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. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com & www.asterisk.org _______________________________________________ -- 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
