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

2005-01-03 Thread Jens Vagelpohl
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  [EMAIL PROTECTED]
Software Engineer   +49-(0)441-36 18 14 38
Zetwork GmbHhttp://www.zetwork.com/
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


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

2005-01-03 Thread Claus Futtrup
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 [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@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 [EMAIL PROTECTED]
Software Engineer +49-(0)441-36 18 14 38
Zetwork GmbH http://www.zetwork.com/
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users