Displaying 20 results from an estimated 3000 matches similar to: "R: R: Echo cancellation"
2006 Mar 29
0
R: RE : Echo cancellation
Hi Francois,
I kwnow, but I have "DSP:on" also if i not have an hardware echocan module :/ and I always have "Echo Cancellation: 0 taps, currently OFF".
This is my zapata.conf
[channels]
language = it
usecallerid = yes
callwaiting = yes
usecallingpres = yes
callwaitingcallerid = yes
threewaycalling = yes
cancallforward = yes
callreturn = yes
switchtype = euroisdn
2006 Mar 28
3
R: Echo cancellation
Ok, but is there a way to check if echo cancellation is active on a call in progress ?
Thanks
Giordano
-----Messaggio originale-----
Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Steve Davies
Inviato: marted? 28 marzo 2006 16.43
A: Asterisk Users Mailing List - Non-Commercial Discussion
Oggetto: Re: [Asterisk-Users] Echo cancellation
2005 Sep 27
1
R: Best drivers for HFC-S ISDN cards
Mine is very similar: i don't have echocancelwhenbridged=yes because it seems work only on TDM, is it ?
And in Italy, I often have set pridialplan = unknown
About echo I have some problems, but only at the beginning of the call. After 3-4 seconds the echo became almost null, specially with snom 190; with pa168s and ywh10 I have again some problem, the echo come up also after 1 minute of
2005 Sep 30
0
R: chan_capi-0.3.5
Thanks Jorg,
it's worked, but what modules i need to use it with asterisk?
I insert load => chan_capi.so in /etc/asterisk/modules.conf and chan_capi.so=yes under [globals] section.
When asterisk start, I get this error:
== Parsing '/etc/asterisk/modules.conf': Found
[chan_capi.so] => (Common ISDN API for Asterisk)
== Parsing '/etc/asterisk/capi.conf':
2005 Oct 03
1
R: codec g723 on Via C3
Thanks...which version of IPP did u use ?
I do not have Makefile file....there is only a .sh script
Thanks
Giordano
________________________________
Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Juan Salas
Inviato: luned? 3 ottobre 2005 15.41
A: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Oggetto: RE:
2006 Jan 12
1
R: app_rxfax.so and app_txfax.so
I have to re-compile also app_rxfax.so and app_txfax.so or just spandsp ?
Thanks
Giordano
________________________________
Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Colin Anderson
Inviato: gioved? 12 gennaio 2006 17.20
A: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Oggetto: RE: [Asterisk-Users]
2005 May 31
2
R: R: R: R: AT-320 + supervised transfer
Good...it almost works fine! I just have 't' in command Dial, but i also have 'T'. Is it a problem ?
This is my Dial()
exten => 605,1,Dial(${GIORDANO NAT},60,Ttr)
I have only a problem: A and B are speaking, B calls C and ask it if wants speak with A, C accept but if B hang up A is waiting and C get busy tone. To make it works B don't have to hangup but habe to press
2005 Sep 29
2
R: PRI value
Perfect, thanks very much hth. I just set it to unknown, but it doesn't work.
Have I to use also prilocaldialplan ?
Thanks again
Giordano
________________________________
Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Colin Anderson
Inviato: gioved? 29 settembre 2005 16.22
A: 'Asterisk Users Mailing List -
2006 Mar 07
0
R: Capturing DTMF during a call
Thanks Kristian,
but i just answered to call, how can i use the Read application?
Thanks
Giordano Grandis
-----Messaggio originale-----
Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Kristian Kielhofner
Inviato: luned? 6 marzo 2006 18.15
A: Asterisk Users Mailing List - Non-Commercial Discussion
Oggetto: Re: [Asterisk-Users] Capturing
2006 Jan 31
2
R: Kirk IP600
I'm going to try,
Thanks very much
Giordano
-----Messaggio originale-----
Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Remco Barende
Inviato: luned? 30 gennaio 2006 20.04
A: Asterisk Users Mailing List - Non-Commercial Discussion
Oggetto: Re: [Asterisk-Users] Kirk IP600
Hi!
Yes, it works (sort of) but I still have some issues.
