I'm hoping someone can help me with a problem I've been having for a while now. I've googled and wiki'd to no avail.
Whenever I place an outbound call from * to a PSTN through a SIP or IAX provider (e.g. Voicepulse or Broadvoice), the first 1/2 to 2 seconds of the remote call are clipped (muted). For example, if I call a remote voicemail system that usually answers with "Nortel Call Pilot, Mailbox?" I might get "ilot, Mailbox?". Everything works fine if I dial an internal extension or through the PSTN. Is this just something I'm going to have to live with if using an Internet-based termination provider? I'm using Asterisk 1.0.3 and have tested on different systems, different providers, different phones, etc. Thank you, Reid Forrest, CISSP Max-IS, Inc. [EMAIL PROTECTED] ofc: 407.786.9600 x1200 cell: 321.439.8903 _______________________________________________ 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
