On 12/31/09 13:06, Kevin P. Fleming wrote: >Joseph wrote: >> I'm having problem passing Caller ID to asterisk from AudioCodes MP-114 (FXO) >> >> AudioCodes is passing the caller ID to Asterisk but Asterisk is trying to >> interpret it as authentication: >> >> [Dec 31 11:41:07] WARNING[9752]: chan_sip.c:8553 check_auth: username >> mismatch, have <pstn-5665>, digest has <pstn-1270> >> [Dec 31 11:41:07] NOTICE[9752]: chan_sip.c:14316 handle_request_invite: >> Failed to authenticate user "KMIEC Z" >> <sip:7804715...@10.0.0.157>;tag=1c354211286 >> >> Calls go through but not Caller ID. >> Any suggestions? > >Asterisk does not fully support domain authentication yet, so the >'username' present in the From header is used for authentication *and* >Caller ID. That means that if you want proper Caller ID to be extracted >from a SIP INVITE, you can't request authentication on that INVITE. > >If you configure the SIP user/peer that you are using for that gateway >with 'insecure=invite', Asterisk will accept INVITEs from it without >requiring authentication.
type=peer insecure=invite I get the same and in addition call is not even forwarded to asterisk, it just keeps ringing. [Dec 31 14:42:34] WARNING[13715]: chan_sip.c:8553 check_auth: username mismatch, have <pstn-5665>, digest has <pstn-1270> [Dec 31 14:42:34] NOTICE[13715]: chan_sip.c:14316 handle_request_invite: Failed to authenticate user <sip:pstn-1...@10.0.0.157>;tag=1c1796801183 -- Joseph _______________________________________________ -- 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