2005 May 30
4
R: R: R: AT-320 + supervised transfer
I known. I'm using the 1.44 firmware version relesed on 26 may. I worked for italian IVR an HTTP pgaes.
So i can only update asterisk with CVS and try atxfer.
Thanks for all
-----Messaggio originale-----
Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Gavin Hamill
Inviato: luned? 30 maggio 2005 18.40
A:
2005 Jun 01
1
R: R: R: R: R: AT-320 + supervised transfer
No...maybe i don't explain u well.
After that B call C andC not answer (go in timeout), B hear first the beeperr and then, together A the busy tone. Now i can't re-take the call :|
Thanks
Giordano
-----Messaggio originale-----
Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Gavin Hamill
Inviato: mercoled? 1 giugno 2005 12.34
A:
2005 May 30
2
R: R: AT-320 + supervised transfer
The procedure that will do asterisk is very nice ;) but whe it was available ?
Currently is there any way to emprove the transfer? I tryied the scenario that u suggest me but it doesn't work :| and i don't why.
Here my sip.conf for the phone, can u say me if there is somethingh wrong ?
[2391]
type=friend
username=2391
secret=2391
language=it
host=dynamic
context=intern
dtmfmode=rfc2833
2005 Sep 16
2
R: direct sip call pickup
I cannot use CVS, is there anoyher way to use direct pickup ?
Thanks again
Giordano
________________________________
Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Alexander Lopez
Inviato: venerd? 16 settembre 2005 17.53
A: Asterisk Users Mailing List - Non-Commercial Discussion
Oggetto: RE: [Asterisk-Users] direct sip call
2006 Jan 20
0
R: Dect to SIP PCI card
And a PRI/BRI gateway? Is there any product ?
Thanks
Giordano
-----Messaggio originale-----
Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Peer Oliver Schmidt
Inviato: venerd? 20 gennaio 2006 10.17
A: Asterisk Users Mailing List - Non-Commercial Discussion
Oggetto: Re: [Asterisk-Users] Dect to SIP PCI card
Giordano Grandis wrote:
>
2005 Oct 03
4
R: Diva
Which models of Diva could work with CAPI and asterisk?
Thanks
Giordano
________________________________
Da: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] Per conto di
gw@adcomcorp.com
Inviato: sabato 1 ottobre 2005 23.46
A: asterisk-users@lists.digium.com
Oggetto: RE: [Asterisk-Users] Diva
Nope. At least I tried and never could get it
2005 Jun 30
3
R: Music oh hold
This is my musiconhold.conf and my folder:
root@voip:/etc/asterisk# less musiconhold.conf
[classes]
default => quietmp3:/var/lib/asterisk/mohmp3
;loud => mp3:/var/lib/asterisk/mohmp3
;random => mp3:/var/lib/asterisk/mohmp3,-z
;unbuffered => mp3nb:/var/lib/asterisk/mohmp3
;quietunbuf => quietmp3nb:/var/lib/asterisk/mohmp3
; Note that the custom mode cannot handle escaped parameters
2005 May 30
3
R: AT-320 + supervised transfer
Hi,
Thanks for yuor answer.
The boot time of the phone is very very fast, 10 sec to startup and 2 or 3 second to login to asterisk. I set the NTP server to 255.255.255.255 so it don't try to get time.
I thinked carefully to your scenario and i am going to try it, but i don't known if it could like to my customer
I will try also to use CVS, but i am skeptic to utilize asterisk to
2005 Jun 27
2
R: zaphfc: empty HDLC frame or bad CRC received
I have the same problem in a box with 2 HFC-PCI, but i already remove the row in modprobe.conf and load the module manually.
Both cards works fine
Any idea ?
Giordano
-----Messaggio originale-----
Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Julian J. M.
Inviato: luned? 27 giugno 2005 12.04
A:
2005 Jun 29
4
Music oh hold
Sorry, i also tried this:
exten => 6000,1,Answer
exten => 6000,2,MusicOnHold(default)
and i got this result:
*CLI> -- Executing Answer("SIP/2391-8cdd", "") in new stack
-- Executing MusicOnHold("SIP/2391-8cdd", "default") in new stack
Jun 29 19:33:47 WARNING[1616]: res_musiconhold.c:354 moh0_exec: Unable to start music on hold (class