Displaying 14 results from an estimated 14 matches for "skuse".
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2003 Dec 03
2
"oh323 calling party number"
...rty Number" field in an
H.323 call?
I have asterisk configured to accept a SIP call and connect it to an H.323
IVR system. The call goes through, but the caller id is put in the "Display"
field rather than the "Calling Party Number" field.
-----Original Message-----
From: Skuse, Phil [mailto:Phil.Skuse@vicorp.com]
Sent: 01 December 2003 17:23
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] How do I get caller's number in oh323 ?
We have an h.323 based IVR platform. When we make a call to it using an
h.323 phone, it can see the callers number (ANI), bu...
2003 Apr 23
4
Grandstream BudgeTone 100
...5 SIP phones on this list, I purchased a couple
for evaluation. They do work with asterisk - and are good value for money,
but as somebody commented: they are not yet perfect.
I just wondered if anybody had managed to get either the message-waiting
indicator or the conference button to work?
Phil Skuse <phil.skuse@vicorp.com>
****************************************************
Unix System Administrator, Vicorp Group Limited.
Tel +44 (0)1753 660523 Fax +44 (0)1753 660501
The Telephony Engine Company http://www.vicorp.com
****************************************************
2003 Sep 09
5
Xlite = no sound
...s the secret to getting sound through Xlite? The SIP messages all look
OK to me, but the sound isn't coming through.
It was trying to use GSM, so I searched the archive and tried:
disallow=gsm
allow=ulaw
Now it says that it's using ULAW but I still get no sound in either
direction.
Phil Skuse (MBJEJPIEUI) <phil.skuse@vicorp.com>
****************************************************
UNIX System Administrator, Vicorp Group Limited.
Tel +44 (0)1753 660523 Fax +44 (0)1753 660501
The Telephony Engine Company http://www.vicorp.com
*************************************************...
2003 Oct 13
1
oh323 inband dtmf - Possible bug?
...08.398 ThreadID=0x00022012 h323.cxx(4594) H323
SendUserInputIndicationQ931("1")
My initial examination of the code suggests that it should be calling
SendUserInputAsTone instead.
(Please don't use this thread as an excuse to start the h323 vs oh323 war
AGAIN)
Phil Skuse <phil.skuse@vicorp.com>
***************************************************
UNIX System Administrator. NIC Handle: MBJEJPIEUI
Vicorp UK Limited: The Telephony Engine Company.
Tel +44 (0)1753 660523 http://www.vicorp.com
***************************************************
2003 Oct 14
0
Has something changed with AGI recently?
...to
priority 2 and connect the call. But now, when it terminates, it starts all
over again in a continuous loop and never gets to priority 2. Do I need to
update the priority in the script, or return a value to indicate successfull
completion?
Here is the (unchanged) AGI script...
#!/bin/sh
# Phil Skuse 16/4/2003
# Writes caller information to /home/asterisk/call_log.txt
export PATH=/bin:/usr/bin
read STDIN
while [ "x$STDIN" != "x" ]; do
export VARNAME=`echo $STDIN | awk '{print $1}' | tr -d ":"`
export VARVALUE=`echo $STDIN | awk '{print $2}'`...
2003 Apr 25
3
Internet Dial-in security questions
...PSTN or voicemail? Anybody have an example of this?
Are there any particular security risks that I need to defend against?
Would it be better to put a secured asterisk server outside the firewall and
connect it to the internal one with IAX? Does this require less ports open
on the firewall?
Phil Skuse <phil.skuse@vicorp.com>
****************************************************
Unix System Administrator, Vicorp Group Limited.
Tel +44 (0)1753 660523 Fax +44 (0)1753 660501
The Telephony Engine Company http://www.vicorp.com
****************************************************
2003 Jul 17
0
Sip call question
...would do this? I'm guessing that I either need to collect the
DTMF, format it into a sip address and then somehow get asterisk to dial
that address, or perhaps I can take it from $EXTEN somehow.
