Displaying 20 results from an estimated 800 matches similar to: "Incoming call doesn't finish when internal phone hangs up"
2003 Sep 25
6
E1 in Brazil
Hey all!
I had an experience trying to set up an E1 in Brazil which could help
somebody. In Brazil is very common telcos to have just R2 digital as their
primary signaling. As I were trying to set up an E100P, which does not
support R2 yet, I had to test an other signaling which works perfectly with
Asterisk.
They call this signaling as RDSI, using ccs as framing and PA (primary
access) as
2004 Sep 21
1
RDSI vs Analogic
Hi. I'm getting new lines for using with Asterisk. In my Telco they said
I could choose between Analogic lines and RDSI lines... I've already
bought a TDM400P with FXO modules. Can you give some hints on the
differences between RDSI and normal Analogic lines? Would I have
problems for using a RDSI line with the TDM? Any other issue in general?
Thanks in advance,
RODOLFO
---
avast!
2005 Mar 10
2
QuadBRI ,TDM400 and SuSE9.2
Hi all,
We need help with our SuSe9.2 asterisk box
We have one QuadBRI and one TDM40B in an ASUS pundit R-2 barebone.
We have downloaded the bristuff (0.2.0-RC7j) and installed it without
problems.
once we downloaded and compiled asterisk, zaptel and all other stuff,
the module installation succed in this order:
modprobe zaptel
modprobe qozap
modprobe wcfsx
then the ztcfg output this:
Zaptel
2008 Nov 12
1
How to get correct dial result for outgoing calls thru ISDN?
Hi everyone,
Currectly I'm having some troubles to get correct status of my calls throug
ISDN lines, when outbound calls don't get its destination I always receive
NO ANSWER as ${DIALSTATUS} despite the fact I know the target number
doesn't exists or is busy at that time.
Maybe there is something I must change in my zaptel.conf or zapata.conf,
current configs follows:
####
2005 Mar 11
1
QuadBRI ,TDM400 and SuSE9.2 (Sencond try)
Hi all, this time with the complete configuration files...
We need help with our SuSe9.2 asterisk box
We have one QuadBRI and one TDM40B in an ASUS pundit R-2 barebone.
We have downloaded the bristuff (0.2.0-RC7j) and installed it without
problems.
once we downloaded and compiled asterisk, zaptel and all other stuff,
the module installation succed in this order:
modprobe zaptel
modprobe qozap
2000 May 09
1
Windows 2000 crashes
We used samba 2.0.6 under Redhat 6.1 with W98/NT4 clients without any
problems. Now we upgraded one PC with Windows2000. In the beginning everything
worked fine. But suddenly there are problems: Browsing a specific subdirectory
of a share on the server cause a inmediate crash of W2k with a automatic
restart. I upgraded to Samba2.0.7, again W2k crashes. I connect to this
subdirectory from a
2010 May 18
1
Callerid with DAHDI
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hi all!
I'm testing a telephone connected to FXS port of a Sangoma A200 card.
But I'm observing that callerid is not working. The configuration that
I'm using in chan_dahdi.conf is the following one:
- ---------------------------------------------------------------------
;autogenerated by /usr/sbin/wancfg_dahdi do not hand edit
2004 Sep 28
1
CAPI channels
Hello all,
I`ve got an AVM c2 card instaled on my SuSE box.
I?m having problems configuring its channels.
I don?t know how to set up asterisk to use the CAPI channels. I don?t know
how to call them.
My capi.conf is as follow,
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
[interfaces]
;#########config de la primera interface CAPI##########333
2009 Dec 02
2
Help configuring Audiocodes MP-104 FXO
Hi list,
I'm trying to get ready the MP-104 FXO to use qith my box, but when I send
calls I hear only dial tone and after a few seconds I get busy signal.
I very appreciate your advices.
Command line results and SIPconfigurations follows:
*CLI>*
-- Executing [7991696900 at total:1] Playback("SIP/101-09dd8918", "beep")
in new stack
-- <SIP/101-09dd8918>
2008 Oct 14
1
SIP channels seem not to close after call is finished
Hello everyone,
I'm getting DIALSTATUS=CHANUNAVAIL when a call is trying to get one of my
queue interfaces, despite the fact it is free at that time, can you give
help?
