()
exten = s-CONGESTION,2,HangUp
exten = s-CANCEL,1,Congestion()
exten = s-CANCEL,2,HangUp
As anyone tried similar scenario?
Thanks all
Giordano Grandis
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AstriCon 2008
Hi guys,
I installed asterisk 1.4.17 and mISDN 1.1.7 with AGX addons to test the faxs.
If I receive i do not have any problem, but i'm not able to send put any fax, i
get always the same error:
txfax_exec: transmission done with ast_read(chan) == NULL
Anyone has txfax working with asterisk
Thanks very much, i will test it.
Hi and thanks again
Giordano Grandis
e-mail : HYPERLINK mailto:[EMAIL PROTECTED][EMAIL PROTECTED]
VoIP: HYPERLINK mailto:[EMAIL PROTECTED] sip:[EMAIL PROTECTED]
_HYPERLINK
http://%5c
and I have had no further troubles.
Hope this helps you.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giordano Grandis
Sent: Thursday, February 14, 2008 3:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] R
when you try making a phone call?
Why don't you try it with a later version of astrisk and a Phone?
On Feb 13, 2008 10:58 AM, Giordano Grandis [EMAIL PROTECTED] wrote:
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
?
On Feb 13, 2008 10:58 AM, Giordano Grandis [EMAIL PROTECTED] wrote:
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
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 =
Hi guys,
is it possible to set caller presentation with mISDN? I tryie with
SetCallerPres() and CallingPres without success...
exten = s,1,ChanIsAvail(mISDN/1)
exten = s,2,CallingPres(32)
exten = s,3,Set(CALLERID(num)=e.164_number)
exten = s,4,Dial(${CUT(AVAILCHAN||1)}/${ARG2})
Anyone can help
Look at here
http://lists.digium.com/pipermail/asterisk-users/2005-May/102837.html
Giordano
-Messaggio originale-
Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Johann
Steinwendtner
Inviato: mercoledì 7 febbraio 2007 14.56
A: Asterisk Users Mailing List - Non-Commercial
Hi all,
i downloaded and installed mISDN with 2.6.8 kernel, but when i try
mISDN-init scan (or config)
i get this error: [!!] FATAL: bc not in path, please install.
Anyone can help me.
Tnx
Giordano
--
No virus found in this outgoing message.
Checked by AVG Free Edition.
Version:
Did u try this SetTransferCapability ?
Hi Giordano
-Messaggio originale-
Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di [EMAIL PROTECTED]
Inviato: giovedì 18 gennaio 2007 17.47
A: [EMAIL PROTECTED]; asterisk-users@lists.digium.com
Oggetto: [asterisk-users] Passing video
I'm not sure that u have to use a crossover cable. Your telco give u a network
emulation, and u emulate a cpe, so i think u need a straigh cable.
Giordano
-Messaggio originale-
Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Noc Phibee
Inviato: lunedì 18 dicembre 2006 12.53
(cross
connection)
DB9 RJ45
3 4
8 5
2 1
6 2
All connections are in 120 ohm. It works in Italy with a E1 Sagem applaiance.
Thanks to all and good work.
_
Da: Giordano Grandis
Inviato: venerdì 24 novembre
Hi all,
anyone known how can i create an adapter from a DB9 port of a E1 to an RJ45
plug?
My telco left active the db9 port, but on my te407p card i have rj45 connection.
Anyone can help me pls ?
Thanks in advance
--
No virus found in this outgoing message.
Checked by AVG Free Edition.
Not exctally Tomislav, i need to get if an outbound calls (over Zap channels)
is ringing (so if is not busy or not aviable for some reason) before that the
operator transfer the calls to normal SIP user.
I would just able to get the ringing state fo the call, if so, i can transfer
the call.
Hi
guys,
i would check the
state of a number on a Zap channel, i suppose that i cannot use ExtensionState
that works only for SIP and IAX.
Anyone has any ides
? Could i check the state of a pubblic number before transfer it a internal
call?
Thanks in
advance
Giordano
--
No virus
Hi all,
I just configured a quescom 400 to route all gsm incoming calls to
asterisk, now i would route all outgoing asterisk calls to gsm port of
the quescom.
Anyone has any idea how implement it?
