IPO connection to Asterisk over SIP Trunks 1

[Sorpio]

Hello,

is it possible to connect an asterisk server over SIP-Trunks? I´m not familiar with Asterisk and don´t have that deep knowledge in SIP Trunking.

Thanks Sorpio

 

[MrIPO]

Yes this is possable we have done it in the early days of 4.0 trial, our setup was

Nokia E61, connected via Wireless LAN to the Asterisk as a SIP extension, connected via SIP trunk on the IPO. Could call from the 5410 to the E61 Nice ...

I will see if I can find the Asterisk config when I get to the office.

http://www.iporesource.com

 

[Sorpio]

fine, if you can provide the config, it would be great. Thanks a lot

 

[MrIPO]

This is all I could fine, I assume this was our setup.

Incase anyone is trying this and needs a basic setup. This was done using asterisk trixbox. To enable h323 in trixbox follow these steps first:

cp /etc/asterisk-1.2.7.1_samples/ooh323.conf /etc/asterisk
amportal stop
amportal start


sip.conf

[general]
context=internal
port=5060
bindaddr=0.0.0.0
srvlookup=yes
disallow=all
allow=ulaw


[3000]
type = friend
context=internal
username = 3000
host = dynamic
mailbox = 3000
dtmfmode=rfc2833

Extensions.conf

[general]

autofallthrough=yes

[internal]
exten => 3000,1,Dial(SIP/3000,30,Ttm)
exten => 3000,2,Playback(invalid)
exten => 3000,3,Hangup
exten => 3000,102,Voicemail(u3000)
exten => 3000,103,Hangup

exten => _89XX,1,Dial(OOH323/${EXTEN}@ipo) ;change _89XX to whatever extensions are on the IPO

OOH323.conf

[general]

faststart=yes
h245tunneling=yes
gatekeeper = DISABLE


[ipo]
type=peer
context=internal
ip=XXX.XXX.XXX.XXX ;Your IPO IP
port=1720
disallow=all
allow=ulaw
canreinvite=no
dtmfmode=rfc2833

IP Office Line

Voip Settings

Gateway IP = asterisk ip
compression mode = automatic
h450 support = h450

enable fast start = checked
Out of band dtmf = checked
allow direct media path = checked

http://www.iporesource.com

 

[wenicolas]

hello i use your configuration in my trixbox and in my IPoffice SOE but i have no communication...

In asterisk i use tribox and i write your configuration in config edit and in my ipoffice i configure just a new line IP. So i don't use sip on my ipoffice soe okay ? And in Ipoffice i must configure other ? In asterik i use SIP count for my extension with no problem but it's impossible to call extension on my soe.

I am lost and i want your help if possible.

thank you very much.

Nicolas.

 

[kwing112000]

do you have SIP licenses in the IPO. You will need them.

 

[wenicolas]

ok i don't know licence is compulsory...And if i just use your configuration i have no problem with a telephon sip on asterisk and a 5410 on my soe ? i do not need another configuration ? So in my soe i create a sip line and not IP line it's really ?

thank you very much for your very fast answer.

Thank you.

Nicolas

 

[wenicolas]

I read your configuration and I test this but it's wrong. But i am testing connection between two trixbox, i use IAX protocol. I create two extensions on asterisk server and on server in my dial patern i write if you don't know extension on the server put the call on this extension and asterisk1 call asterisk2 with extension. I use this with no problem. And with Ip office why i don't create extension ? Why MrIPO create ligne IP and not SIP line ?

Thank you very much for your answer.

 

[chkbrt]

Mr IPO used my post from awhile back. Its not a sip trunk config but a h323 conf. This is free to do with an IPO and an astrisk box. it does work, very well actually...