Displaying 20 results from an estimated 1000 matches similar to: "Asterisk crashes with segmentation fault on using many OH323 calls"
2003 Jun 10
3
s extension don't work on TDM40B
Hi all,
i have read in the * whitepaper the following:
"s: The "start" extension. A call which does not have digits associated with
it (for
example, a loopstart analog line) begins at the "s" extension."
I think this means the s extension will be execute when the phone is picked
up.
But when i pick up the phone the s extension will be never executed.
Whats wrong
2003 Oct 27
1
get IP Address from caller using oh323
Hi all (Michael),
how it is possible to get the ip address of the calling party ?
(i know by using h323... but there're a few unknown segfaults...) and so i
want to use oh323, but i have to get the ip from the caller to permit or
deny the call with AGI.
Is it possible at all ?
Thanks,
Thomas.
*******************************************
beroNet technologies GmbH
Dipl.- Ing. Thomas H?ger
2003 Jun 20
1
where to get adsi phones in europe ?
Hi all,
have anybody an idea where to get adsi phones in europe ?
Thanks,
Thomas.
*******************************************
beroNet technologies GmbH
Dipl.- Ing. Thomas H?ger
Potsdamer Str. 18 A
14513 Teltow
FON: +49 (0) 3328 3077731
FAX: +49 (0) 3328 334779
Email: thomas.haeger@beronet.com
*******************************************
2003 May 26
1
Bug in PGSQL
Hi all,
i use the PGSQL App, and i have found out, if you use a "(" or a ")" in your
query the query crashes...
My sample query was :
SELECT count(*) from tbltest where fldtest='xxx'
can somebody fix this ...??
Regards,
Thomas.
*******************************************
beroNet technologies GmbH
Dipl.- Ing. Thomas H?ger
Potsdamer Str. 18 A
14513 Teltow
FON:
2003 Jun 13
1
Problem with outgoing spool...
Hi all,
i 've written a little Callgen script for generating calls through the
outgoing spool directory.
The calls goes over 8 ttyI devices to another pbx and come in through other
8 ttyI devices.
But when i generate the calls, sometimes * register the calls but never
initiate them.
Especially when the files come to fast into the outgoing dir.
What can be wrong ?
Is it possible that the
2003 Sep 16
1
h323 gatekeeper registration failed
Hi all,
i have tried to connect to a clarent gatekeeper.
I have used both of h323 drivers chan_h323.so and chan_oh323.so.
But no one can register to this gatekeeper.
Our ip is activated on this gatekeeper.
Maybe, i do wrong anything....
I have only set the "gatekeeper" option in the h323.conf or oh323.conf to
the ip address from the gatekeeper.
gatekeeper=x.x.x.x
But no one of the
2003 Sep 19
1
codec probs wit g723.1
Hi all,
i don't know how often someone ask for this, but i ask agian:
Is it possible to use G723.1 with * or not ?
I tried to use G723.1 from * over OH323 to a gatekeeper from my provider.
The situation is following:
Zap/analog ---> IAX -----INTERNET-----IAX--->OH323---->GATEKEEPER/PROVIDER
The provider supports G723.1.
Can someone help me ?
Regards,
Thomas.
2003 Sep 23
4
Dial over IAX ahngs up after 3 rings
Hi all,
can somebody explain this ?
Thanks,
Thomas.
*******************************************
beroNet technologies GmbH
Dipl.- Ing. Thomas H?ger
Potsdamer Str. 18 A
14513 Teltow
FON: +49 (0) 3328 3077731
FAX: +49 (0) 3328 334779
Email: thomas.haeger@beronet.com
*******************************************
2003 Jun 24
1
"NoOp" gives an ringing indication ?
Hi all,
i want lock Zap channels via global var FREE1
if FREE1 = 1 then call should go on with nothing and waiting for digits to
go in _X.
Otherwise hangup the channel
But if the GotoIf goes to s|4 (NoOp) then comes a ringing indication.... !?
The "immediate" property in the zapat.conf is "yes"
[tel1]
exten => s,1,GotoIf($[${FREE1} = 1]?s|4:s|2)
exten =>
2003 Sep 01
1
some pri questions...
Hi all,
i have a few questions about PRI/ISDN:
1. Are "supplementary services" like conferencing, call brokering or call
forwarding supported by * ?
2. Is there a way to switch calls "transparent" through * from one port to
another port ?
3. Is it possible to configure the * so that * detecting dtmf during a call
?
Thanks for answering questions, regards,
Thomas. :-)
2003 Sep 22
2
how to dial a h323 destination ?
