[asterisk-users] H.323 Trunk between Asterisk 11 and Avaya

jorgearturo at protoboardmx.com jorgearturo at protoboardmx.com
Fri Jun 7 19:35:04 CDT 2013



Hello, 

I'm trying to create a H.323 trunk between Asterisk 11 and Avaya. I have
done this before between Asterisk 1.6 and Avaya but had some issues placing
external calls from the Asterisk to the Public network which is connected
to Avaya. I'm trying to create that trunk on Asterisk 11 because the 1.6 is
outdated and has no support. 

On the Asterisk side I have Aastra 6731i SIP phones and on the Avaya I'm
using only H.323 phones. So far I can call from Asterisk to Avaya but
there's no sound and when calling from Avaya to Asterisk the call is
dropped. 

Here are my config files: 

sip.config: 

[general]
allowguest=no
srvlookup=no
udpbindaddr=192.168.1.252 ; IP of the Asterisk box
tcpenable=yes ; if I set it to no I can't call between the Aastra phones

[office-phone](!)
type=friend
secret=1234
context=LocalSets
host=dynamic
nat=force_rport,comedia
dtmfmode=auto
disallow=all
allow=alaw
allow=ulaw
port=5060

[426](office-phone)
[427](office-phone) 

ooh323.conf:


[general]
port=1720
bindaddr=0.0.0.0
gateway=no
canreinvite=no
faststart=yes
h245tunneling=yes
h323id=ObjSysAsterisk
e164=100
callerid=asterisk
gatekeeper = DISABLE
context=trunk
disallow=all
allow=alaw
allow=ulaw
dtmfmode=outofband

[avaya]
type=friend
ip=192.168.1.150 ; IP address of the Avaya
port=1720 

extensions.conf: 

[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no

[trunk]
exten=>_8xx,1,Dial(OOH323/${EXTEN}@AVAYA)
exten=>_426,1,Dial(SIP/${EXTEN})
exten=>_427,1,DialIP/${EXTEN})

[LocalSets]
include=>trunk 

When I try to place a call from Avaya to Asterisk this is what appears on
the CLI: 

 -- Executing [426 at trunk:1] Dial("OOH323/avaya-2", "SIP/426") in new stack
 == Using SIP RTP CoS mark 5
 -- Called SIP/426
 == Spawn extension (trunk, 426, 1) exited non-zero on 'OOH323/avaya-2'

When I make a call from Asterisk to Avaya this is the output, the last line
appears when I hang the call: 

 == Using SIP RTP CoS mark 5
 -- Executing
[821 at LocalSets:1] Dial("SIP/426-00000003",
"OOH323/821 at AVAYA") in new stack
 -- Called OOH323/821 at AVAYA
 -- OOH323/avaya-3 is ringing
 -- OOH323/avaya-3 answered SIP/426-00000003
 > 0xb6d02370 -- Probation passed - setting RTP source address to
192.168.1.215:3000
 == Spawn extension (LocalSets, 821, 1) exited non-zero on
'SIP/426-00000003'

Hope you can give me a guide on what to change in order to get this trunk
to work. 

Thank you.
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