Hi, I'm having problems when I try to place a phonecall using a
gatekeeper, it connects through, but then disconnects straight away.
I have attached the Asterisk debug output (ast.txt) and my ooh323.conf
file.
Not really sure whats going on but I had the same configuration working
with OH323 fine.
If anyone can point out whats going wrong I would really appreciate
it! :-)
Thanks,
Pat
*CLI> 2005-11-04 16:39:58 DEBUG[4361]: chan_sip.c:3082 sip_alloc: Allocating
new SIP dialog for [EMAIL PROTECTED] - INVITE (With RTP)
2005-11-04 16:39:58 DEBUG[4361]: chan_sip.c:10887 handle_request: **** Received
INVITE (5) - Command in SIP INVITE
2005-11-04 16:39:58 DEBUG[4361]: chan_sip.c:1005 parse_sip_options: * SIP
extension value: 2 for call [EMAIL PROTECTED]
2005-11-04 16:39:58 DEBUG[4361]: chan_sip.c:7076 check_user_full: Setting NAT
on RTP to 0
2005-11-04 16:39:58 DEBUG[4361]: chan_sip.c:10298 handle_request_invite:
Checking SIP call limits for device
2005-11-04 16:39:58 DEBUG[4361]: chan_sip.c:2176 update_call_counter: Updating
call counter for incoming call
2005-11-04 16:39:58 DEBUG[4361]: chan_sip.c:5977 build_route: build_route:
Record-Route hop: <sip:202.89.128.51;ftag=D3F0A0F-E00EB3EA;lr=on>
2005-11-04 16:39:58 DEBUG[4374]: pbx.c:1527
pbx_substitute_variables_helper_full: Function result is ''
2005-11-04 16:39:58 DEBUG[4374]: pbx.c:1682 pbx_extension_helper: Launching
'Set'
2005-11-04 16:39:58 DEBUG[4374]: pbx.c:1682 pbx_extension_helper: Launching
'Goto'
2005-11-04 16:39:58 DEBUG[4374]: pbx.c:1682 pbx_extension_helper: Launching
'Goto'
2005-11-04 16:39:58 DEBUG[4374]: pbx.c:1682 pbx_extension_helper: Launching
'Dial'
--- ooh323_request - data 6499160277 format 0x100 (g729)
2005-11-04 16:39:58 DEBUG[4374]: src/chan_h323.c:360 ooh323_alloc: ---
ooh323_alloc
2005-11-04 16:39:58 DEBUG[4374]: src/chan_h323.c:443 ooh323_alloc: +++
ooh323_alloc
--- find_peer
+++ find_peer
2005-11-04 16:39:58 DEBUG[4374]: src/chan_h323.c:242 ooh323_new: ---
ooh323_new - 6499160277
2005-11-04 16:39:58 DEBUG[4374]: src/chan_h323.c:348 ooh323_new: +++ h323_new
+++ ooh323_request
2005-11-04 16:39:58 DEBUG[4374]: channel.c:2788 ast_channel_inherit_variables:
Not copying variable STACK-h323-call-outgoing-6499160277-1.
2005-11-04 16:39:58 DEBUG[4374]: channel.c:2788 ast_channel_inherit_variables:
Not copying variable STACK-route-sip-client-099160277-1.
2005-11-04 16:39:58 DEBUG[4374]: channel.c:2788 ast_channel_inherit_variables:
Not copying variable STACK-incoming-sip-client-099160277-2.
2005-11-04 16:39:58 DEBUG[4374]: channel.c:2788 ast_channel_inherit_variables:
Not copying variable STACK-incoming-sip-client-099160277-1.
2005-11-04 16:39:58 DEBUG[4374]: channel.c:2788 ast_channel_inherit_variables:
Not copying variable SIPCALLID.
2005-11-04 16:39:58 DEBUG[4374]: channel.c:2788 ast_channel_inherit_variables:
Not copying variable SIPUSERAGENT.
2005-11-04 16:39:58 DEBUG[4374]: channel.c:2788 ast_channel_inherit_variables:
Not copying variable SIPDOMAIN.
2005-11-04 16:39:58 DEBUG[4374]: channel.c:2788 ast_channel_inherit_variables:
Not copying variable SIPURI.
