search for: progress_audio

...want to have sip connection between two asterisk systems (145 and 146). connection from 145 to 146 is ok but i can not call from 146 to 145. this is h323.conf file in 145: [peer146] host=192.168.0.146 type=friend context=from-trunk [to-146] type=peer host=192.168.0.146 faststart=yes tunneling=no progress_audio=yes disallow=all allow=alaw allow=ulaw this is mu extensions.conf file in 145: [from-trunk] exten=>_1.,1,Dial(SIP/to-231/1${EXTEN:1}) [line-231] exten=>_2.,1,Dial(H323/to-146/2${EXTEN:1}) i have this error: dropping call because extensions '100', 's' and 'i' doesn...

...neral] port=1720 bindaddr=my.ipaddr dtmfmode=rfc2833 gatekeeper=DISABLE AllowGKRouted=no context=default [termination] type=peer host=some.ipaddr noFastStart=no noH245Tunneling=no noSilenceSuppression=yes progress_setup=3 progress_alert=8 progress_audio=yes disallow=all allow=ulaw Any Suggestion is welcome. Thanks! __________________________ Laurent TOSTAIN

Greetings, I have a problem making a call from Asterisk to Cisco H323 PSTN gateway using H323 channel. I can call but there are no sound in both way. If I call H323 gateway directly from SJPhone I have no problem with sound. Any advice are welcome. Thanks in advance.

...kwared voice path. Valid values are: ; 0 - don't add PROGRESS IE ( default); ; 8 - in-band information or an appropriate pattern is now available; ;progress_alert = 8 ; ; Generate PROGRESS message when H.323 audio path has established to create ; backward audio path at other end of a call. ;progress_audio = yes ; ; Specify how to inject non-standard information into H.323 messages. When ; the channel receives messages with tunneled information, it automatically ; enables the same option for all further outgoing messages independedly on ; options has been set by the configuration. This behavior is re...