Displaying 20 results from an estimated 2000 matches similar to: "codec g723 on Via C3"
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]
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
2006 Jan 12
6
app_rxfax.so and app_txfax.so
Hi,
I just installed spandsp 0.0.3pre6 with libtiff 3.7.1 and evrythinghs is
ok. I recompiled asterisk and run ldocnfig, but whrn I start asterisk I
get this error:
[app_rxfax.so]Jan 12 16:40:42 WARNING[1569]: loader.c:258
ast_load_resource: /usr/lib/asterisk/modules/app_rxfax.so: undefined
symbol: fax_set_phase_d_handler
Jan 12 16:40:42 WARNING[1569]: loader.c:440 load_modules: Loading
2006 Jan 20
3
Dect to SIP PCI card
Hi all,
I'm looking for a PCI card which i could install on asterisk box, with
purpose to use 15-20 cordless dect phone in a very "dect cell".
Is there anyone that could help me pls ?
Thanks
Giordano
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2005 Sep 29
2
PRI value
Hi group,
anyone can explain me the exact difference between pri value in
zapata.conf ?
; PRI Dialplan: Only RARELY used for PRI.
;
; unknown: Unknown
; private: Private ISDN
; local: Local ISDN
; national: National ISDN
; international: International ISDN
If I use it, I also must use prilocaldialplan = local ?
Thanks
Giordano
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2005 Sep 30
2
chan_capi-0.3.5
Hi all,
i'm tryinf to install chan_capi but i get this error
root@obelix:/usr/src/chan_capi-0.3.5# make
gcc -pipe -Wall -Wmissing-prototypes -Wmissing-declarations -g
-I/usr/include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i586
-DNEVER_EVER_EARLY_B3_CONNECTS -DCAPI_ES -DCAPI_GAIN
-DDEFLECT_ON_CIRCUITBUSY -DCAPI_SYNC -DUNSTABLE_CVS
-Wno-missing-prototypes -Wno-missing-declarations -DCRYPTO
2005 Oct 05
2
Intel Pentium Celeron
Hi all,
i'm going to install asterisk with a 4 BRI (HFC chipset) on a Celeron at
2.6 GHz
I don't known Celeron performance, but i listen that is not very good.
Could I have some performance isuue with this kind of processor ?
Thanks for all
Giordano
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2004 Apr 26
3
Compiling asterisk
I got the error below while compiling asterisk. Please offer me some help.
hubert
for x in res channels pbx apps codecs formats agi cdr astman stdtime; do
make -C $x depend || exit 1 ; done
make[1]: Entering directory `/etc/asterisk/asterisk-0.7.2/res'
make[1]: Nothing to be done for `depend'.
make[1]: Leaving directory `/etc/asterisk/asterisk-0.7.2/res'
make[1]: Entering
2006 May 04
2
PCI voltage
Hi all,
I have to bought a PCI with 4 PRI but on digium site I saw that there a
re two different kind (3,3V and 5v). What's the difference?
Which one I have to buy for do not have any problem with this
motherboard? (Gygabyte GA-8S661FXM-775). I checked on Gigabyte website
but I don't find any kind of this value :-|
Thanks all
Giordano
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2005 May 31
2
R: R: R: R: AT-320 + supervised transfer
Good...it almost works fine! I just have 't' in command Dial, but i also have 'T'. Is it a problem ?
This is my Dial()
exten => 605,1,Dial(${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 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 Sep 01
2
ztcfg problem
Hi all,
I'm installing two HFC pci card (both in TE mode), I don't have problem
when load module, but whrn I give "ztcfg -vv", I see 6 the six channels
that I configured, then my computer hang and I have to reboot it. (I'm
using a VIA Epia-M 1000 with Via C3 processor)
root@test:~# modprobe zaptel
root@test:~# insmod /usr/src/bristuff-0.2.0-RC8n/zaphfc/./zaphfc.o
2008 Feb 13
1
GXP2000 and asterisk 1.0.9
Hi all gusy,
i have a big problem with gxp2000 and asterisk 1.0.9 The phones after a few go in "busy" state, if you call it get the busy tone but the phone can male any type of call.
This is my sip.conf
[502]
language = it
username = 502
secret = <password>
host = dynamic
type = friend
context = local
canreinvite = yes
dtmfmode = info
callgroup = 1
pickupgroup = 1
callerid = 502
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 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 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
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
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 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: