Hi,
I have problem with Asterisk 1.6.0.1. I need to change language for
playing prompts in Lithuanian. But in Asterisk 1.6.0.1 version everytime
plays in English, but in Asterisk 1.4.x I haven't any problem. Maybe it is a
bug ...? So I paste my test dialpan and prompt's locations. I hope this
helps you.
Files are:
[root at voip asterisk]# find /var/lib/asterisk/sounds/test -name
2008 Nov 26
1
language and meetme issue
Hello,
I have created a dynamic conference into two languages (english and
russian). Client calls to confrence number and interactive choose the
language. Meetme runs with 'dMi' options. Everything works perfect if one
conference room clients have choosed the same language. If clients had
choosed different language , there is a problem with user join/leave
announcements. For example:
2008 Dec 17
1
ael queue gosub already has PBX structure??
Hello,
I want that after client and queue member call would be established, cmd
queue runs some 'procedures' . So I am using cmd Queue option 'gosub'. This
is my example of ael :
context QUEUE {
_X. => {
Ringing();
Wait(4);
Answer();
Queue(${Queue},wr,,,60,,,check-record);
Hangup();
};
};
macro check-record() {
2006 Jun 14
0
NCS patch
Hi,
I have cable modems Arris with MGCP protocol. And I need PacketCable
NCS patch for Asterisk. http://asterisk.urtho.net/ doesn't work!
--
Pagarbiai,
Giedrius Augys
Siauliu Universitetas, IST
IP telefonijos inzinierius
Tel. 8 41 590408
Mob. Tel. 8 678 05790
el. pastas voipas@gmail.com
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2007 Oct 17
2
asterisk hylafax iaxmodem
Hi,
I have problems with asterisk and hylafax+ iaxmodem. I can successfully send
faxes to Panasonic KX-FT932 fax, but with Xerox WorkCentre M20i I have
problems: No carrier. This is hylafax log, maybe you can suggest me where
to find ...
Oct 17 07:38:48.22: [22428]: SESSION BEGIN 000000041 180037052390906
Oct 17 07:38:48.22: [22428]: HylaFAX (tm) Version 4.4.2
Oct 17 07:38:48.22: [22428]: SEND
2009 Aug 18
1
avoid indicate condition 9 and starting music on hold
Hello,
I've a problem. I've asterisk 1.6.0.5 version. And I've created
callcenter, but agents registers to another SIP server. When agent tries
transfer a client to another operator , pressing flash, I get this:
[Aug 18 16:06:37] WARNING[5259]: chan_sip.c:5349 sip_indicate: Don't know
how to indicate condition 9
[Aug 18 16:06:37] WARNING[5259]: channel.c:2858 ast_indicate_data:
2008 Nov 24
1
play sound while executing agi script
Hello,
Is it possible to do like this: play a sound file (if needed play in loop)
while php agi script finishes work ? And how to do this? When on my server
is huge load , I don't want that client hears silent , but hears music.
Thanks
--
Pagarbiai / Best Regards,
Giedrius Augys
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2008 Dec 01
1
func_odbc questions
Hello,
I'm working with asterisk 1.6. And I have success using func_odbc with one
row query results (SELECT source,destination from cc WHERE ... ):
exten => s,1,Ringing
exten => s,n,Wait(4)
exten => s,n,Answer
exten =>
s,n,Set(ARRAY(NUMBER,REALNUMBER1,REALNUMBER2,STATUSAS)=${ODBC_GETVARIABLES(${NUMERIS})})
exten => s,n,Verbose(1| ${NUMERIS}, ${REALNUMBER1} ${REALNUMBER1},
2008 Dec 02
1
func_odbc and hash problem
Hello,
Now I'm testing func_odbc and hash. My configurations are:
func_odbc.conf
[GETNUMBER]
dsn=sqlserver
;mode=multirow
;rowlimit=10
readsql=SELECT number,real_number1,real_number2,status FROM ivr.dbo.numbers
WHERE number=${SQL_ESC(${ARG1})}
extensions.conf
exten => s,1,Ringing
exten => s,n,Wait(4)
exten => s,n,Answer
exten => s,n,Set(NUMERIS=37037210602)
exten =>
2008 Dec 16
0
realtime odbc queue member cache problem
Hello,
I have asterisk 1.6.0.1. I'm using realtime asterisk with MS SQL.
Everything is OK, but I have noticed strange thing with queue members. If I
modify just 'membername' - asterisk do not refresh this info. But if I make
changes also in 'interface' column, and after executing command `queue show
my-queue` and I see changes.
So is it possible that asterisk after
2008 Dec 19
0
realtime queue change ring strategy
Hello,
I'm using asterisk 1.6.0.1 and realtime queues. But when I make changes
in database (for example: change strategy from ringall to random), but
asterisk shows old strategy, doesn't update this parameter.
My question is, how I can dynamically change ring strategy.
Thanks in advance.
--
Pagarbiai / Best Regards,
Giedrius Augys
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2009 Mar 24
0
originate and local channel problem
Hello,
I want originate a call to some destination, and when B side answes to
play a prompt. Asterisk version is 1.6.0.5. But also I need to insert a SIP
header to Invite, that's why I'm using Local Channel. This is my
extension.ael:
context autodialer-local {
_X. => {
SipAddHeader(P-Asserted-Identity:
<sip:${CALLERID(num)}@xxx.xxx.xxx.xxx;user=phone>);
2009 Sep 25
0
ignore flash hook
Hello,
I've created Call Center with Asterisk (1.6.0.5). Call Center's agents
are not Asterisk SIP user's, but other's voip gw SIP 5 class users.
Everything works fine, except when one agent wants transfer call to other
agent. They do it with flash hook. So and two voip gws (Asterisk and other
gw) detects this flash and both starts playing MOH. The transfer is
unsuccessfully,
2009 Nov 06
1
app read accept # sign
hello,
I'm using Asterisk 1.6.0.5 . And I'm creating IVR, and I need that Read
application accepts # sign,
So is it possible? And maybe there is a workaround?
Thanks
--
Pagarbiai / Best Regards,
Giedrius
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2010 Jan 07
1
compile one additional module without recompiling all asterisk
Hello,
Maybe there is the easiest way to compile additional my module without
recompiling all asterisk?
Thanks
--
Pagarbiai / Best Regards,
Giedrius
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2010 Jan 21
1
odbc question
Hello,
I want to know what is timeout for MS SQL connection? My config is:
[mydb]
enabled => yes
dsn => MYDB
pooling => yes
limit => 200
share_connections => no
username => login
password => password
pre-connect => yes
backslash_is_escape => no
In the peak , I can see :
ODBC DSN Settings
-----------------
Name: mydb
DSN: MYDB
Pooled: Yes
Limit: 200