Displaying 20 results from an estimated 2000 matches similar to: "spandsp and freebsd"
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
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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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 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
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 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 Aug 28
1
deadagi and billsec or answeredtime
Hello,
I want to create php rate script and I'm using Deadagi. But I allways get
billsec 0 , or nothing. Can you help me to solve this problem...
My extension.conf:
exten => _123,1,DeadAgi(rate.php)
exten => _123,2,hangup
And my simple test php script rate.php
#!/usr/local/bin/php -q
<?php
include_once (dirname(__FILE__)."/phpagi.php");
$AGI = new AGI();
2006 Dec 04
1
Nokia E60 problems
Hi,
I am testing Nokia E60 with Asterisk. And I noticed that if another side
is busy, nokia is still calling (I hear alerting), it do not show that
another side is busy. Maybe somebody has noticed the same problem too adnd
solved this one. I made the same tests with Xlite and don't have any
problems like nokia.
Please help me
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2008 Feb 01
1
play promt at the same time to calling and callee
Hello,
I want that, when call is answered , callee and calling would hear
different prompts and after promts the calls would be bridged. I've tried
this situation:
exten => s,1,Set(LIMIT_CONNECT_FILE=hello-world)
exten =>
s,2,Dial(SIP/trunk-out/37052390920|60|rL(10000000000000)A(conf-enteringno))
But these prompts play not in the same time: just after conf-enteringno
prompt
2006 May 26
8
Which libtiff 3.8.2 do I grab?
Via google: libtiff 3.8.2
http://rpm.pbone.net/index.php3/stat/4/idpl/2839214/com/libtiff-3.8.2-2.x86_64.rpm.html
This got me to a page all about libtiff 3.8.2 for 'Fedora other' and a link
http://rpm.pbone.net/index.php3?stat=3&search=libtiff&srodzaj=3
for other distributions.
So which do I take?
?
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
2006 May 23
3
libtiff 3.6.1 and Asterisk spandsp
Per: http://soft-switch.org/spandsp_faq/ar01s07.html
FAXing with spandsp keeps giving me core dumps. Its crap, isn't it?
Actually the crap in this case is probably libtiff. Versions 3.5.7
and 3.6.0 seem reliable. The FAX image handling in version 3.6.1 is
broken. All the core dump problems I have investigated turned out to
be due to a very old version of libtiff, or version 3.6.1.
I am
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() {
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
2006 Jun 06
4
Fedora source packages on Centos -- libtiff again
In my seeking a managable way to build libtiff, on the Asterfax forum
I got the following recommendation (from another fax user that made
the move to libtiff 3.8.2):
================================================================
Hi Robert,
I have compiled libtiff-3.8.2 from the source package for my Fedora
Core 4 machine.
I am not sure about CentOS, but I think you can try to do the same
2010 Feb 21
2
add Reason header on hangup
Hello,
I have asterisk 1.6.0.20 and Is it possible to add Reason header on
Hangup:
Reason: q.850;cause=17
Thanks
--
Best Regards,
Giedrius
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2006 Nov 20
3
Spandsp rxfax txtax fails no errors
I'm using Slackware 11.
I unistalled the package that provides libtiff 3.8.....
and installed the most current 3.7.... for lib tiff.
I downloaded asterisk 1.4 beta3 and the 1.4 beta2 addons and untared them.
created a simlink:
ln -s asterisk-1.4.0-beta3 asterisk
I've compiled spandsp from as follows
cd /usr/src
wget