Displaying 20 results from an estimated 700 matches similar to: "Asterisk crashes"
2002 Jul 11
3
Printing from W2K clients
Hi,
I have Slackware 8 Linux Box with Samba-2.2.5 and HP LJ 1200 printer shared by
samba (with LPRng).
The problemm is: when printing from W2K clients users cannot change
print options (like portrait/landscape page orientation, number of
copies etc). When printing from Win98 clients all is ok.
Could someone help vt with this problemm?
--
Sincerely,
Elman Efendiyev
elman@megacom.com.ua
2007 May 07
2
h323 problem with asterisk 1.2.18
i am experiencing problem with asterisk 1.2.18
I've downloaded and installed pwlib and openh323 with the following commands:
cd /path/to/pwlib
./configure
make clean opt
cd /path/to/openh323
./configure
make clean opt
then 'ive set the corresponding PATH
PWLIBDIR=/data/programmi/asterisk_1.2.18/pwlib_v1_10_0/
export PWLIBDIR
OPENH323DIR=/data/programmi/asterisk_1.2.18/openh323_v1_18_0/
2004 Sep 06
1
T.38 "pass-thru"
Hello,
As I understand * don't supports T.38 in Zap channels (please correct me
if I'm wrong, BTW is there plans for such support?)
I believe it's should support T.38 in "pass-thru" mode. I mean setup
like this:
Hardware gate with T.38 ------ Asterisk ------ Hardware gate with T.38
But I had troubles with this setup (no faxing) while two gates conneted
directly with same
2009 Jan 12
1
problem with dahdi and meetme
Hi to all.
I'm trying to use meetme on asterisk 1.4.22.1.
On a debian i've compiled (as i need h323 support)
openh323_v1_18_0
pwlib_v1_10_0
dahdi-linux-2.1.0.3
dahdi-tools-2.1.0.2
asterisk-1.4.22.1
All works fine, dahdi status is:
asterik:/data/programmi# /etc/init.d/dahdi status
### Span 1: DAHDI_DUMMY/1 "DAHDI_DUMMY/1 (source: RTC) 1" (MASTER)
asterik:/data/programmi#
2004 Sep 13
4
Unknown RTP codec 72 received
Hi all,
I get "Unknown RTP codec 72 received" message in console when call in
progress, but all seems OK. Calls is from X-Lite SIP softphone to PSTN
over voicepulse connect (IAX) and to FWD echo test (SIP). But this
message only with one SIP client, others (X-Lite too) not giving this
message. All X-Lite settings are identical. Asterisk is last cvs version
This what I see in console
2004 Jul 12
0
IP Soft Phone with FAX
Hi,
I need to send and receive faxes over VoIP in realtime.
I mean: user ? calls from VoIP network to fax machine on PSTN, but
starts voice conversation with user B on that fax machine. Then users
agree to send a fax (any direction), pressed "start", completed fax
transmission and then continue a voice conversation.
This is one of generic ways to use analog fax machine.
As I understand
2004 Jul 24
0
PBX functions and different channels grouping
Hi All,
I need to replace old analog PBX with Asteriskl and X-Lise SIP
SoftPhones as client phones.
First: I have problems with implementation of PBX functions. I need and
unsuccesfully tried theese functions (took info at
http://voip-info.org/wiki-Asterisk+PBX+functions)
Call Pickup: Supported in the standard installation (*8 - defined in
res_parking.c +54)
- Just don't understand how to
2004 Jul 28
0
D-Link DG-104SH H323 problemm
Hi,
I'm using D-Link DG-104SH (H323 4 port FXS gateway) with analig phone
connected to it and X-Lite softphone as endpoints with *
When I calling from X-Lite to analog phone it's ok
When I dilaling X-Lite from analog phone, X-Lite si ringind but whei I
picked up X-Lite connection drops
IP of DG-104SH is 192.168.1.3, H323 ID is GW1
X-Lite number is 233
Here is * output:
-- Executing
2007 Apr 27
1
SIP<->H323 calls without proxying RTP
Hello,
Could somebody tell me is it possible to use asterisk without RTP proxying
in SIP<->H323 calls?
I mean exactly what canreinvite=yes option do in SIP<->SIP calls.
I don't need a transcoding, only a signaling conversion, and this is
possible with some softswitches, so i wondering what about asterisk.
Same question about H323<->H323 calls
I'm using NuFone
2004 Sep 20
2
H.323 call problemm (no sound)
Hi all,
I'm having trouble with H.323 outbound calls, * connects but there is no
sound in both ways.
I'm using X-Lite as SIP client with GSM codec and dialing to ITSP (which
using cisco, I think) over H.323 with G.729 codec. I have 4 digium G.729
licenses installed and this is onli one call.