Is this possible with the existing apps/scripts/macros, or do I need to
write some new ones?
Phil Skuse <phil.skuse@vicorp.com>
****************************************************
UNIX System Administrator, Vicorp Group Limited.
Tel +44 (0)1753 660523 Fax +44 (0)1753 660501
The Telephony Engine Company http://www.vicorp.com
****************************************************
2003 Dec 01
0
How do I get caller's number in oh323 ?
...through OK, but we don't get the number. How
can I make this work?
h323.conf
=======
[general]
port = 1720
bindaddr = 0.0.0.0
disallow=all
allow=alaw
dtmfmode=inband
[ivr]
type=h323
context=default
extensions.conf
===========
exten => 602,1,Dial,h323/7002@ivr
exten => 602,2,HangUp
Phil Skuse <phil.skuse@vicorp.com>
***************************************************
UNIX System Administrator. NIC Handle: MBJEJPIEUI
Vicorp UK Limited: The Telephony Engine Company.
Tel +44 (0)1753 660523 http://www.vicorp.com
***************************************************
2004 Oct 28
0
Permission denied creating Clearcase view on Samba share.
...smb_reh=0
smb_err=49152
smb_flg=136
smb_flg2=51265
smb_tid=2
smb_pid=3456
smb_uid=101
smb_mid=40512
smt_wct=0
smb_bcc=0
[2004/10/27 17:36:41, 6] lib/util_sock.c:(449)
write_socket(22,39)
[2004/10/27 17:36:41, 6] lib/util_sock.c:(452)
write_socket(22,39) wrote 39
Thanks.
Phil Skuse <phil.skuse@vicorp.com>
***************************************************
Network Administrator.
Vicorp UK Limited: The Telephony Engine Company.
Tel +44 (0)1753 660523 http://www.vicorp.com
***************************************************
2003 Apr 24
2
Anyone using Asterisks and a Quicknet Lineja ck in the UK?
I don't have any experience of your problem - but I thought this might help.
http://www.hut.fi/Misc/Electronics/circuits/uk_wiring.html
<http://www.hut.fi/Misc/Electronics/circuits/uk_wiring.html>
The UK (and some of it's former colonies) use a system called 3-wire
ringing. Some equipment from overseas requires an adaptor to make it work. I
don't know if the LineJack is one
2003 Jul 31
1
RTP codec 13 received - Cisco incompatibilit y?
I have a similar setup to you and get the same message regularly. I don't
think it's the cause of your problem. I did some research on it a while ago:
IIRC the cisco uses codec 13 for "silence suppression" whereas asterisk
(correctly) uses codec 19. The router can be configured to use 19 also, but
I didn't bother. I'm sure somebody will correct me if I'm wrong about
2003 Apr 14
2
SIP hanging
I too am having this problem reported by Frank Hoonhout. Asterisk runs fine
for a few minutes and then stops accepting new calls. (I have a standalone
server with SIP phones and I'm not doing any external registration).
Asterisk CVS-04/07/03-09:28:50
0x420e0037 in poll () from /lib/i686/libc.so.6
(gdb) info threads
16 Thread 14351 (LWP 7258) 0x420e187e in select () from
2003 Mar 07
70
unsubscribe
Gautham Kasinath
Software Engineer
Arkin Systems Pvt Ltd
T. Nagar
Chennai
Ph. (91) (44) 8216686 Extn 14
2003 Jul 24
0
the 'pound' and '#' are the same? (OT Rambli ng)
Some more unusual ones:
http://www.muppetlabs.com/~breadbox/intercal-man/tonsila.html
-----Original Message-----
From: Gary Gapinski [mailto:Gary.Gapinski@grc.nasa.gov]
Sent: 24 July 2003 14:37
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] the 'pound' and '#' are the same? (OT
Rambling)
On Thursday 24 July 2003 01:21, John Laur wrote:
> I haven't ever