search for: rychtecky

...Host Mask Port Status SERVER1 xxx.xxx.xxx.xxx (D) 255.255.255.255 9785 (T) OK (29 ms) SERVER2 xxx.xxx.xxx.xxx (D) 255.255.255.255 4569 OK (95 ms) 2 iax2 peers [2 online, 0 offline, 0 unmonitored] thanks, Marian -- Marian Rychtecky marian@mari.cz Tel. +420 724 397 441 ICQ 76582857

...IAX Subclass: NEW Timestamp: 00013ms SCall: 00004 DCall: 00000 [213.160.177.186:4569] VERSION : 2 CALLED NUMBER : 1299 CODEC_PREFS : () CALLING NUMBER : 1199 CALLING PRESNTN : 0 CALLING TYPEOFN : 0 CALLING TRANSIT : 0 CALLING NAME : Marian_Rychtecky LANGUAGE : en USERNAME : <some_username> FORMAT : 2 CAPABILITY : 2097151 ADSICPE : 2 DATE TIME : 2006-10-18 10:16:14 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 00006ms SCal...

All, Has anyone seen INVAL messages on an IAX link before? I'm occasionally getting them from my Gateway provider, and I need to narrow down the potential cause. Symptoms are: Incoming calls fail, I see NEW, AUTHREQ then INVAL messages between the two A*k boxes... then for no reason at all it'll start working ok again.. My Asterisik: 1.2.10, Gateway A*k : 1.2.0

hello all, isit possible to do sth. so that if the sip-phone is in use and a call is incoming that the caller gets a busy signal? coz i wan to have the incoming party get a busy signal if i'm at phone. regards rene -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061120/6f438f28/attachment.htm

Hello all, iam setting up an asterisk box behind NAT to get SIP calls from outside or internet. In that eschema i can setup SIP calls but, while from the outside nat people can hear me, Im unable to listen anything behind NAT. Out of firewalls settings( I checked this to port fowarding) what can i do to get this working fine?. Thanks G.