Hi all, I have installed asterisk latest stable version 1.6.1.0, with dahdi driver (tdm410p). then i try to use my older 1.4 extensions.conf. . now it wont work with 1.6.
I managed to register my phone on asterisk. but i cant hear any dial tone on my phone. these are my configs. it will detect incoming calls and transfer the call to ext 312. but sip phone users voice is not clear..., but sip phone user can hear the other party (PSTN) very clearly. please help me to solve the issue. all work on asterisk 1.4. [general] port = 5060 bindaddr = 0.0.0.0 context = sip disallow=all allow=all ;allow=g729 ;allow=gsm allow=alaw allow=ulaw transfer=yes tos=lowdelay dtmfmode = rfc2833 [312] type=friend ; Friends place calls and receive calls context=sip2 ; Context for incoming calls from this user secret=312 host=dynamic ; This peer register with us dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info username=312 ; Username to use in INVITE until peer registers mailbox=312 qualify=yes disallow=all pickupgroup=1 allow=all ;allow=alaw ; dtmfmode=inband only works with ulaw or alaw! ;allow=gsm ;;canreinvite=no ;;progressinband=yes ;;reinvite=no ;;callerid=tharanga <312> extensions.conf channel.dadhi.conf [channels] signalling=fxs_ks ;toneduration=100 callwaiting=yes threewaycalling=yes callreturn=yes echocancel=128,param1=32,param2=0,param3=14 echocancelwhenbridged=yes echotraining=yes echotraining=800 busydetect=yes busycount=2 hanguponpolarityswitch=yes ringtimeout=8000 group=1 context=sip immediate=yes jitterbuffers=4 jbenable = yes echocancel=yes channel=>1-4 ;overlapdial=yes ;pulsedial=yes dtmfmode=rfc2833 ;relaxdtmf=yes ;rxgain=10.0 ;txgain=8.0 Many thanks Tharanga _______________________________________________ -- 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