On Sun, 18 Jan 2009, Lyle Giese wrote: >Joseph wrote: >> We have a caller ID from our phone provider "Shaw Cable" (digital phone) and >> it was working OK until recently. >> I get an error: >> >> WARNING[6769]: chan_sip.c:8553 check_auth: username mismatch, have <4>, >> digest has <pstn-4444> >> NOTICE[6769]: chan_sip.c:14316 handle_request_invite: Failed to authenticate >> user THELMA >> <sip:7804789...@10.10.0.103>;tag=50e17675d59121c4o1 >> >> at this point call fails, it is not being passed through to asterisk. >> >> I'm using Linksys 3102, PSTN answer delay is set to 3sec. to allow for >> caller ID to pass through. >> When I decrease timing to 1sec. or eliminate it 0sec the call goes through >> but there is no caller ID being forwarded. >> >> It was working OK for a while. So I'm not sure if Shaw Cable have upgraded >> something on their >> digital phone or there is a problem with asterisk/ >> >> <4> is a Line1 >> <pstn-4444> is PSTN Line >> >> >Have you tried to extend that delay to 5 or 6 seconds? It's possible >that caller id is being sent a second or two later/longer, but your 3 >seconds is now cutting off a portion of the caller id data. > >Lyle
Extending delay to 6sec. didn't help. It is a SIP bug I believe and some temporary work around I was able to find on the net are: insecure=invite insecure=very insecure=port,invite Previously I had insecure=port but it stop working; changing it back to: insecure=invite solved the problem. -- #Joseph GPG KeyID: ED0E1FB7 _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