I did a configuration but i always get this error
-- Got SIP response 503 Service Unavailable
How do u call the quescom? With Dial() command?
exten = s,1,Dial(SIP/172.30.1.199:1123/${ARG2},Tt)
Did u set any port, or just call the ip address witout 1123 port ?
Thanks in advance
-Messaggio originale-
Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di [EMAIL PROTECTED]
Hi
list,
I'm
testingtransfer withsip re-inviteand
bristuff-0.0.8-RCnusing anHFC pci card connetced directly to telco;
this is what happen:
1.SIP phone
calls a mobile phone (or another residential phone)
2. The called party
answers the call
3. Now the sip phone
puts on hold the calland
Hi guys, i just
installed the flortz patch with bristuff-0.2.0-RC8r but when i load module
zaphfc i get this warning message:
Warning: ignoring
syns_slave=0, no such parameter in this moduleWarning: ignoring
timer_card=1, no such parameter in this moduleModule zaphfc loaded, with
: GPL
parm:modes int
parm:debug int
...i have again the kpj ?
Thanks again
Giordano
-Messaggio originale-
Da: Tzafrir Cohen [mailto:[EMAIL PROTECTED]
Inviato: martedì 3 ottobre 2006 15.43
A: Giordano Grandis
Oggetto: Re: [asterisk-users] Zaphfc woth florz patch
On Tue
Look at your
extensionsincontext "from-zaptel" adding the s extensionsand
add immediate=yes in zapata.conf
Ciao
Da: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Per conto di
bivioInviato: lunedì 2 ottobre 2006 10.04A:
asterisk-users@lists.digium.comOggetto: [asterisk-users] Siemens
Try to set
overlapdial=yes in your zapata, so thta whenu access to line ushould
have somethinghs of this
-- Starting simple switch on
'Zap/5-1' -- Accepting overlap voice call from '405' to
'unspecified' on channel 0/2, span 2
at this point u
can ear a continuos tone and input your dnid
Hi
all,
I'm using * 1.0.9
which use mpg123 for music on hold. But sometimes starts eating up a lot of
CPU.
I sthere any
alternative method to use moh without use mpg123?
I tryied this http://astrecipes.net/?n=152but i
doesn't wotks for me.
Anyone can help me
pls ?
Thanks in
advance.
Hi
guys,
i have asterisk
1.0.9 with bristuff 0.2.0 rc8n running on a VIA motherboard with processor C3
and i have this kind of problem: during the office time the system work
perfectly, but on the next moring, if i try to make an outgoing call i get this
message
== Primary
D-Channel on
the phone, leaving the caller and callee connected.
-Brodie
On Tuesday 22 August 2006 02:44 am, Giordano Grandis wrote:
Hi all,
I'm using a Snom360 with bristuffed asterisk and i want to known if is
possibile realize somthing of this: I receive an incoming call and then
answered I want
Hi
all,
I'm using a Snom360
with bristuffed asterisk and iwant to known if is possibile
realizesomthing of this: I receive an incoming call andthen
answeredI want to transfer it to a cell phone (or another pubblic number),
so press "transfer" on the phone, call the number and only if the
Hi all,
I installed cepstral and patched the Makefile in asterisk/apps, but when i do
'make install' i get this error
ntercom.o app_intercom.c
gcc -shared -Xlinker -x -o app_intercom.so app_intercom.o
gcc -D_GNU_SOURCE -shared -Xlinker -x -o app_cepstral.so app_cepstral.c -lz -lm
-lceplex_uk
on hold and then check with the destination, then transfer the
call People are strange ...
On 8/22/06, Giordano Grandis [EMAIL PROTECTED] wrote:
Hi all,
I'm using a Snom360 with bristuffed asterisk and i want to known if is
possibile realize somthing of this: I receive an incoming
How can I check if
SIP re-invite is really working ?
I'm trying it with
two grandstream gxp2000.
