similar to: R: Capturing DTMF during a call

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

I did it Steve, but on some calls i still have the EC on OFF. What can i check? Could it depend of my zapata.conf ? Thanks Giordano Grandis -----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 17.08 A: Asterisk Users Mailing List - Non-Commercial Discussion

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:

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

Hi all, I have a simple and maybe also stupid question: if i'm in coversation on a Zap channel and the remote party send me a DTMF, could I capture it? Thanks all Giordano -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060306/35273048/attachment.htm

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':

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]

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.

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:

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

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

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 -

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

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

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

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

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:

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:

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

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: >