Displaying 20 results from an estimated 3000 matches similar to: "language and meetme issue"
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 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/${NUMBER})
exten => _X.,4,Hangup
exten => h,1,DEADAGI(cdr.php)
When project is running , I had a lot of defunct php scripts (I've exceed
mysql connection limits and so on, deadagi help a bit). The
2007 Oct 17
2
asterisk hylafax iaxmodem
Hi,
I have problems with asterisk and hylafax+ iaxmodem. I can successfully send
faxes to Panasonic KX-FT932 fax, but with Xerox WorkCentre M20i I have
problems: No carrier. This is hylafax log, maybe you can suggest me where
to find ...
Oct 17 07:38:48.22: [22428]: SESSION BEGIN 000000041 180037052390906
Oct 17 07:38:48.22: [22428]: HylaFAX (tm) Version 4.4.2
Oct 17 07:38:48.22: [22428]: SEND
2008 Dec 16
2
starting call recording using AMI or other stuff
Hello,
Is it possible, that during the call one side , for examples clicks the
button on the web, and this call starts recording? It's possible with
asterisk feature automon and DTMF. So it is possible to start recording the
channel using AMI or ... ?
Thanks
--
Pagarbiai / Best Regards,
Giedrius Augys
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2008 Dec 17
1
ael queue gosub already has PBX structure??
Hello,
I want that after client and queue member call would be established, cmd
queue runs some 'procedures' . So I am using cmd Queue option 'gosub'. This
is my example of ael :
context QUEUE {
_X. => {
Ringing();
Wait(4);
Answer();
Queue(${Queue},wr,,,60,,,check-record);
Hangup();
};
};
macro check-record() {
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 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 =>
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:
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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2009 Dec 07
3
show queue's name and other info in incoming call to queue member
hello,
I've callcenter and our queue members want to see on their IP phone's
display queue's name , from which incoming call was originated, for
example "<client's_number> -> Sales". This problem appears when one member
can belong to couple queues. Work around would be setting calling name with
such information.
Maybe there is another way (setting SIP
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
2009 Feb 09
2
meetme application
hi guys:
recently I want to buinding a meeting confence on asterisk and use the meetme application.
I have a ztdummy kernel
afteri the lsmod commond:
ztdummy 5768 0
zaptel 182660 28 zttranscode,ztdummy
crc_ccitt 3008 1 zaptel
I also configure the meetme.conf
conf => 1000;
my extensions.conf
[default]
exten =>
2008 Oct 18
1
strange h323 delay issue
Hello,
I have a strange h323 issue. After executing command
"Dial("SIP/333-0d1dfe00", "H323/361737052390920 at ccg|5|tT")" at Oct 18
22:32:23. Meanwile I have sniffing traffic on port 1720. The call was
established just at Oct 18 22:33:03 (New H.323 Connection created.) and also
packet sniffer grabs the h323 invites at this time also. So my question is
what
2007 Nov 11
3
detect asterisk pbx via sip
Hello,
My situation is that , I can't make calls with asterisk, but with x-lite
works fine. Asterisk shows , that successfully registers with another SIP
server, asterisk sends invite, gets trying, and after 30 secs asterisk gets
408 Request timeout. And as I said , with x-lite no problems. I heard that
for comercial purposes, this SIP server detects asterisk , and ignores him.
Or maybe it
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>);
2007 Apr 02
3
misdn and debian
Hi,
I have FRITZ!Card PCI card. I have installed misdn-1.1.0 on stable debian
3.1 with 2.6.8. kernel. Then I reboot system, and it doesn't boot, it stops
near "Apache2 starting...". I started my system with "recovery" kernel,
and tun off misd, then my system works fine. I think it's problem with
memory.
Has anybody debian and misdn working fine? Maybe you can