# Sip to h323; h323 to sip

**URL:** <https://community.asterisk.org/t/sip-to-h323-h323-to-sip/3547>\
**Category:** Asterisk Support\
**Created:** [January 9, 2006, 11:08am UTC](https://community.asterisk.org/t/sip-to-h323-h323-to-sip/3547 "2006-01-09T11:08:29Z")\
**Posts on this page:** 2\
**Page:** 1

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**Author:** ![mole\_13](https://avatars.discourse-cdn.com/v4/letter/m/71c47a/32.png) [@mole\_13](https://community.asterisk.org/u/mole_13)\
**Post date:** [January 9, 2006, 11:08am UTC](https://community.asterisk.org/t/sip-to-h323-h323-to-sip/3547/1 "2006-01-09T11:08:29Z")

</div>

hi everybody!

i would like to ask has anyone tried sip to h323 and h323 to sip calls?

using GNUGK and OOh323 driver module for asterisk 1.2

thank you very much.

ive tried but i cant make ti have a successful call.

DIAL PLAN:

[default]  
exten =\> \_X.,1,Dial(SIP/${EXTEN},60)  
exten =\> \_X.,2,Hangup  
include =\> h323\_in  
include =\> to\_h323

;from h323  
[h323\_in]  
exten =\> \_5.,1,Dial(SIP/&(EXTEN),60)  
exten =\> \_5.,2,Hangup

;to h323  
[to\_h323]  
exten =\> \_6.,1,Dial(H323/&(EXTEN),60)  
exten =\> \_6.,2,HangUp

OOH323.CONF

[general]  
;Define the asetrisk server h323 endpoint  
listenAddress=192.x.x.x  
listenPort=1720  
tcpStart=10000  
tcpEnd=20000  
udpStart=10000  
udpEnd=10000

[test\_2]  
type=friend  
context=h323\_in  
port=1720  
h323id=sipman  
ip=192.x.x.x  
disallow=all  
allow=ulaw  
rtptimeout=60  
dtmfmode=rfc2833

😄

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<div class="post-metadata">

**Author:** ![jsgoecke](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/jsgoecke/32/585_2.png) [@jsgoecke](https://community.asterisk.org/u/jsgoecke)\
**Post date:** [January 9, 2006, 2:02pm UTC](https://community.asterisk.org/t/sip-to-h323-h323-to-sip/3547/2 "2006-01-09T14:02:24Z")

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I have with an Avaya Prologix using ooh323. With v1.2.1 of Asterisk you need to address the H323 channel as follows:

Dial(OOH323/${EXTEN}@89.1.250.101|45);

If you still have problems, please post your CLI output when you attempt to dial.