Displaying 20 results from an estimated 300 matches similar to: "zapras - and kernel ??"
2004 May 31
1
zapras how to
hi!
I'm trying to get zapras working in GSM csd network. Whenever a dialup call is initiated from the mobile to the * gateway the following appears in the log and zapras terminates. Phone gives the error dialup not answered.
==> /var/log/messages <==
pppd[2310]: Plugin zaptel.so loaded.
pppd[2310]: Zaptel Plugin Initialized
pppd[2310]: Using zaptel device 'stdin'
pppd[2310]:
2007 May 04
0
Error compiling patched pppd for zapras
hi everybody,
i'm tryint to install a asterisk system which acts as a dialin server
using a Digium Wildcard 205P.
acording to http://www.voip-info.org/wiki/view/Asterisk+cmd+ZapRAS i
need a patched version of pppd, but it does not compile on my system.
Linux box 2.6.17-gentoo-r8 #1 SMP Tue Sep 26 13:17:23 CEST 2006 x86_64
AMD Athlon(tm) 64 Processor 3200+ GNU/Linux
gcc -4.1.1, glibc-2.4
2006 Nov 30
0
Problem with ZapRAS and asterisk
Hi,
I am trying to use Asterisk cmd ZapRAS
(http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ZapRAS),
I pathed the ppp daemon ftp://ftp.digium.com/pub/zaptel/misc/, but when I
try to use it, I obtain the following error:
Connected to Asterisk 1.2.4 currently running on TSU-R1 (pid = 7242)
Verbosity was 0 and is now 44
-- Accepting call from '123456789' to
2005 Jun 22
5
ZapRAS
I'm trying to use ZapRAS to enable ppp connection through my E1.
After the ZapRAS command is executed, all sound is crappy on all lines!
The only solution is to reboot the machine (or halt it, and then power
it on since Digium's hardware doesn't like reboots).
Anyone know how this can happen?!
I'm using * 1.0.6 on Dell PowerEdge 1850 which are told (too late
though) not to work
2010 Jan 04
1
ZapRAS priviledge error
Hi,
I'm trying to get ZapRAS working but not getting very far..
Asterisk CLI shows:
WARNING[3355]: app_zapras.c:173 run_ras: wait4 returned -1: No child processes
and /var/log/messages shows:
using the plugin option requires root privilege
Can anyone shed any light on this and any fix? Googling the error doesn't find much..
I'm not sure what 'plugin' it is talking about,
2003 Oct 30
1
ZapRAS docs needed...
hi all
Where can I find documentation about how to setup ZapRAS?
What I want to do (optimally) is to allow for automatic dial-up to
external sites, each having an ISDN router. Today we use a small ISDN
router for this, but it'd be a lot better, IMHO, to have asterisk do
this (functioning as a ISDN router), as we may cancel our BRIs then.
Is this possible? And if so, how can I do it? I
2004 May 24
0
ZapRas problems
Hi
I try to use zapras. I am using zaptel-bri-0.0.2
I compiled and patched pppd from ftp://ftp.digium.com/pub/zaptel/misc/
pppd is /usr/sbin/pppd
Any idea whats going wrong here?
Thomas
-- Accepting call from '95' to '8526' on channel 1, span 1
-- AGI Script nuller.agi completed, returning 0
-- Executing ZapRAS("Zap/1-1",
2003 Apr 22
0
Re: [Asterisk] Kernel panic, ZapRAS & E400p
[ZapRAS triggering a kernel panic]
>> Kernel panic: Aiee, killing interrupt handler!
>> In interrupt handler - not syncing
>> HDLC Receiver overrun on Channel Tor2/0/2/25 (master: Tor2/0/2/25)
>
> Hrm, I haven't seen this before. Please contact me off-list and I'll
> give you more debugging instructions that may be helpful, as well as
> enquire additionally
2005 May 30
1
Serious ZapRAS problem!
Hi!
I've been trying to get ZapRAS or PPPD to work. Neither does!
All i get is LCP: timeout sending Config-Requests
But after trying, all voicelines get crazy! It sounds like robots when
somebody calls!
And since the zaptel drivers can't unload (the server hangs totaly if I
try!), I have to reboot the whole server!
The robot-voice is only on our side, it sounds fine at the other end.
2003 Nov 13
2
IAX trunk monitoring
I have an issue where * tries to route a call over IAX to another server
even if the server is down. I have included the relevant entries from
my iax.conf, extensions.conf, and some debug output. If someone could
tell me what I have configured incorrectly, I would appreciate it.
