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

Claus Futtrup cf at internetit.dk
Mon Jan 3 03:57:23 MST 2005


Hi

It could be that the RTP sessions aren't completely setup when you get 
connected to the destination.

Kind Regards

Claus Futtrup
----- Original Message ----- 
From: "Jens Vagelpohl" <jens at dataflake.org>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
<asterisk-users at lists.digium.com>
Sent: Monday, January 03, 2005 11:03 AM
Subject: Re: [Asterisk-Users] Clipping on outbound calls via SIP/IAX


> On Jan 2, 2005, at 21:16, Reid Forrest wrote:
>
>> 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.
>
> I have the same symptom, dialing from a phone hanging off a iaxy that 
> talks to my * and then outbound through NuFone. Where I expect to hear 
> "Thank you for calling foo..." I get "calling foo..." only.
>
> jens
>
> ---------------
>
> Jens Vagelpohl jens at zetwork.com
> Software Engineer +49-(0)441-36 18 14 38
> Zetwork GmbH http://www.zetwork.com/
>
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