I tested my * with another ITSP over SIP and G.729 codec and there was
all ok
Here is my configs
2007 Dec 14
2
chan_h323 compilation
Hi All;
I am trying now to compile h323 to be able to use it,
I did the pwlib and openh323 successfully and I
exported the PWLIBDIR=/usr/src/pwlib_v1_10_0 and the
OPENH323DIR=/usr/src/openh323_v1_18_0, then I was need
to compile h323 as following:
cd /usr/src/asterisk-1.4/channels/h323
When I type make, it gives me:
make: Nothing to be done for 'degault'
And when I type make opt, it
2007 Feb 11
0
TE110P working hardware configurations
Helo,
I have a troubles getting to stable work of Digium TE110P card (mailed some
time earlier in the list) - I can't get 100% pseudo zap interface accuracy
(zttest), so getting HDLC aborts and call drops. I tried number
motherboards, hardware and software configs according to info in wiki, thisl
list and number of websites - no luck.
So I ask everyboby who successfully use Digium TE110P card
2008 Oct 18
1
strange h323 delay issue
Hello,
I have a strange h323 issue. After executing command
"Dial("SIP/333-0d1dfe00", "H323/361737052390920 at ccg|5|tT")" at Oct 18
22:32:23. Meanwile I have sniffing traffic on port 1720. The call was
established just at Oct 18 22:33:03 (New H.323 Connection created.) and also
packet sniffer grabs the h323 invites at this time also. So my question is
what
2004 Nov 23
1
Fax over SIP Problems (sorry for this topic ...)
Hello everyone!
I tried to send a fax over SIP with an Asterisk Server in the middle (no
Digium Cards, etc. installed, everything is SIP only, the PSTN-Gateway
is external). Whenever I start sending a Fax to a PSTN destination, the
Call gets answered and asterisk tries to build a native bridging:
-- Attempting native bridge of SIP/sip.westend.com-082fd1b8 and
SIP/xxx-3ef8
Then the following
2005 Jan 29
2
TE405P w/ Intel SE7210TP1_E Motherboard
Hello,
I'm looking at building a couple new PRI Gateway boxes using
TE405P cards, and was wondering if anyone has had any experiences (good or
bad) with the Intel SE7210TP1_E motherboards from Intel. General Technics
builds some really nice (and cost effective) 1U servers based on the
board:
Server: http://www.gtweb.net/gt637.html
Specs:
2007 Feb 16
7
Summary of "Trixbox vs. custom install"
Hello everybody. First of all thanks to all the people giving their
opinion on the subject I proposed: "Trixbox vs. custom install".
You've all been very helpful.
I try to summarize what has emerged from the various messages.
Forgive me if I miss or forget something or if I simplify too much
some of your messages...
- Elman Efendiyev says that you should install from sources if
2009 Jul 16
1
H323 situation
Hi all,
I have this installation:
Asterisk 1.6.1.1 with h323 support, pwlib_v1_10_3 and openh323_v1_18_0.
I have a problem that is, when a call comes from H323 and goes to a Sip
phone the asterisk sends two rtp streams to the sip. I checked this with
tcpdump, save the payload (voice is in G711u), one is the ringing indication
and the other is the voice coming from the user in h323 side. And
2007 Feb 07
9
Digium TE110P
Helo,
I have problem with Digium TE110P connected to CISCO 3640 (port on
NM-HDV-2E1-60) wth PRI E1 link. I use CISCO now for testing but when I
tried with real PBX problem was exactly same.
I have this messages in Asterisk conole and log sometimes:
NOTICE[1115] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel
of span 1
Usually 2-5 such messages in series, can be repeated after 10
2004 Jul 25
1
Busydetect problems
Hi guys.
I have a XP100P Clone , and the busydetect dont work for me..
PSTN---Asterisk---Sip---Asterisk----PBX
Any call from pstn side dont disconnect ... I have no disconnect supervision and busydetect dont work...
Please Help me.
Zapata.conf
[channels]
echocancel=yes
usecallerid=no
hidecallerid=no
rxgain=0.0
txgain=0.0
signalling=fxs_ks
callprogress=no
context=entrada
channel=>1
2004 Sep 17
1
How would you handle a fax without T.38orG.711uLaw?
asterisk-users-bounces@lists.digium.com wrote:
> Isn't it possible to use T.38 for interconnecting hardware gates
> supporting T.38 with asterisk using SIP REINVITE?
> I'm not shure but but think its's might be possible because after
> reinvite traffic goes directly from one gate to anotger, not over
> Asterisk
We've seen a problem here with asterisk. Wehn