Ümit AYDINLI
2006-Dec-07 07:19 UTC
[asterisk-users] -- Called 12127773456@OOH323 Segmentation fault (core dumped)
OOH323 Debugging Enabled
-- Executing Answer("SIP/3513-090f7d40", "") in new
stack
-- Executing Wait("SIP/3513-090f7d40", "1") in new stack
-- Executing DeadAGI("SIP/3513-090f7d40",
"a2billing.php|1") in new
stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php
a2billing.php|1: line:58 - IDCONFIG : 1
a2billing.php|1:
a2billing.php|1: line:67 - MODE : standard
a2billing.php|1:
-- AGI Script Executing Application: (Dial) Options: (
OOH323/12127773456@OOH323|60|HLxyz(5400000:31000:00000))
--- ooh323_request - data 12127773456@OOH323 format 0x4 (ulaw)
--- find_peer
+++ find_peer
+++ ooh323_request
--- ooh323_call- 12127773456@OOH323
+++ ooh323_call
-- Called 12127773456@OOH323
Segmentation fault (core dumped)
[root@asterisk1 ~]#
show version
Asterisk 1.2.12.1 built by root @ localhost.localdomain on a i686 running
Linux on 2006-10-18 18:35:57 UTC
; Objective System's H323 Configuration example for Asterisk
; ooh323c driver configuration
;
; [general] section defines global parameters
;
; This is followed by profiles which can be of three types -
user/peer/friend
; Name of the user profile should match with the h323id of the user device.
; For peer/friend profiles, host ip address must be provided as
"dynamic" is
; not supported as of now.
;
; Syntax for specifying a H323 device in extensions.conf is
; For Registered peers/friends profiles:
; OOH323/name where name is the name of the peer/friend profile.
;
; For unregistered H.323 phones:
; OOH323/ip[:port] OR if gk is used OOH323/alias where alias can be
any H323
; alias
;
; For dialing into another asterisk peer at a specific exten
; OOH323/exten/peer OR OOH323/exten@ip
;
; Domain name resolution is not yet supported.
;
; When a H.323 user calls into asterisk, his H323ID is matched with the
profile
; name and context is determined to route the call
;
; The channel driver will register all global aliases and aliases defined in
; peer profiles with the gatekeeper, if one exists. So, that when someone
; outside our pbx (non-user) calls an extension, gatekeeper will route that
; call to our asterisk box, from where it will be routed as per dial plan.
[general]
;Define the asetrisk server h323 endpoint
;The port asterisk should listen for incoming H323 connections.
;Default - 1720
port=1720
;The dotted IP address asterisk should listen on for incoming H323
;connections
;Default - tries to find out local ip address on it's own
bindaddr=213.138.36.153
;This parameter indicates whether channel driver should register with
;gatekeeper as a gateway or an endpoint.
;Default - no
gateway=yes
;Whether asterisk should use fast-start and tunneling for H323 connections.
;Default - yes
faststart=no
;h245tunneling=yes
;H323-ID to be used for asterisk server
;Default - Asterisk PBX
h323id=ObjSysAsterisk
;e164=100
;CallerID to use for calls
;Default - Same as h323id
callerid=asterisk
;Whether this asterisk server will use gatekeeper.
;Default - DISABLE
;gatekeeper = DISCOVER
gatekeeper = 213.138.36.153
;gatekeeper = DISABLE
;Location for H323 log file
;Default - /var/log/asterisk/h323_log
logfile=/var/log/asterisk/h323_log
;Following values apply to all users/peers/friends defined below, unless
;overridden within their client definition
;Sets default context all clients will be placed in.
;Default - default
context=default
;Sets rtptimeout for all clients, unless overridden
;Default - 60 seconds
;rtptimeout=60 ; Terminate call if 60 seconds of no RTP activity
; when we're not on hold
;Type of Service
;Default - none (lowdelay, thoughput, reliability, mincost, none)
;tos=none
;amaflags = default
;The account code used by default for all clients.
;accountcode=h3230101
;The codecs to be used for all clients.Only ulaw and gsm supported as of
now.
;Default - ulaw
; ONLY ulaw, gsm, g729 and g7231 supported as of now
allow=all ;Note order of disallow/allow is important.
;allow=g729
;allow=ulaw
; dtmf mode to be used by default for all clients. Supports rfc2833,
q931keypad
; h245alphanumeric, h245signal.