Thanks
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To UNSUBSCRIBE or update options visit:
-users] Canreinvite
- Original Message -
From: Giordano Grandis
[mailto:[EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Non-Commercial Discussion [mailto:[EMAIL PROTECTED]
Sent:
Fri, 28 Jul 2006 07:01:08 -0300
Subject: [asterisk-users] Canreinvite
How can I check if SIP re-invite
Hi
all,
I'm trying the
ZapRAS application and I followed this giude. Could it work with an HFC pci isdn
card? I tryied and i get this error:
*CLI -- Executing Answer("Zap/1-1", "") in
new stack -- Accepting data call from '123456789' to
'987654321' on channel 0/1, span 1 -- Executing
Hi
all,
I'm using spandsp
0.0.2pre21 with tiff 3.7.1 and asterisk 1.0.9
With rxfax
application, everythinghs is ok, but when i try to send a fax whit txfax
applicationthe channel got hangup. I'm testing send fax towards an ATA
grandstrem 488, but if i sent it to a normal fax i got the same
] Called number on ISDN
On Fri, Jul 14, 2006 at 03:34:04PM +0200, Giordano Grandis wrote:
I cannot use it, I have the immediate=yes in my zapata, the extension
will be always 's'
Why would you set immediate=yes ?
--
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755 iax:[EMAIL
Hi
all,
I have an ISDN
connection in Italy with MSN. On incoming call how can i check the dialed number
?
DNID varible could
works fine ?
Thanks in
advance
Giordano
___
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asterisk-users
- Non-Commercial Discussion
Oggetto: Re: [asterisk-users] Called number on ISDN
Check it ${EXTEN}
On 7/14/06, Giordano Grandis [EMAIL PROTECTED] wrote:
Hi all,
I have an ISDN connection in Italy with MSN. On incoming call how can i
check the dialed number ?
DNID varible could works fine
Anyone can help with
this?
cli.c:49:30:
asterisk/version.h: No such file or directorycli.c: In function
`handle_version':cli.c:414: error: `ASTERISK_VERSION' undeclared (first use
in this function)cli.c:414: error: (Each undeclared identifier is reported
only oncecli.c:414: error: for
Anyone can help with
this?
cli.c:49:30:
asterisk/version.h: No such file or directorycli.c: In function
`handle_version':cli.c:414: error: `ASTERISK_VERSION' undeclared (first use
in this function)cli.c:414: error: (Each undeclared identifier is reported
only oncecli.c:414: error: for
Hi
all,
I'm testing the
re-invite with an * box and two phones (ywh500 ewith pa1688 chipset) but it
doesn't wotk.
The phones support
the re-invite, and i use this row to dial (without Ttr)
exten =
s,22,Dial(${tjext})
and put canreinvite
= yes in my sip.conf.
Anyone can help me
pls ?
Hi
all,
I have to connect an
asterisk box to a legacy pbx using QSIG signalling : where could i find more
information or any sample ocnfiguration file?
Has anyone never
used it?
Thanks in
advance.
Giordano
Grandis
___
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Hi
all,
just a question: how
can i known the number ofSIP session?
In general and not
for a single user.
Thanks
Giordano
Le
informazioni contenute nella presente e-mail e nei documenti eventualmente
allegati possono essere confidenziali e sono comunque riservate al destinatario
della
Hi all,
i just have a question: could i Known the state of a
SIP phone without make it a Dial ?
Thanks
Giordano
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Hi all,
i just have a question: could i Known the state of a
SIP phone without make it a Dial ?
Thanks
Giordano
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Hi all,
anyone could pls explain me what does it means ?
It a part of zaptel.conf file.
LBO= Line Build Out
0:0dB(CSU)/0-133feet(DSX-1)
1:133-266feet(DSX-1)
2:266-299feet(DSX-1)
3:399-533feet(DSX-1)
4:533-655feet(DSX-1)
5:-7.5dB(CSU)
6:-15dB(CSU)
7:-22.5dB(CSU)
Thanks
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). Whats 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 dont find
Hi
all,
just a
questionhope quite simple! Why if my internet connection goes down, my
clients and sip phones stop to work? (go in logedoff)?
I have
aregister with a sip server ? It could be it ? If my * box do not register
with the server it exclude the rest of the file ?
:/
Thanks
this product myself, but according to their spec it's only
1 call.
There's another free SIP-Skype gateway from www.nch.com.au called uplink.
http://www.nch.com.au/skypetosip/index.html
Giordano Grandis wrote:
Hi all,
anyone never used PSGW as gateway beeween * and SkyPe? If yes, how
does
Hi
all,
anyone get it worked
? Uplink route me the call incoming from skype but when i answer, my skype go in
error on sound card ?