Thanks,
Stephen
-----------iax.conf on voip2----------
[voip1]
type=friend
username=voip1
host=x.x.x.x (ip
2006 Dec 13
0
Help with voicemail
I'm looking to use * for a HQ/branch office topology with fairly few calls
over the WAN. The questions I have all pertain to the following
architectural pic: http://www.45891.com/misc/arch.jpg
I'm looking at a distributed architecture so users are somewhat functional
when the link to HQ is down, with a centralized voicemail server to allow
for transfer of voicemail messages from user to
2006 Oct 27
1
Iax bug ?
Hello,
I'm french, so excuse my poor English.
I'm face to a terrible thing, with has stole a lot of my time.
On the .184 machine, I've the following iax.conf :
[general]
rtcachefriends=yes
bandwidth=high
tos=reliability
jitterbuffer=no
autokill=yes
#include "iax.voip1.conf"
#include "iax.renoir.conf"
The iax.voip1.conf file contains :
[VOIP1]
type=friend
2008 Apr 02
1
show uptime and last reload
Hi,
I just upgraded from 1.2 to 1.4.
In 1.2, when I did a "show uptime" I used to see a
second line telling me the time since the last reload.
Has this been removed in 1.4?
The following is the output of my two test boxes:
Connected to Asterisk 1.4.18.1 currently running on
voip2 (pid = 10605)
Verbosity is at least 3
voip2*CLI> show uptime
System uptime: 15 hours, 55 seconds
2005 Jun 28
0
Asterisk dies with Meetme
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hi List
I'm trying to create a conference room using H323 channels.
If i start asterisk normally (service asterisk restart) and connect to
cli using -vvvvvvvr options, when a user enters the Conference,
asterisk says "You are the only ..." and then dies, withou any error
message, nothing at all.
But, if i start asterisk with cli
2014 Feb 16
0
SIP TLS question for asterisk 11
Hi All,
I'm on a middle of an asterisk installation/configuration for my company
and I'm testing the TLS configuration.
For this reason, I used the ast_tls_cert script to build the ssl
certificates for my server.
On sip.conf file:
tlsenable=yes
tlsbindaddr=0.0.0.0
tlscertfile=/etc/asterisk/keys/asterisk.pem
tlscafile=/etc/asterisk/keys/ca.crt
tlscipher=ALL
tlsclientmethod=tlsv1
and on
2004 Apr 26
0
Help with connecting 2 servers via iax
I have successfully configured two servers and I am now trying to connect
via iax. When I attempt to call from one ext, 2006(server viop1) to
extension 3006 (server voip2) I receive a timeout or "call failed 403
forbidden.
The information I am receiving from the console is below.
Apr 26 10:53:32 WARNING[311313]: channel.c:1745 ast_request: No channel type
registered for 'IAX'
2004 Jul 17
1
MYSQL_FRIENDS and IAX problem
Hi,
I had compiled support for MYSQL_FRIENDS and it works for SIP, but when use
tiwh IAX2 I have some problem,
I can register with a client, but when I try to make a call I got this
error:
Jul 17 12:52:03 NOTICE[229387]: chan_iax2.c:5183 socket_read: Rejected
connect attempt from <IP-ADRRESS>
When I google'ed this problem I can see other users also found this error
(bug ?) But no-one
2007 Feb 12
4
Zaptel install...
I am having trouble getting Asterisk to compile the zaptel stuff.
Here are the specifics:
Linux Kernel 2.5.9-42.0.8.EL
Asterisk 1.4.0
I compiled libpri, zaptel, asterisk and asterisk-addons (in that
order). This is a fresh install of CentOS. Following the CentOS
install, I did "yum -y update" until there were no updates left.
Here is my src directory:
drwxr-xr-x 24 root root
2011 Nov 17
0
2 same sip extension number on 2 asterisk - call not passing on certain condition
Hi list,
something crazy here. 2 asterisk on 2 different place (1.4 and 1.8) both
having an extension [115], one as type peer (caller side 1.4) and one as
friend (callee side 1.8). Phones from both location connect to Asterisk
from LAN. Router are Linux boxes.
Connection between the 2 sites is done like this:
On the callee side
[115] ;callee
type=friend
host=dynamic
secret=otherSecret
2005 Oct 05
0
Unwieldy outbound macro
I have the following pair of macros defined to handle outbound calls from *.
Rather than specifying full dialstrings in the main body of extensions.conf,
outbound dial commands are made using a macro call as follows:
Macro
(outbound,number_to_dial,callerid_to_present,gateway1,gateway2,gateway3,gate
way4)
The final gateway defined is nearly always a fallback to PSTN if none of the
IAX or SIP