Hi all,
i have big problems to make a h323 call over the gatekeeper from my
provider.
The provider demanded following account data:
H323 ID: XXX-XXX-XX-X
DetinationNumer: XXXXXXXXXXX
I have configured the oh323.conf following:
gatekeeper=XX.XX.XXX.XXX
alias=XXX-XXX-XX-X
Isx the alias equal to the h323id ?
And how i have to make a call with the dial app ?
I have following config:
exten
2003 Oct 13
1
PRI/E1: machine freeze/dies after a few calls
Hi all,
inside my * is a E400P. The machine is a PII 400Mhz with 256MB Ram. OS is
Debian woody. * is the newest cvs co.
I have written a little callgen script which make outgoing calls through my
*:
#! /bin/sh
set -e
n=$1 # Nummer
anz=$2 # Anzhal der Versuche
anz2=$3 # Kan?le
sle=$4 # Timeout bis zum n?chsten Versuch
if [ -z $4 ]; then
sle=0
fi
s=1
2003 Jun 12
1
Callerid Modem I4l and outgoing spool
Hi all,
i tried to write a call generator script. It generate i file like
sample.call in ast src tree.
But when i set the callerid and make a call over Modem[i4l] then the caller
id is not set for the outgoing call.
Is it impossible to set the callerid at this time for ttI devices?
Thanks for help,
Thomas.
*******************************************
beroNet technologies GmbH
Dipl.- Ing.
2003 May 26
1
Quetsion about DISA...
Hi all,
i use the DISA app for giving the user a trunk after a authentication
through PGSQL as follows
.... auth via PGSQL
exten => s,1,DISA,no-password|test
I think the user is now in context "test" and he could dial any number if
the extension-conf in "test" is for example
exten s,1,Dial,OH323/<myip>
But if the user dial one digit the call build up
2003 Aug 29
1
additional digit in front of the dialed extenesion by outgoing pri/E1 call
Hi all,
i have configured incoming voip traffic as follows:
[voipin]
exten => _X.,1,SetCallerID(033283077734)
exten => _X.,2,Dial,Zap/g4/${EXTEN}
exten => _X.,3,Hangup
If the call going out the pri dials with an additional '0' before the dialed
number.
This is for caller number AND called number. But i can't see anything that
says set a '0' more in front of the
2003 Sep 01
0
Problem with SIP: Maximum retries exceeded
Hi all,
this message occurs if i was connected or not:
WARNING[213006]: File chan_sip.c, Line 432 (retrans_pkt): Maximum retries
exceeded on call 0b03e0c6189a769b54e49eb471f32454@172.20.23.150 for seqno
102 (Response)
If i was connected, the call will be disconnected after a few seconds.
What does it means ? I don't see anything to configure like Max retries....
Thanks for help,
Thomas.
2003 Sep 05
0
IAX sound probs
Hi all together,
i have following configuration:
ISDN Phone ---> ASTERISK1/PRI ---> ASTERISK1/IAX ---> INTERNET --->INTERNET
ROUTER (Port 5036 nat) ---> ASTERISK2/FXO/ANALOG DEV
The call flows fine, but no sound will be transfered.
On ASTERISK1 a message like "stopped sounds" occurs.....
What' s wrong? Is there another port wich i have to nat ?
Regards, thanks
2003 Sep 18
0
no ring tone analog Zap --> I4L
Hi all,
i have noticed that i can't hear a ring tone if i make a call from my TDM40B
to a chan_modem_i4l endpoint.
I had a look in the code from chan_modem_i4l and there is a function calling
"i4l_handle_escape" that gives a AST_CONTROL_RINGING frame back. But this
seems not work ...(or i4l is not signaling it ?)
Til now i have used the Dail app like
Dial, Zap/g1:XXXXXX|60|r
so it
2003 Sep 19
0
ringing tone on analog Zap channel question
Hi all,
can somebody explain me why i can't hear a ringing tone (alerting) if i'am
going to connect to my destination end point?
Is it basically so that i have to configure like:
exten => xxx,1,Dial,ChanTec/number|timout|r
Is it really nessesary to use the "r" option everytime if i want to indicate
a ringing tone? This suggest a wrong call flow for the user ...
Thanks for
2003 Sep 29
0
How to prevent echo ?
Hi all,
i have following scenario:
____________* 1____
______________* 2______
| | |
|
analog/Zap --> IAX2 ----> DSL ---> INTERNET ---> Backbone/100Mbit ---->
IAX2 ---> Zap/pri(E400P) ----> PSTN
And, if i make a call from *1 over *2 to PSTN, i can hear an echo in my
analog phone,
even though