Thomas Haeger
2003-May-21 08:04 UTC
[Asterisk-Users] Segmentation fault on using SIP -> H323
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.
Thomas Haeger
2003-May-21 08:38 UTC
AW: [Asterisk-Users] Segmentation fault on using SIP -> H323
Hi again, it's only when i use chan_oh323. If i use chan_h323, it works. But, why exist two channel drivers ? Regards, Thomas. -----Urspr?ngliche Nachricht----- Von: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]Im Auftrag von Thomas Haeger Gesendet: Mittwoch, 21. Mai 2003 17:05 An: Asterisk User Betreff: [Asterisk-Users] Segmentation fault on using SIP -> H323 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. _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users