On 28 Dec 2008, at 18:36, Razza wrote: > Please see below Console Messages, Pertinent section of SIP.CONF and > AudioCodes Config. > > Console Messages: > Dec 28 18:14:45 NOTICE[19109]: chan_sip.c:9808 > handle_request_register: Registration from '<sip:[email protected]>' > failed for '192.168.10.4' > Dec 28 18:14:45 NOTICE[19109]: chan_sip.c:9808 > handle_request_register: Registration from '<sip:[email protected]>' > failed for '192.168.10.4' > -- Saved useragent "Audiocodes-Sip-Gateway-MP-114 FXS_FXO/v. > 5.00A.035.003" for peer 272 > -- Saved useragent "Audiocodes-Sip-Gateway-MP-114 FXS_FXO/v. > 5.00A.035.003" for peer 271 > > > SIP.conf (pertinent section): > ;----------------- MP114 Play ----------------------- > [pstn1] > ;MP-114 FXO Port 1 > type=friend > username=276 > secret=mp276 > context=home_phones > allow=alaw > dtmfmode=inband > host=192.168.10.4 > nat=never > canreinvite=no > > [pstn2] > ;MP-114 FXO Port 2 > type=friend > username=277 > secret=mp277 > regextn=277 > context=home_phones > allow=alaw > dtmfmode=inband > host=192.168.10.4 > nat=never > canreinvite=no >
[audiocodes] username=audiocodes type=peer secret=NotTellingYou qualify=no host=123.123.123.123 dtmfmode=rfc2833 disallow=all context=from-provider-someone allow=alaw allow=ulaw insecure=port,invite That is what I use for mine (granted its a PRI gateway but I bet the SIP code is similar). _______________________________________________ -- 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
