[Asterisk-Users] Segmentation fault on using SIP -> H323

[Thomas Haeger]

Hi all,

if i make a call between one SIP soft-phone to an other soft phone over
asterisk, i get a Segmentation fault after take up.

The extension is :

	exten => _00.,1,Dial,OH323/${EXTEN}@<myip>|60|r

This means, if a SIP client comes with 00* then dial to <myip> over H323. If
the H323 client takes up, a Segmentation fault occures.


But, if the extension is

	exten => _00.,1,Dial,SIP/${EXTEN}@<myip>|60|r

it works.

And, the other direction works , too....

What's wrong ?




Thanks for Help,
Thomas.