Hello,
I am currently using asterisk-1.2.19 and asterisk-addons-1.2.7. I am trying to interface Asterisk with Avaya via H323. I am getting only one-way communication between Avaya and Asterisk. When a call is made from Avaya to Asterisk the phone rings and when answered only the calling party can be heard. Calls fail from Asterisk to Avaya. Below is a sample of my configuration.
Avaya to Asterisk (working)
CLI>
– Executin Dial("OOH323/Heal(“OOH323/Henlin, Michael-c178”, “SIP/200”) in new stack
– Called 200
+++ onAlerting ooh323c_1
– SIP/200-0a06a608 is ringing
----- ooh323_indicate 3 on call ooh323c_1
++++ ooh323_indicate 3 on ooh323c_1
– SIP/200-0a06a608 answered OOH323/Henlin, Michael-c178
----- ooh323_indicate -1 on call ooh323c_1
Jul 4 11:28:31 WARNING[1586]: src/chan_h323.c:1024 ooh323_indicate: Don’t know how to indicate condition -1 on ooh323c_1
++++ ooh323_indicate -1 on ooh323c_1
— ooh323_answer
+++ ooh323_answer
– Attempting native bridge of OOH323/Henlin, Michael-c178 and SIP/200-0a06a608
— onCallEstablished ooh323c_1
— find_call
+++ find_call
+++ onCallEstablished ooh323c_1
— close_rtp_connection
— find_call
+++ find_call
+++ close_rtp_connection
— onCallCleared ooh323c_1
— find_call
+++ find_call
== Spawn extension (default, 200, 1) exited non-zero on ‘OOH323/Henlin, Michael-c178’
— ooh323_hangup
hanging Henlin, Michael
+++ ooh323_hangup
— ooh323_destroy
Destroying Henlin, Michael
+++ ooh323_destroy
Asterisk to Avaya (failing)
CLI>
– Executing Dial(“SIP/200-0a0638c8”, “OOH323/062158@ecsoutg”) in new stack
— ooh323_request - data 062158@ecsoutg format 0x4 (ulaw)
— find_peer "ecsoutg"
comparing with "10.10.37.18"
found matching peer
+++ find_peer “ecsoutg”
+++ ooh323_request
— ooh323_call- 062158@ecsoutg
+++ ooh323_call
– Called 062158@ecsoutg
— onNewCallCreated ooh323c_o_2
— find_call
+++ find_call
setting callid number 200
Outgoing call ecsoutg(ooh323c_o_2) - Codec prefs - (ulaw)
Adding capabilities to call(outgoing, ooh323c_o_2)
Adding g711 ulaw capability to call(outgoing, ooh323c_o_2)
— configure_local_rtp
+++ configure_local_rtp
+++ onNewCallCreated ooh323c_o_2
— onCallCleared ooh323c_o_2
— find_call
+++ find_call
— ooh323_hangup
hanging ecsoutg
+++ ooh323_hangup
== Everyone is busy/congested at this time (1:0/1/0)
– Executing Congestion(“SIP/200-0a0638c8”, “5”) in new stack
== Spawn extension (from-internal, 062158, 2) exited non-zero on ‘SIP/200-0a0638c8’
— ooh323_destroy
Destroying ecsoutg
+++ ooh323_destroy
extensions.conf
[general]
autofallthrough=yes
[from-internal]
include => default
exten => _065XXX,1,Dial(IAX2/East_Parade/${EXTEN:3})
exten => _067XXX,1,Dial(IAX2/Rockfort/${EXTEN:3})
exten => _046XXX,1,Dial(IAX2/Paradise/${EXTEN:3})
exten => _0XXXXX,1,Dial(OOH323/${EXTEN}@ecsoutg)
exten => _0605XXX,1,Dial(${OOH323/${EXTEN:3}@ecsoutg) ; calls to Call Centre
[internal]
exten => 200,1,Dial(SIP/200)
exten => 200,2,Voicemail(u200)
exten => 200,101,Voicemail(b200)
sip.conf
[general]
context=from-internal
disallow=all
allow=alaw
allow=ulaw
srvlookup=yes
canreinvite=no
[200]
type=friend
regexten=200
callerid=200
host=dynamic
context=from-internal
disallow=all
allow=ulaw
secret=1234
ooh323.conf
[general]
;Whether asterisk should use fast-start and tunneling for H323 connections.
;Default - yes
faststart=yes
h245tunneling=yes
;Alias address for for asterisk server
;Default - "Asterisk PBX"
h323id=Asterisk-GW-2
e164=100
[ecsinc]
type=user
context=from-internal
disallow=all
allow=ulaw
[ecsoutg]
type=peer
context=default
ip=10.10.37.18
port=1720 ; UPDATE with appropriate port
h323id=Asterisk-GW-2
disallow=all
allow=ulaw
canreinvite=no