Quintum A800 with asterisk

Asterisk Asterisk Support
1

hi,

can anybody tell me how I can configure my asterisk to place call to quintum A800???

which conf file I have to configure in asterisk ??? I am really confused!!! whether it is sip.conf or oh323.conf or h323.conf???

Thanks in advance

2

I’ve got it working with Asterisk 1.8 and a Quintum CRSP. Try the following in your /etc/asterisk/ooh323.conf – changing 1.2.3.4 to your Asterisk IP and 5.6.7.8 to the A800 IP.

[general]
port=1720
bindaddr=1.2.3.4
h323id=ObjSysAsterisk
e164=100
callerid=asterisk
gatekeeper=DISABLE
faststart=yes
h245tunneling=yes
mediawaitforconnect=no
logfile=/var/log/asterisk/h323_log
tracelevel=6
context=default
rtptimeout=60
tos=lowdelay
amaflags=default
accountcode=h3230101
dtmfmode=rfc2833
disallow=all
allow=g729
allow=g723
;g729onlyA=yes

[a800]
type=friend
ip=5.6.7.8
port=1720

Then in your /etc/asterisk/extensions.conf:

exten => _X.,n,Dial(OOH323/a800/${EXTEN})

3

Hi,Thanks for your post.

I already set tenor A800 using sip module.Working fine.But anyway thanks for your reply.

4

Do you have it converting SIP<->H323 ?

5

Hi both of you, your experience could probably be usefull to me.
I use Asterisk 1.6.2, I have a Quintum AX800 for placing call to pstn and some analog extyensions.
I use SIP on quintum to connect to Asterisk.
I can send call to pstn from IP Phone logged on asterisk, or analog phones on the Quintum, I can also call Quintum analog extension from IP phones logged on asterisk, etc… so almost everything is working fine, except that inbound calls are not received by asterisk.
I can see on the quintum config manager status page, when I call from pstn, that there is an inbound call active, but asterisk is no receiving anything, do you have an idea where I could have missed something ?

Below is my asterisk config for the trunck
I have also created the outbound route ( that works fine)
and the inbound route, which is where i seem to have a problem, but i created it accepeting any DID or CID, so wide open, I dont see why the error could be in the configuration of this route.

Asterisk TRUNK PEER Details:

context=from-trunk
host=192.168.50.76
insecure=invite,port
nat=yes
port=5060
secret=Quintum
type=peer
username=TenorAX
qualify=yes
canreinvite=yes

Asterisk TRUNK USER Details:

secret=Quintum
type=user
context=from-trunk
qualify=yes
canreinvite=no
insecure=port,invite
host=192.168.50.76
username=TenorAX

for configuring the Tenor I got inspired from this tutorial :
http://jazinga.helpserve.com/index.php?_m=knowledgebase&_a=viewarticle&kbarticleid=25
I just didn’t folow their step 5, otherwise the Quintum would not register at all on asterisk.

Thanks for you help to allow incoming pstn calls get up to asterisk !

Vincent

6

Well, I found a solution, I don’t know if it is very clean, but creating and additional extension in Asterisk let’s say 199 with the password abc123 and puting those into the quintum as the credentials for the first user agent of the SIP signaling group, it worked… if I find a cleaner method I’ll keep posting here.
Cheers.

7

I’m new to asterisk. i have installed Asterisk 1.8 on Centos and factory reset Quintum A800.
I’m looking for copy/paste solution for connecting Quintum tenor a800 via SIP or H323 to Asterisk Server.
I would like to use Quintum A800 Tenor as an analog Gateway to send/receive calls to/from PSTN.