1. I see many sip channels from that extension:
[root at mysweetpbx]# asterisk -rx "*sip show channels*" |grep 648
Peer User/ANR Call ID Seq (Tx/Rx)
Format Hold
2007 Jun 14
0
b410p
Hello, I'm trying to set up a b410p rdsi card, and I'm having problems
getting it up.
I followed the instruction on asteriskguru and everything seem to be fine
but all leds on the card are in red.
[root@rdsipbx ~]# uname -a
Linux rdsipbx 2.6.15.7 #2 Tue Jun 5 16:37:07 CEST 2007 i686 i686 i386
GNU/Linux
[root@rdsipbx ~]# dmesg |grep Digium
HFC-multi: card manufacturer: 'Cologne Chip
2005 Feb 09
3
ISDN in Spain
Hi list!
Sorry for this slightly off-topic message but does anybody know if the
standard for ISDN BRI is the same in Spain as it is in the rest of Europe
(or the Netherlands).
Will a standard HFC-S card work?
2007 Jun 28
2
E1 not coming up
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hello List,
since some days i run into the problem that one span on a TE407P is not
comming up correctly. With intense debug on that span i get:
< [ 02 01 7f ]
< Unnumbered frame:
< SAPI: 00 C/R: 1 EA: 0
< TEI: 000 EA: 1
< M3: 3 P/F: 1 M2: 3 11: 3 [ SABME (set asynchronous balanced mode
extended) ]
< 0 bytes of data
2011 Feb 17
1
Setting two E1 cards
Dear, I always had one E1 card with one span, so I've never had any
problem in set it up through /etc/dahdi/sustem.conf and
/etc/asterisk/chan_dahdi.conf because I put span=1.
But now I have a PBX with two E1 cards with 4 span (8 span in total).
How do I have to define both card in system.conf and chan_dahdi.conf,
and how do I have to refer each span to the corresponding card ???
Thanks a
2004 Sep 10
14
Asterisk newbie questions
Hi everyone.
I'm a bit of a Linux newbie, but I've been doing tech stuff for ages.
I'm also brand new to *.
I've been reading the Voip.org wiki, and perusing the list archives for a
while since I've been asked to investigate using IP telephone / soft phones
for a call-center type scenario. People (marketing folks) have pointed me
at Cisco, but I really don't wanna.
2005 Mar 03
0
problem registering a bt100 with 1.0.5.11 firmware
hi all
I can not register my new granstream bt100 phone with asterisk, i have old of they working perfectly but they have an older firmware(1.0.5.3).
any bady now where i can read about this or now what i have to do???
My sip.conf:
[10]
type=friend
context=unr
username=10
callerid=10
usecallerid=yes
hidecallerid=no
canreinvite=yes
host=dynamic
dtmfmode=info
nat=no
mailbox=10
callgroup=1
2007 Feb 09
1
Detect hang-up
I've got an X100P that doesn't seem to be detecting hang-ups. I'm not sure
what it's supposed to do, but I wouldn't expect it to continue processing
the dial plan.
Any pointers? Documentation locations that address hanging up would greatly
appreciated!
TIA!!
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network Engineer Safe Data, Inc.
(910) 285-7200 david@safedatausa.com
2011 Oct 11
1
high and lowest with names
Hello,
I'm looking to get the values, row names and column names of the largest and
smallest values in a matrix.
Example (except is does not include the names):
> x <- swiss$Education[1:25]
> dat = matrix(x,5,5)
> colnames(dat) = c('a','b','c','d','c')
> rownames(dat) = c('z','y','x','w','v')
>
2010 Jul 19
1
Problem with E1
Hi All,
I am facing problem with E1 line. I have installed Asterisk
(1.4.20.1) on a system with Digium TE420 card (Zaptel- 1.4.10)
But every
now and then I face problem of down E1's. The log show lot of entries like
"pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of
span 2"
This happens on a regular basis and the E1 becomes up after some
time.
My
2004 Jan 21
3
[Bug 792] mtu and NAT wrong solution
http://bugzilla.mindrot.org/show_bug.cgi?id=792
Summary: mtu and NAT wrong solution
Product: Portable OpenSSH
Version: -current
Platform: All
OS/Version: Linux
Status: NEW
Severity: major
Priority: P4
Component: Miscellaneous
AssignedTo: openssh-bugs at mindrot.org
ReportedBy: