hi all
i have installed a network as follows:
SS1—>asterisk---->SS2
asterisk is used to translate between h323(SS1) to SIP(SS2)
the calls passes through the second SS put when the called party answers the call get disconnected.
extensions.conf file:
[general]
static=yes
writeprotect=no
autofallthrough=no
[globals]
TRUNK=Zap/g2 ; Trunk interface
[default]
;exten => _0000.,1,Answer()
;exten => _0000.,2,Wait(1)
;exten => _0000.,3,Dial(ooH323/0001${EXTEN:4}@xx.xx.xx.xx)
;exten => _0000.,3,Hangup()
exten => 9,1,Dial(ooh323/*****)
exten => 9,1,Hangup()
;exten => _0000.,1,Answer()
;exten => _0000.,2,Dial(ooH323/0001${EXTEN:4}/*****)
;exten => t, 1 ,Answer()
;exten => t, 2 ,Dial(ooH323/0001${EXTEN:4}/*****)
exten => _0000.,1,Answer()
exten => _0000.,2,Wait(1)
exten => _0000.,3,Dial(SIP/${EXTEN:4}@xxx.xxx.xxx.xxx//SS2 IP)
;exten => _0000.,4,Hangup()
;exten => s,1,Dial(SIP/${EXTEN:4}@xxx.xxx.xxx.xxx//SS2 IP)
;------------TEST Extensions—;
exten => _1XX,1,Dial(SIP/${EXTEN},20,r)
exten => _1XX,2,Hangup
;------------------------------;
SIP.conf file:
[general]
context=default
allowguest=yes
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
maxexpirey=3600
canreinvite=yes
;allow=all
disallow=all
;allow=g729
;allow=alaw
;allow=ulaw
allow=g723
;allow=gsm
nat=yes
relaxdtmf=yes
rtptimeout=120
[xxx.xxx.xxx.xxx//SS2 IP]
type=peer
username=xxxxx
secret=xxxxx
host=xxx.xxx.xxx.xxx//SS2 IP
nat=yes
context=default
[xxxx]
type = friend
secret = xxxxx
username = xxxxx
host = dynamic
context = default
[xxx]
type = friend
secret = xxxx
username = xxxx
host = dynamic
context = default
[xxxx]
type = friend
secret =
username = xxxxx
host = dynamic
context = default
ooh323.conf file:
[general]
bindaddr=0.0.0.0
;h323id=ObjSysAsterisk
;e164=101
callerid=asterisk
gatekeeper=DISABLE
logfile=/var/log/asterisk/h323_log
faststart=yes
tos=lowdelay
;allow=all
disallaw=all
;allow=g729
allow=g723
;allow=alaw
;allow=ulaw
;allow=gsm
context=default
[xx.xx.xx.xx//asterisk server IP]
type=friend
context=default
ip=xx.xx.xx.xx//asterisk server ip
port=1720 ; UPDATE with appropriate port
;allow=all
disallow=all
;allow=g729
allow=g723
;allow=alaw
;allow=ulaw
e164=12345
rtptimeout=60
dtmfmode=rfc2833
[teleco]
type=friend
context=default
ip=xx.xx.xx.xx//asterisk server ip
port=1720 ; UPDATE with appropriate port
;allow=all
disallow=all
;allow=g729
allow=g723
;allow=alaw
;allow=ulaw
;allow=gsm
when i call i get these logs :
From: NOC [mailto:noc@telecolink.com]
Sent: August-23-07 1:29 PM
To: jehan younis (jehanzaib_kiani@hotmail.com)
Subject: logs for g723
*CLI> Aug 23 13:27:58 DEBUG[4424]: channel.c:775 channel_find_locked: Avoiding initial deadlock for ‘OOH323/xx.xx.xx.xx //SS1 IP-42c7’
– Executing Answer(“OOH323/ xx.xx.xx.xx //SS1 IP -42c7”, “”) in new stack
– Executing Wait(“OOH323/ xx.xx.xx.xx //SS1 IP -42c7”, “1”) in new stack
Aug 23 13:27:59 DEBUG[4458]: src/chan_h323.c:3241 ooh323_rtp_read: Oooh, format changed to 1
Aug 23 13:27:59 WARNING[4458]: channel.c:2403 set_format: Unable to find a codec translation path from g723 to ulaw
Aug 23 13:27:59 WARNING[4458]: channel.c:2403 set_format: Unable to find a codec translation path from g723 to ulaw
Aug 23 13:27:59 WARNING[4458]: channel.c:2403 set_format: Unable to find a codec translation path from g723 to ulaw
Aug 23 13:27:59 WARNING[4458]: channel.c:2403 set_format: Unable to find a codec translation path from g723 to ulaw
– Executing Dial(“OOH323/ xx.xx.xx.xx //SS1 IP -42c7”, "SIP/phone number@ xx.xx.xx.xx //SS2 IP “) in new stack
Aug 23 13:27:59 DEBUG[4458]: chan_sip.c:1889 create_addr_from_peer: Setting NAT on RTP to 524288
Aug 23 13:27:59 DEBUG[4458]: chan_sip.c:2085 sip_call: Outgoing Call for phone number
– Called phone number@ xx.xx.xx.xx //SS2 IP
Aug 23 13:27:59 DEBUG[4439]: chan_sip.c:1468 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '5697f1194ab55b2614c1c7e846f20857@ xx.xx.xx.xx //asterisk server IP ’ Request 102: Found
Aug 23 13:28:01 DEBUG[4439]: chan_sip.c:1468 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '5697f1194ab55b2614c1c7e846f20857@ xx.xx.xx.xx //asterisk server IP ’ Request 102: Found
