Hi all,
Anyone has seen/heard of a set of french
prompts
for a calling-card application such as/similar
toastcc ?
Thanks,
Fred
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Walid,
Check the ASTCC agi script ; it just does exactly
that:
http://www.voip-info.org/wiki-ASTCC
Cheers,
Fred
- Original Message -
From:
Walid Azab
To: asterisk-users@lists.digium.com
Sent: Thursday, June 15, 2006 11:14
Subject: [Asterisk-Users] AGI to read
Hi Douglas,
Try this:
open(STDERR, /etc/asterisk/agi-bin/errors.txt)
Fred
- Original Message -
From: Douglas Garstang [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, June 12, 2006 11:32
Subject:
12, 2006 11:52
Subject: RE: [Asterisk-Users] AGI Stderr
Oh yeah, I also won't get time/date stamps if I redirect stderr to a file
like that
-Original Message-
From: Douglas Garstang
Sent: Monday, June 12, 2006 8:51 AM
To: 'Frederic Jean'
Subject: RE: [Asterisk-Users] AGI
Hello all, and especially Steve,
It seems my libunicall installation is having a
little problem when initializing.
Should I play with these ?
#define
DEFAULT_BLOCKING_RELEASE_TIME
450#define
DEFAULT_ANSWER_GUARD_TIME
100#define
DEFAULT_RELEASE_GUARD_TIME
20#define
DEFAULT_T1
Hi, downloaded the latest snapshot just right now at soft-switch for
MFC/R2 support and I get this message when trying to
do libmfcr2; any idea ? it looks like he's not reaching Unicall.
Thanks,
Fred
# make
make all-am
make[1]: Entering directory `/usr/src/libmfcr2-0.0.3'
if /bin/sh
You can do it adding a parameter to the ${EXTEN}:
exten = _91NXXNXX,1,AGI(call_log.agi,${EXTEN:1})
exten = _91NXXNXX,2,Dial(${SIPtrunk}/${EXTEN:1},55,o)
:1 would strip the first digit.
Fred
- Original Message -
From: Erick Perez [EMAIL PROTECTED]
To: Asterisk Users Mailing
Ok,
I got rtptimeout setup to 60 in sip.conf and it go better ; came back to
normal
as soon as I put it.
If anybody knows if rtpkeepalive and rtptimeout can work in conjunction,
please share your toughts !
Thanks,
Fred
- Original Message -
From: Frederic Jean
To: Asterisk Users
Hi all,
Simple question;
What are the possible reasons for a SIP channel to hang there for hours
after
the call has terminated ?
Calls are sent from 1.2.6 to SER using DeadAGI over the internet, and
yesterday there
were 100 calls that hanged for hours over a total of 15k calls.
Typical
Hi everybody,
I am almost there on that one :-) Transfering a DID from SER to
Asterisk 1.2.6, but I get 403 forbidden. I tried this example
but without success and I also looked at last year's posts..
http://www.cs.colostate.edu/~somlo/CSU-SIP-notes/#3.3
SER is the public access and is on a
Good morning list !
I have an Intel P4 775 D915GAG-L motherboard with just
one CPU (3.2ghz 640) and I tried to install the latest zaptel using
Mandrake 10.1 (i586) but the udev devices are not being
created; it usually works for me on lower ends machines
so I was wondering if my distro is
Hi Geoff,
You might want to try tcdump, specifying the source
and destination IP (to minimize the info)
and see where are the RTP packets going ;
youwill see if they change port or
something like that
after a while.
Cheers,
Frederic
- Original Message -
From:
Geoff
Hello all,
Is there a way to check who is using the VAD option whenever
we get the message VAD frame at the end at the CLI ?
--- The IP is not listed.
Some people use it but I can't tell them to turn it off for better audio
performance,
and I know it generates a lot of messages on the
Josue, bom dia !
Can you provide us with the iax.conf for each
server ?
Cheers,
Frederic Jean
- Original Message -
From:
Josué
Conti
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Tuesday, March 28, 2006 08:48
Subject: [Asterisk-Users] IAX
Hi,
Can someone explain to me how to set up the sip_buddies
table from 1.2.5 properly so my users can authenticate correctly
without using the name field ? (if it's possible)
First I was assuming that it would be possible for a user
to connect and dial just providing username,secret,host and
Hello,
I use Mandrake 10.1 and I had no problem, just
had to install the kernel sources and follow the
instructions in README.udev
I do make linux26.
you have to reboot after you follow the instructions in REAME.udev
so it can take effect. Make sure zapata.conf is in /etc and check
for
Hello,
I upgraded from 1.0.9 to 1.2.5 yesterday and it went ok except
for one little change that happened in the CDRs, and it's concerning the
CallerID.
Let me explain..
With 1.0.9 I used to get this in the CDRs: COMPANY LTDA 2153
Now with 1.2.5 I get one part only: 2153
The number 2153
Hello,
I installed 1.2.5 and realtime SIP. The connection
to the DB is OK
because I can get the values from the
CLI.
Here are my 3 different cases:
1- If I put an unexisting user, I get 404 and I am
not able to dial.
2- If I check "Disable registration" within Firefly
itdoes
is
below.
Thanks.
Frederic
- Original Message -
From: Frederic
Jean
To: asterisk-users@lists.digium.com
Sent: Tuesday, March 21, 2006 14:21
Subject: SIP Realtime 1.2.5 and Username/auth name mismatch
?
Hello,
I installed 1.2.5 and realtime SIP. The connection
to the DB is OK
]
Sent: Tuesday, March 21, 2006 4:06 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Fw: anybody has SIP realtime working ?
Message: 16
Date: Tue, 21 Mar 2006 18:51:29 -0300
From: Frederic Jean [EMAIL PROTECTED]
Subject: [Asterisk-Users] Fw: anybody has SIP realtime working
Good morning everybody,
Can someone explain to me the interconnection
between
thesefour things: indications.conf,
SetLanguage(), zaptel.conf
and ring-back ? If
there is any !! :- )
I am having this case where some users cannot hear
ring back
from a DeadAGI script and it seems to be
Hello everybody,
I have this problem where I can't get a ring tone when
SIP devices dial an IAX2 route. I get the ring tone
using IAX2 devices to dial the same route. The call
completes normally in both cases...
Facts:
- Asterisk 1.0.9
- The Dial command is within an AGI.
- ATA (grandstream)
Hi everybody,
I sent an e-mail this morning regarding SIP / IAX2
with no ring-back, I now succeeded to pin-point the
problem, here it is, if I dial a provider directly from
extensions.conf I get ring-back, if I dial from an AGI
script I don't get the ring-back but it calls anyway.
I use 1.0.9.
Thanks Eric,
I'll try that, but I am using ztdummy, will it work ?
Thanks a lot for your attention,
Frederic
- Original Message -
From: Eric ManxPower Wieling [EMAIL PROTECTED]
To: Frederic Jean [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion asterisk-users
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