I also set in my
hosts this value:
127.0.0.1
pgp01.televolution.net 127.0.0.1 stun01.sipphone.com
This is my
sip.conf
[skype]language
= itusername =
= -2.0
immediate = no
mailbox = [EMAIL PROTECTED]
callgroup = 1
pickgroup = 1
group = 1
musiconhold = default
context = incoming
channel = 1-2
Thanks
Giordano Grandis
-Messaggio originale-
Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di [EMAIL PROTECTED]
Inviato: giovedì 30
switchtype =
euroisdnsignalling = bri_cpe_ptmppridialplan =
unknownprilocaldialplan = unknownechocanel =
yesechocancelwhenbridged = yesechotraining = yesmusiconhold =
defaultimmediate = yesgroup = 1context=incomingchannel =
1-2
Anyone can explain
me what happen ?
Thanks
Giordano
Grandis
Le in
Discussion
Oggetto: Re: [Asterisk-Users] Echo cancellation
On 3/28/06, Giordano Grandis [EMAIL PROTECTED] wrote:
Hi all,
I'm using bristuff 0.2.0 RC8o with a HFC pci card and on several calls
I saw that the echo cancellation is on OFF
Echo Cancellation: 0 taps, currently OFF (the result
Hi
all,
anyone never used
PSGW as gateway beeween * and SkyPe? If yes, how does it works? How many session
could I have on a single user ?
Thanks
all
Giordano
ThanksThis
e-mail may contain confidential and/or privileged information. If you are not
the intended recipient (or have
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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Steve Davies
Inviato: martedì 28 marzo 2006 17.08
Hi ll,
anyone never used this QuesCom 400 IP/GSM with Asterisk ? Should it works?
Thanks all
Giordano
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] QuesCom 400 IP/GSM
Giordano Grandis a écrit :
Hi ll,
anyone never used this QuesCom 400 IP/GSM with Asterisk ? Should it works?
Yes, it works fine.
Cheers,
Jean-Michel.
--
Jean-Michel Hiver - http://ykoz.net/
Découvrez la Réunion des Technologies IP Telecom
TEL: +262 (0)262 55 03 98
Thanks Kristian,
but i just answered to call, how can i use the Read application?
Thanks
Giordano Grandis
-Messaggio originale-
Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Kristian Kielhofner
Inviato: lunedì 6 marzo 2006 18.15
A: Asterisk Users Mailing List - Non
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
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Hi
guys,
just a question: can
i use the pppd application with a HFC PCI card using
bristuff.
Thanks for
all
Giordano
___
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Hi
all,
could ZapRas work on
system with a HFC isdn card?
Tnaks
Giordano
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And about app_pppd, could it work with bristuff
?
Thanks
Giordano
Da: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Per conto di Giordano
GrandisInviato: venerdì 10 febbraio 2006 17.44A: Asterisk
Users Mailing List - Non-Commercial DiscussionOggetto:
[Asterisk-Users] ZapRas
Hi
all,
://www.ecem-it.nl/hardware/Asterisk-Kirk-IP600.txt
Let me know if you find any errors / omissions, or the solution to the ringing
problem :)
On Mon, 30 Jan 2006, Giordano Grandis wrote:
Hi all,
has anyone tryied to configure asterisk with Kirk IP600 Dect-IP gateway?
Could it works using the skinny
for it, you can find it here:
http://www.ecem-it.nl/hardware/Asterisk-Kirk-IP600.txt
Let me know if you find any errors / omissions, or the solution to the ringing
problem :)
On Mon, 30 Jan 2006, Giordano Grandis wrote:
Hi all,
has anyone tryied to configure asterisk with Kirk IP600 Dect-IP
Hi
all,
has anyone tryied to
configure asterisk with Kirk IP600 Dect-IP gateway?
Could it works using
the skinny channel ?
Thanks
Giordano
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Hi
all,
i need to install
chan_h323, just a question: to use it i need the CVS ?
Is therea
version that i could use with asterisk 1.0.9 ?
Thanks
Giordano
___
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Asterisk-Users mailing
Hi all,
Im 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
___
--Bandwidth and Colocation
-Users] Dect to SIP PCI card
Giordano Grandis wrote:
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 ?
