Displaying 20 results from an estimated 4000 matches similar to: "Playing announcement to called user prior toConfirmation"
2003 Nov 07
2
Callgroups and Pickupgroups in Console/dsp
Hi all.
I've made a patch for chan_oss.c to enable
callgroups and pickupgroups in it (since wasn't enabled).
I needed it for a special use of the console (pickup
calls arriving to the console from another phone)
btw, If someone is interested, I can submit a patch
to the bugtracker. I won't do it until
that's usefult for someone... since is a very special
features that probably no
2004 Feb 02
1
Playing announcement to called user prior to Confirmation
Hello all,
As I'm sure is pretty common, I have some extensions that dial mobile numbers
after a local timeout. I would like to prompt the caller to record their
name after the local timeout and have the recipient be able to hear the name
prior to accepting the call.
Recording the message is easy enough, so I thought about doing something like
dumping them into MeetMe after they record
2003 Jul 23
2
SIP info
I was wondering what are the values for sending dmtf
via sip info.
I mean, when I use dtmf relay via sip info, the sip/sdp message
contains a Signal=X where X is the dmtf.
That's ok for dtmf 0-9 . but what when dtmf is * or # ?
we must send signal=# ?
I ask that because I noticed that budgetones phone sends out
* as signal=10 and # as signal=11 . but asterisk
don't detect them, 'cause
2003 Jul 14
3
New budgetone firmware
Hi.
Has anyone experienced with the new firmware .77 ?
There's Day Light Saving time now, but haven't
time to play with it, till now.
Matteo.
--
Matteo Brancaleoni
Espia System Administrator - IT services
Website : http://www.espia.it
Email : mbrancaleoni@espia.it
2004 Jan 08
5
AbsoluteTimeout Users Messages
Hi, All
Is there a provision for "AbsoluteTimeout" application to notify
called and calling party of the reason why the call suddenly ended?
This way, the parties will be much better informed, hence they
will/should not think that
their VOIP/telco provider(s) are providing bad service.
Ta
SJ
2003 Jun 13
3
Call queues for phone operator
Hi.
I was wondering how can I make incoming calls to wait if the phone
operator is busy. I've 8 incoming lines, with 30 extensions.
What I need is if the operator is busy with call nr #1 , the new
incoming call waits until the op. is free.
Looking into app_queue seems the way to go.
So I want to ask if I'm right or wrong:
I set up only a queue , is to say operatorq, where
the only member
2003 May 01
2
Max number of connection in IAX ?
Hi.
I was wondering if there's a parameter to limit
the number of concurrent sessions in IAX, globally or
on a per-user basis.
That could be needed for security purposes
(to prevent dos attacks), to limit bandwidth / cpu usage, or
to not allow more than N guest connections, for example.
Any other VoIP channel support that?
(like SIP, MGCP)
Matteo.
--
Brancaleoni Matteo
2003 May 25
2
Message Waiting and VoiceMail 2
Hi.
I noticed that if new messages are recorded
with voicemail2 , they're not detected by
the message waiting indicator, so
the mailbox=XXXX param has no effect, and
no message waiting is sent to the phone
(sip & adsi, or stutter dialtone)
Any hint?
--
Brancaleoni Matteo <mbrancaleoni@espia.it>
Espia - Emmgi Srl
2003 Apr 18
1
Account code on SIP
I was wondering if the accountcode flag works
with sip channels. I was looking into the
debug and ,even if I have the line accountcode=XXX
into the users sections of my sip.conf, I don't see
it logged into the cdr.
Matteo Brancaleoni
mbrancaleoni@espia.it
Emmegi System Administrator
EspiA - EMMEGI Srl - e*solution provider
Uffici: Via Pascoli, 37
20129 Milano - Italy
Sede Legale: Corso
2003 Apr 08
1
Wiki for the * community.
Hi 2 all.
I was thinking to start a little web site with phpwiki,
to let the * community build a sort of shared
documentation 'bout * & related.
