Hi List, Additional with the latest tries from the below I get a nice random seg fault when I hangup on PSTN. (With obviously no sound on x-lite, still!)
asterisk -vvvvgc results after hanging up the pstn line in: -- Executing Hangup("SIP/1087997-d79f", "") in new stack == Spawn extension (sip-phone-out, h, 2) exited non-zero on 'SIP/phonenumber-d79f' Segmentation fault > Since there is no normal release cycle can somebody give us advise which asterisk/X-Lite/chan_capi versions work well together ? (date and time of CVS version) Thanks in adavnce, Thorsten --------------------------------------------------------------- Hi all, Still having the one way sound problem. Any suggestions how to hunt the problem down ? Regards, Thorsten --------------------------------------------------------------- Hi all, We have a very basic * installation for testing purposes. The * is connected to PSTN with BRI and setup with X-Lite over plain lan. (local IP's) OS: Linux/Debian unstable. Asterisk CVS-10/29/03-23:46:26 chan_capi On the IP side: X-lite (build: 1084) Calling and get calls on PSTN from X-Lite is no problem. We only get sound from PSTN to X-lite. Never from X.-lite to PSTN. The soundmeter on X-lite shows activity ... (not muted, correct device...) When pressing numbers while having these silent calls in x-lite is playing DTMFs at the PSTN phone side. sip.conf: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to allow=all [1*phonenumber*] type=friend username=NAME secret=testpass auth=md5 nat=no host=dynamic reinvite=no dtmfmode=inband callerid="Test" <*phonenumber*> context=sip-phone-out Any suggestions ? Thanks, Thorsten _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users