– SIP/ xx.xx.xx.xx //SS2 IP -08531998 is making progress passing it to OOH323/ xx.xx.xx.xx //SS1 IP -42c7
Aug 23 13:28:20 DEBUG[4439]: chan_sip.c:1392 __sip_ack: Acked pending invite 102
Aug 23 13:28:20 DEBUG[4439]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '5697f1194ab55b2614c1c7e846f20857@ xx.xx.xx.xx //asterisk server IP ’ of Request 102: Match Found
Aug 23 13:28:20 DEBUG[4439]: chan_sip.c:6282 build_route: build_route: Contact hop: <sip: xx.xx.xx.xx //SS2 IP:5060;transport=udp>
– SIP/ xx.xx.xx.xx //SS2 IP -08531998 answered OOH323/ xx.xx.xx.xx //SS1 IP -42c7
– Attempting native bridge of OOH323/ xx.xx.xx.xx //SS1 IP -42c7 and SIP/ xx.xx.xx.xx //SS2 IP -08531998
Aug 23 13:28:20 DEBUG[4439]: chan_sip.c:1392 __sip_ack: Acked pending invite 103
Aug 23 13:28:20 DEBUG[4439]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '5697f1194ab55b2614c1c7e846f20857@ xx.xx.xx.xx //asterisk server IP ’ of Request 103: Match Found
Aug 23 13:28:20 DEBUG[4439]: chan_sip.c:6225 build_route: build_route: Retaining previous route: <sip: xx.xx.xx.xx //SS2 IP:5060;transport=udp>
Aug 23 13:28:32 DEBUG[4458]: channel.c:3637 ast_channel_bridge: Returning from native bridge, channels: OOH323/ xx.xx.xx.xx //SS1 IP -42c7, SIP/ xx.xx.xx.xx //SS2 IP -08531998
Aug 23 13:28:32 DEBUG[4458]: chan_sip.c:2450 sip_hangup: update_call_counter(phone number) - decrement call limit counter
Aug 23 13:28:32 DEBUG[4458]: app_dial.c:1661 dial_exec_full: Exiting with DIALSTATUS=ANSWER.
== Spawn extension (default, 0000phne number, 3) exited non-zero on 'OOH323/ xx.xx.xx.xx //SS1 IP -42c7’
Aug 23 13:28:32 DEBUG[4458]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '” xx.xx.xx.xx //SS1 IP " <18884923464>'
Aug 23 13:28:32 DEBUG[4458]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '18884923464’
Aug 23 13:28:32 DEBUG[4458]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '0000phone number’
Aug 23 13:28:32 DEBUG[4458]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is 'default’
Aug 23 13:28:32 DEBUG[4458]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is 'OOH323/ xx.xx.xx.xx //SS1 IP -42c7’
Aug 23 13:28:32 DEBUG[4458]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is 'SIP/ xx.xx.xx.xx //SS2 IP -08531998’
Aug 23 13:28:32 DEBUG[4458]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is 'Dial’
Aug 23 13:28:32 DEBUG[4458]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is 'SIP/phone number@ xx.xx.xx.xx //SS2 IP '
Aug 23 13:28:32 DEBUG[4458]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '2007-08-23 13:27:58’
Aug 23 13:28:32 DEBUG[4458]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '2007-08-23 13:27:58’
Aug 23 13:28:32 DEBUG[4458]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '2007-08-23 13:28:32’
Aug 23 13:28:32 DEBUG[4458]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '34’
Aug 23 13:28:32 DEBUG[4458]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '34’
Aug 23 13:28:32 DEBUG[4458]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is 'ANSWERED’
Aug 23 13:28:32 DEBUG[4458]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is 'DOCUMENTATION’
Aug 23 13:28:32 DEBUG[4458]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is 'ast_h323’
Aug 23 13:28:32 DEBUG[4458]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '1187890078.4’
Aug 23 13:28:32 DEBUG[4458]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '(null)'
Aug 23 13:28:32 DEBUG[4439]: chan_sip.c:1392 __sip_ack: Acked pending invite 104
Aug 23 13:28:32 DEBUG[4439]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '5697f1194ab55b2614c1c7e846f20857@ xx.xx.xx.xx //asterisk server IP ’ of Request 104: Match Found
Aug 23 13:28:32 DEBUG[4439]: chan_sip.c:6225 build_route: build_route: Retaining previous route: <sip: xx.xx.xx.xx //SS2 IP:5060;transport=udp>
Aug 23 13:28:32 DEBUG[4439]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '5697f1194ab55b2614c1c7e846f20857@ xx.xx.xx.xx //asterisk server IP of Request 105: Match Found
any ideas?
thnx
Emil