Hi list, why I cannot hear voice when I call from a SIP telephone (Budgetone and others) to a H323 telephone (several models)? could anybody please give any idea to solve this issue? Please, let me know. Thanks in Advance. N.B. The configuration for "extensions.conf", "sip.conf" and "h323.conf" files are: *************************************** extensions.conf >>>> *************************************** [default] ..... ..... [outgoing] exten=>_7XX,1,Goto(voip-h323|${EXTEN}|1) exten=>_*XX,1,Goto(servicios|${EXTEN}|1) exten=>_XXXXXXXXX,1,Dial(Zap/1/${EXTEN}|30) exten=>_XXXXXXXXX,2,Playback(invalid) exten=>_XXXXXXXXX,3,Hungup() exten=>_X,1,Playback(invalid) exten=>_X,2,Hungup exten=>_XX,1,Playback(invalid) exten=>_XX,2,Hungup exten=>_XXXX,1,Playback(invalid) exten=>_XXXX,2,Hungup exten=>_XXXXX,1,Playback(invalid) exten=>_XXXXX,2,Hungup exten=>_XXXXXX,1,Playback(invalid) exten=>_XXXXXX,2,Hungup exten=>_XXXXXXX,1,Playback(invalid) exten=>_XXXXXXX,2,Hungup exten=>_XXXXXXXX,1,Playback(invalid) exten=>_XXXXXXXX,2,Hungup exten=>i,1,Playback(invalid) exten=>t,1,Hungup [voip-h323] ; SIP:: exten=>701,1,Dial(SIP/701) ; SIP:: exten=>702,1,Dial(SIP/702) ; H323:: exten=>703,1,Agi(AceptaLlamada.php) exten=>703,2,Dial(h323/3|17|tTm) exten=>703,3,VoiceMail(u703) exten=>703,103,VoiceMail(b703) ; H323:: exten=>710,1,Agi(AceptaLlamada.php) exten=>710,2,Dial(h323/10|17|tTm) exten=>710,3,VoiceMail(u710) exten=>710,103,VoiceMail(b710) ...... ********************************* sip.conf >>>> ******************************** [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = outgoing ; Default for incoming calls srvlookup = yes ; Enable SRV lookups on outbound calls ;pedantic = yes ; Enable slow, pedantic checking for Pingtel tos=lowdelay ;tos=184 ;maxexpirey=3600 ; Max length of incoming registration we allow ;defaultexpirey=120 ; Default length of incoming/outoing registration ;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY ;videosupport=yes ; Turn on support for SIP video disallow=all ; Disallow all codecs allow=alaw allow=ulaw ; Allow codecs in order of preference ;allow=ilbc [701] type=friend username=701 fromuser=701 secret=701 host=dynamic defaultip=192.168.0.151 ;mailbox=701 context=outgoing canreinvite=yes dtmfmode=info callgroup=1 pickupgroup=1 [702] type=friend username=702 fromuser=702 secret=702 host=dynamic defaultip=192.168.0.152 mailbox=702 context=outgoing canreinvite=yes dtmfmode=info callgroup=1 pickupgroup=1 ..... *************************************** h323.conf *************************************** [general] port = 1720 bindaddr = 0.0.0.0 tos=lowdelay ; amaflags=billing ; disallow=all ; turns on all installed codecs ;disallow=g723.1 ; Hm... Proprietary, don't use it... allow=gsm ; Always allow GSM, it's cool :) ;allow=ulaw allow=alaw ;allow=g729 ; noFastStart=yes noH245Tunneling=yes noSilenceSuppression=yes ; jitter=20 ; dtmfmode=inband ; gatekeeper = 192.168.0.207 ; AllowGKRouted = yes ; context=outgoing ; [CAC-IP] ;our computer. type=h323 prefix=9,7,*,8 context=outgoing ; -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031104/5d2ba009/attachment.htm