You might find something by search for Com-On-Air. But, AFAIK
It works.
Thanks
Giordano Grandis - Tecnojest srl
Phone1: 085.445.00.11 _ Phone2:
055-398.69.69 _ Fax: 085.445.94.77
VoIP-- sip:[EMAIL PROTECTED]
Le
informazioni contenute nella presente e-mail e nei documenti eventualmente
allegati possono essere confidenziali e sono
Patch it by hand. Follow this help
http://www.asteriskguru.com/tutorials/spandsp.html
Hi
Giordano
-Messaggio originale-
Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Tomislav Parcina
Inviato: venerdì 13 gennaio 2006 12.28
A: asterisk-users@lists.digium.com
Oggetto:
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:
'
Oggetto: RE: [Asterisk-Users]
app_rxfax.so and app_txfax.so
0.0.3 series releases are for development only. Roll back to
0.0.2-pre21 and you should be good.
hth
-Original Message-
From: Giordano Grandis
[mailto:[EMAIL PROTECTED]
Sent: Thursday, January 12, 2006
8:39 AM
Hi all,
anyone known if is there any SIP client to install on
an I-Mate SP5m with Windows Mobile ?
Thanks
Giordano
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Cem
Giordano Grandis
a écrit:
Hi all,
anyone known if is there any SIP
client to install on an I-Mate SP5m with Windows Mobile ?
Thanks
Giordano
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Hi all,
Im testing canreinvite = yes in my sip.conf
with snom190 and a Atcom320.Atcom320 seems support re-invite, but the snom190?
Does anyone known if this phone support it?
How I can be sure that it works?
Giordano
___
Hi all,
Im testing canreinvite = yes in my sip.conf
with snom190 and a Atcom320.Atcom320 seems support re-invite, but the snom190?
Does anyone known if this phone support it?
How I can be sure that it works?
Giordano
___
Hi everyone,
just a question: is there a way to remove this
message on the CLI ?
== Manager 'root' logged on from 127.0.0.1
== Manager 'root' logged off from 127.0.0.1
Thanks
Giordano
___
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http://www.voip-info.org/wiki-Asterisk+zaphfc
look this
Giordano
Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] Per conto di Pedro Nunes
Inviato: giovedì 27 ottobre 2005
16.23
A: asterisk-users@lists.digium.com
Oggetto: [Asterisk-Users] Bristuff
question
Hi
Hi all,
im looking for an utility that let me trace an
ISDN trunk (or all ISDN traffic) on HFC PCI card.
Is there anyone who could help me ?
Any ideas ?
Giordano
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Why don't u attach the setup page of the phone ?
Giordano
-Messaggio originale-
Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di FaberK
Inviato: giovedì 13 ottobre 2005 17.56
A: Asterisk Users Mailing List - Non-Commercial Discussion
Oggetto: Re: [Asterisk-Users]
: [Asterisk-Users]
codec g723 on Via C3
try compiling with 586 and change the makefile to disable mmx codes (if
any). I remember tohave this working on a few different processors, but
forgot how I did it.
-apu
On 10/3/05, Giordano
Grandis [EMAIL PROTECTED]
wrote
Hi!
I'm using bristuff-0.2.0-RC7k with 2 hfc single bri pci card. I load the zapata
module but whrn i try to load the zaphfc module by this command (insmod
/usr/src/bristuff-0.2.0-RC7k/zaphfc/./zaphfc.o) my system crash and froze,
i have to reboot it.
This is my interrupts:
CPU0
Hi all,
im going to install asterisk with a 4 BRI (HFC
chipset) on a Celeron at 2.6 GHz
I dont 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
Hi,
just a question: anyone has never installed g729
codec on VIA motherboard with C3 processor ?
Im having problem with IPP libraries, and
Intel said that it works only on Inter processor.
Any suggestion?
Thanks
Giordano
___
Discussion'
Oggetto: RE: [Asterisk-Users]
codec g723 on Via C3
I have a VIA Samuel 2, I use the Intel
primitives (g729)
setting the Makefile to a 586 processor.
Maybe you can test with this.
Regards.
Jsalas.
