All, I've almost got my Asterisk PBX setup, but I've having some problems with the VoicePulse IAX trunk.
On outbound calls, when dialing a PSTN number through the IAX2 trunk, music on hold (moh, using the m option in the dial command) does not work. The console states that "stop sound" on IAX2 channel. Ring works, but only without the r option. MOH works when trying to dial a non-PSTN terminated IAX2 calls (e.g. a softphone). I've read that with SIP connetions, the originating line is not held open by the PBX, so the can be no timing sync with the client, but I don't know if that's also the case here. The setup I have is: [sip softphone Xten] ==> [ * ] ==> [IAX2 VoicePulse Trunk] => [PSTN Number (SprintPCS Cell)] The relevant iax.conf sections are: [voicepulse] context=voicepulse-incoming dtmfmode=rfc2833 secret=mysecret auth=md5 type=user host=gw5.voicepulse.com [voicepulse-peer] qualify=yes trunk=yes dtmfmode=rfc2833 secret=mysecret auth=md5 type=peer host=gw5.voicepulse.com My extensions.conf has: TRUNK=IAX2/[EMAIL PROTECTED] exten => 15,1,Playback(transfer) exten => 15,2,Dial(IAX2/ckm,20,rt) exten => 15,3,VoiceMail(u${EXTEN}) exten => 15,4,Hangup exten => 15,103,Dial(${TRUNK}/14155551212,30,t) exten => 15,104,VoiceMail(u${EXTEN}) exten => 15,105,Hangup Any ideas, bug? Thx. Chris. _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users