[Asterisk-Users] Clipping on outbound calls via SIP/IAX

[Reid Forrest]

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.
rforrest at max-is.net
ofc: 407.786.9600 x1200   cell: 321.439.8903