;Default - rfc 2833
dtmfmode=rfc2833
; User/peer/friend definitions:
; User config options Peer config options
; ------------------ -------------------
; context
; disallow disallow
; allow allow
; accountcode accountcode
; amaflags amaflags
; dtmfmode dtmfmode
; rtptimeout ip
; port
; h323id
; email
; url
; e164
; rtptimeout
;
;Define users here
;Section header is extension
;[myuser1]
;type=user
;context=context1
;disallow=all
;allow=gsm
;allow=ulaw
[dol_h323]
type=peer
ip=83.66.50.2 ; UPDATE with appropriate ip address
port=1720 ; UPDATE with appropriate port
disallow=all
allow=g729
;[myfriend1]
;type=friend
;context=default
;ip=10.0.0.82 ; UPDATE with appropriate ip address
;port=1820 ; UPDATE with appropriate port
;disallow=all
;allow=ulaw
;e164=12345
;rtptimeout=60
;dtmfmode=rfc2833
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Ümit AYDINLI
2006-Dec-07 07:20 UTC
[asterisk-users] Re: -- Called 12127773456@OOH323 Segmentation fault (core dumped)
Help me ooh323 core dumped. 2006/12/7, ?mit AYDINLI <uaydinli@gmail.com>:> > OOH323 Debugging Enabled > -- Executing Answer("SIP/3513-090f7d40", "") in new stack > -- Executing Wait("SIP/3513-090f7d40", "1") in new stack > -- Executing DeadAGI("SIP/3513-090f7d40", " a2billing.php|1") in new > stack > -- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php > a2billing.php|1: line:58 - IDCONFIG : 1 > a2billing.php|1: > a2billing.php|1: line:67 - MODE : standard > a2billing.php|1: > -- AGI Script Executing Application: (Dial) Options: ( > OOH323/12127773456@OOH323|60|HLxyz(5400000:31000:00000<OOH323/12127773456@OOH323%7C60%7CHLxyz(5400000:31000:00000> > )) > --- ooh323_request - data 12127773456@OOH323 format 0x4 (ulaw) > --- find_peer > +++ find_peer > +++ ooh323_request > --- ooh323_call- 12127773456@OOH323 > +++ ooh323_call > -- Called 12127773456@OOH323 > Segmentation fault (core dumped) > [root@asterisk1 ~]# > > > show version > Asterisk 1.2.12.1 built by root @ localhost.localdomain on a i686 running > Linux on 2006-10-18 18:35:57 UTC > > > ; Objective System's H323 Configuration example for Asterisk > ; ooh323c driver configuration > ; > ; [general] section defines global parameters > ; > ; This is followed by profiles which can be of three types - > user/peer/friend > ; Name of the user profile should match with the h323id of the user > device. > ; For peer/friend profiles, host ip address must be provided as "dynamic" > is > ; not supported as of now. > ; > ; Syntax for specifying a H323 device in extensions.conf is > ; For Registered peers/friends profiles: > ; OOH323/name where name is the name of the peer/friend profile. > ; > ; For unregistered H.323 phones: > ; OOH323/ip[:port] OR if gk is used OOH323/alias where alias can be > any H323 > ; alias > ; > ; For dialing into another asterisk peer at a specific exten > ; OOH323/exten/peer OR OOH323/exten@ip > ; > ; Domain name resolution is not yet supported. > ; > ; When a H.323 user calls into asterisk, his H323ID is matched with the > profile > ; name and context is determined to route the call > ; > ; The channel driver will register all global aliases and aliases defined > in > ; peer profiles with the gatekeeper, if one exists. So, that when someone > ; outside our pbx (non-user) calls an extension, gatekeeper will route > that > ; call to our asterisk box, from where it will be routed as per dial plan. > > > > [general] > ;Define the asetrisk server h323 endpoint > > ;The port asterisk should listen for incoming H323 connections. > ;Default - 1720 > port=1720 > > ;The dotted IP address asterisk should listen on for incoming H323 > ;connections > ;Default - tries to find out local ip address on it's own > bindaddr=213.138.36.153 > > ;This parameter indicates whether channel driver should register with > ;gatekeeper as a gateway or an endpoint. > ;Default - no > gateway=yes > > ;Whether asterisk should use fast-start and tunneling for H323 > connections. > ;Default - yes > faststart=no > ;h245tunneling=yes > > > ;H323-ID to be used for asterisk server > ;Default - Asterisk PBX > h323id=ObjSysAsterisk > ;e164=100 > > ;CallerID to use for calls > ;Default - Same as h323id > callerid=asterisk > > ;Whether this asterisk server will use gatekeeper. > ;Default - DISABLE > ;gatekeeper = DISCOVER > gatekeeper = 213.138.36.153 > ;gatekeeper = DISABLE > > ;Location for H323 log file > ;Default - /var/log/asterisk/h323_log > logfile=/var/log/asterisk/h323_log > > > ;Following values apply to all users/peers/friends defined below, unless > ;overridden within their client definition > > ;Sets default context all clients will be placed in. > ;Default - default > context=default > > ;Sets rtptimeout for all clients, unless overridden > ;Default - 60 seconds > ;rtptimeout=60 ; Terminate call if 60 seconds of no RTP activity > ; when we're not on hold > > ;Type of Service > ;Default - none (lowdelay, thoughput, reliability, mincost, none) > ;tos=none > > ;amaflags = default > > ;The account code used by default for all clients. > ;accountcode=h3230101 > > ;The codecs to be used for all clients.Only ulaw and gsm supported as of > now. > ;Default - ulaw > ; ONLY ulaw, gsm, g729 and g7231 supported as of now > allow=all ;Note order of disallow/allow is important. > ;allow=g729 > ;allow=ulaw > > > ; dtmf mode to be used by default for all clients. Supports rfc2833, > q931keypad > ; h245alphanumeric, h245signal. > ;Default - rfc 2833 > dtmfmode=rfc2833 > > ; User/peer/friend definitions: > ; User config options Peer config options > ; ------------------ ------------------- > ; context > ; disallow disallow > ; allow allow > ; accountcode accountcode > ; amaflags amaflags > ; dtmfmode dtmfmode > ; rtptimeout ip > ; port > ; h323id > ; email > ; url > ; e164 > ; rtptimeout > > ; > > ;Define users here > ;Section header is extension > ;[myuser1] > ;type=user > ;context=context1 > ;disallow=all > ;allow=gsm > ;allow=ulaw > > > > [dol_h323] > type=peer > ip=83.66.50.2 ; UPDATE with appropriate ip address > port=1720 ; UPDATE with appropriate port > disallow=all > allow=g729 > > ;[myfriend1] > ;type=friend > ;context=default > ;ip=10.0.0.82 ; UPDATE with appropriate ip address > ;port=1820 ; UPDATE with appropriate port > ;disallow=all > ;allow=ulaw > ;e164=12345 > ;rtptimeout=60 > ;dtmfmode=rfc2833 >-------------- next part -------------- An HTML attachment was scrubbed... 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