--- ooh323_call- 6499160277
+++ ooh323_call
2005-11-04 16:39:58 DEBUG[4374]: channel.c:2330 set_format: Set channel
OOH323/6499160277-4aa3 to read format g729
2005-11-04 16:39:58 DEBUG[4374]: channel.c:2330 set_format: Set channel
SIP/sip.qsi.net.nz-09f66e70 to write format g729
2005-11-04 16:39:58 DEBUG[4374]: channel.c:2330 set_format: Set channel
SIP/sip.qsi.net.nz-09f66e70 to read format g729
2005-11-04 16:39:58 DEBUG[4374]: channel.c:2330 set_format: Set channel
OOH323/6499160277-4aa3 to write format g729
2005-11-04 16:39:58 DEBUG[4359]: chan_sip.c:11409 sip_devicestate: Checking
device state for peer sip.qsi.net.nz
--- onNewCallCreated ooh323c_o_1
--- find_call
+++ find_call
Outgoing call 6499160277(ooh323c_o_1) - Codec prefs - ()
Adding capabilities to call(outgoing, ooh323c_o_1)
--- configure_local_rtp
+++ configure_local_rtp
+++ onNewCallCreated ooh323c_o_1
2005-11-04 16:39:58 DEBUG[4359]: devicestate.c:186 do_state_change: Changing
state for SIP/sip.qsi.net.nz - state 2 (In use)
2005-11-04 16:39:58 DEBUG[4375]: app_queue.c:443 changethread: Device
'SIP/sip.qsi.net.nz' changed to state '2' (In use)
--- onAlerting ooh323c_o_1
--- find_call
+++ find_call
+++ onAlerting ooh323c_o_1
2005-11-04 16:39:58 DEBUG[4359]: devicestate.c:186 do_state_change: Changing
state for OOH323/6499160277 - state 6 (Ringing)
2005-11-04 16:39:58 DEBUG[4376]: app_queue.c:443 changethread: Device
'OOH323/6499160277' changed to state '6' (Ringing)
--- onCallCleared ooh323c_o_1
--- find_call
+++ find_call
2005-11-04 16:39:58 DEBUG[4374]: channel.c:1307 ast_hangup: Hanging up channel
'OOH323/6499160277-4aa3'
--- ooh323_hangup
hanging 6499160277
+++ ooh323_hangup
2005-11-04 16:39:58 DEBUG[4359]: devicestate.c:186 do_state_change: Changing
state for OOH323/6499160277 - state 0 (Unknown)
2005-11-04 16:39:58 DEBUG[4377]: app_queue.c:443 changethread: Device
'OOH323/6499160277' changed to state '0' (Unknown)
2005-11-04 16:39:58 DEBUG[4374]: app_dial.c:1676 dial_exec_full: Exiting with
DIALSTATUS=NOANSWER.
2005-11-04 16:39:58 DEBUG[4374]: channel.c:1307 ast_hangup: Hanging up channel
'SIP/sip.qsi.net.nz-09f66e70'
2005-11-04 16:39:58 DEBUG[4374]: chan_sip.c:2382 sip_hangup: Hangup call
SIP/sip.qsi.net.nz-09f66e70, SIP callid [EMAIL PROTECTED])
2005-11-04 16:39:58 DEBUG[4374]: chan_sip.c:2390 sip_hangup:
update_call_counter() - decrement call limit counter
2005-11-04 16:39:58 DEBUG[4374]: chan_sip.c:2176 update_call_counter: Updating
call counter for incoming call
2005-11-04 16:39:58 DEBUG[4374]: chan_sip.c:2360 hangup_cause2sip: AST hangup
cause 16 (no match found in SIP)
2005-11-04 16:39:58 DEBUG[4359]: chan_sip.c:11409 sip_devicestate: Checking
device state for peer sip.qsi.net.nz
2005-11-04 16:39:58 DEBUG[4359]: devicestate.c:186 do_state_change: Changing
state for SIP/sip.qsi.net.nz - state 1 (Not in use)
2005-11-04 16:39:58 DEBUG[4378]: app_queue.c:443 changethread: Device
'SIP/sip.qsi.net.nz' changed to state '1' (Not in use)
2005-11-04 16:39:58 DEBUG[4361]: chan_sip.c:3130 find_call: = Found Their Call
ID: [EMAIL PROTECTED] Their Tag D3F0A0F-E00EB3EA Our tag: as5612a46c
2005-11-04 16:39:58 DEBUG[4361]: chan_sip.c:10887 handle_request: **** Received
ACK (6) - Command in SIP ACK
2005-11-04 16:39:58 DEBUG[4361]: chan_sip.c:1370 __sip_ack: Stopping
retransmission on '[EMAIL PROTECTED]' of Response 2: Match Found
--- ooh323_destroy
Destroying 6499160277
+++ ooh323_destroy
; 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/[EMAIL PROTECTED]
;
; 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=202.89.128.52
;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=no
;H323-ID to be used for asterisk server
;Default - Asterisk PBX
h323id=SIP1
;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 = 202.89.128.228
;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=lowdelay
;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=gsm
;allow=ulaw
;allow=g729
; 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
;[mypeer1]
;type=peer
;context=context2
;ip=a.b.c.d ; UPDATE with appropriate ip address
;port=1720 ; UPDATE with appropriate port
;e164=101
;[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
;[vm1]
;type=peer
;ip=202.89.128.228
;port=1720
;disallow=all
;allow=g729
[cms1mercury]
type=peer
ip=202.89.128.227
port=1720
disallow=all
allow=g729