That because in a wiki "place" all grows faster,
and is also the right place to share experiences.
For example it's right to have documentation
about * installations, ie who has done what with asterisk
Till now we don't know
2004 Jan 22
4
Gsm + snom phones
Hi.
I'm not using snom phones for a while, but
now I want to test again them and I'm gonna
buy a snom 200 & 105 .
Some times ago I had a snom 100 , and gsm wasn't
working with *. How's now the situation?
the snom gsm works well with * ?
Thanks for any info, Matteo.
--
Matteo Brancaleoni
Espia System Administrator
Email : mbrancaleoni@espia.it
Web : http://www.espia.it
2003 Oct 07
1
[PATCH] allow announcements in app_dial
Hi.
Since a customer requested us that feature, I wrote this
little patch for app_dial to allow to play an
announcement to the called party, as soon he answers.
you can define the file to play in the dial() option,
using A(filename).
for example:
exten => blah,1,Dial(Zap/blah,30,rA(/my/own/announce)Tt)
that doesn't break anything ...
feel free to blame me for anything bad this patch
2004 Jan 04
8
Grandstream Handytone 286 RTP Problems
I am trying to get the handytone 286 to make a very simple call to * and
having problems. It registers with * just fine, but when I place a call
(to echo test, for example), the RTP stream seems to have problems
opening. Here is there error I get in *:
WARNING[98311]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries
exceeded on call 20d1c411-e210-5f3d-3f88-19035c8fcb26@192.168.2.6 for
2004 Feb 03
2
busy tones
Hi
When I call a phone with CAPI if the phone available I hear ringing ok
but if the phone is busy I don't hear anything at all.
Also, when I call a mobile phone and it is turned off I don't hear the
operator voice answer me telling me that the request phone is turned off
or unavailable.
Any ideas?
m
2004 Apr 23
1
CAPI and Extensions.conf Security problem
Hi,
I've installing a AVM Fritz Card in my ASterisk Box
I've configured everything and its running perfectly.
The problem is that everybody is allow to call through it.
Explaination:
All users registered in Asterisk can make a call towards the ISDN network
But, everybody from the Internet, knowing the extension of CAPI in the
dialplan, can call through my Asterisk to any phone
2004 May 06
3
mpg123 versions ?
We find that mpg123 0.59r works best. mpg123 0.59s-mh4 = the devil.
What versions does everyone use without problems.
0.59r is PERFECT
bkw
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2004 Jan 31
1
asterisk php status viewer
since I was annoyed this morning, I
wrote this simple php script to output
channel status from asterisk manager.
<disclaimer>
that's very bad written, nor commented...
I wrote that just for fun
</disclaimer>
and if someone will use that / improve
it , just lemme know.
http://asterisk.espia-net.net
(wrote with php 4.3.3 and depends
on Event: StatusComplete, so a recent
* cvs
2003 Jul 05
3
Activate MySQL logging
<P>hi,</P>
<P>Can anybody pls tell me how to activate loggin CDR on mysql db. I tried editing the /etc/asterisk/cdr_mysql.conf file and recompiling asterisk, it didn't work. Normal loggin on Master.csv file works fine.</P>
<P>Thank you inadvance,</P>
<P>Surajee</P>
<P> </P><br>
--------------This mail sent
2003 Sep 11
1
UK Asterisk user, please pick up the white courtesy phone
So, I have submitted my configurations as public samples, and I
should have expected this situation to arise. I changed all the
relevant "private" configuration data in my samples to obfuscate or
alter IP addresses, passwords, etc. However, I left my email address
in voicemail.conf...
Let me tell you, it took THREE messages sent by a distinctly
British-sounding gentleman leaving
2003 Nov 10
1
Jitter Buffer on chan_sip
Hi,
I would like to test chan_sip with a bigger jitter buffer. Does anybody know
where in the code this is defined? I looked through it but could not find
where.
If anybody else can find it please let me know.
Regards,
Andres