-Mensaje original-
De: Giordano Grandis
. At least I tried and never could get
it working. It's a semiactive.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Giordano Grandis
Sent: Friday, September 30, 2005
6:59 AM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: [Asterisk-Users] Diva
Hi all
Hi all,
just a question: can i use this kind of
diva for asterisk?
00:14.0 Network controller: Eicon Networks
Corporation Diva ISDN Pro 3.0 PCI
Thanks all
Giordano
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Hi all,
im tryinf to install chan_capi but i get this
error
[EMAIL PROTECTED]:/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
!
== Unregistered channel
type 'CAPI'
Sep 30 16:00:06
WARNING[8294]: loader.c:391 load_modules: Loading module chan_capi.so failed!
Thanks again!
Giordano Grandis
g.grand[EMAIL PROTECTED]
Le
informazioni contenute nella presente e-mail e nei documenti eventualmente
allegati possono essere
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
are in Europe so I can't comment
on the numbering pattern your telco expects there, but I suspect that
unknown will work fine for you here. However, it should be
explicitly set and not ignored, if only to unknown
hth
-Original Message-
From: Giordano Grandis
[mailto:[EMAIL
with snom 190; with pa168s
and ywh10 I have again some problem, the echo come up also after 1 minute of
conversation. The most strange think is that on older version of bristuff, with
same configuration files, I never had this problem.
Any suggestion? Specially for echo problem ?
Thanks all
Giordano
Well done Tim...could u post here your Zapata.conf ? :)
I'm in Italy and have some issues with echo
Thanks
Giordano Grandis
[EMAIL PROTECTED]
-Messaggio originale-
Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Tim Robinson
Inviato: lunedì 26 settembre 2005 22.30
Hi, im working about sip call pick and *8
works very fine but I pickup ringing phone on the same group. What happen if I have
more than one ringing call?
I tryied *8#exten, *8eten# but it doesnt wotk.
Is it correct? How it does work ?
Thanks
Giordano
: RE: [Asterisk-Users]
direct sip call pickup
On CVS head there is app_directed_pickup
It will let you pickup a ringing extension
directly without having to worry about pickup groups etc.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Giordano Grandis
Sent
Hi all,
Im installing two HFC pci card (both in TE
mode), I dont 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. (Im using a VIA Epia-M 1000 with Via C3
processor)
[EMAIL PROTECTED]:~#
Hi,
Im using asterisk 1.0.6 and I would let media path be connected directly between the
phones without going through Asterisk. I have to it with an AtCom320 (with
pa168s chip).
I just saw and tryied to
do what this page
Hi all,
I just connected 4 * box (by IAX) and now i'm thinking about this: can i
exchange the extensions list between this boxs ? The clinets/phones can known
which other clients are connected ?
Thanks,
Gio
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Asterisk-Users mailing list
8cdd == Spawn extension
(local, 6000, 2) exited non-zero on 'SIP/2391-8cdd'
Any ideas
?
Thanks
Giordano
Da:
Giordano Grandis Inviato:
mercoledì 29 giugno 2005 19.27A:
asterisk-users@lists.digium.comOggetto:
Hi, I installed mpg123 v0.59r
without error and defined as defaut folder /var
Users Mailing List - Non-Commercial Discussion
Oggetto: Re: [Asterisk-Users] Music oh hold
On Wed, 2005-06-29 at 19:35 +0200, Giordano Grandis wrote:
Sorry, i also tried this:
exten = 6000,1,Answer
exten = 6000,2,MusicOnHold(default)
and i got this result:
*CLI -- Executing Answer
n new stackJun 29
19:33:47 WARNING[1616]: res_musiconhold.c:354 moh0_exec: Unable to start music
on hold (class 'default') on channel SIP/2391-8cdd == Spawn extension
(local, 6000, 2) exited non-zero on 'SIP/2391-8cdd'
Any ideas
?
Thanks
Giordano
Da:
Giordano Grandis Inviato:
mercoled
Hi, I
installed mpg123 v0.59r without error and defined as defaut folder
/var/lib/asterisk/mohmp3. When i set a call on hold everythinghs seem ok, but i cannot hear music. I'm using asterisk
1.0.8
*CLI -- Executing Dial("SIP/2339-4da6",
"SIP/2391|60|Thtr") in new stack -- Called
2391 --
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