didn't work :(
Regards,
Santiago
On 9/20/06, Alyed Tzompa <[EMAIL PROTECTED]> wrote:
Not an expert at reading Polycom config files, but guess g729 and ulaw are
both preference 1 isn't it?
hey... you have in your sip.conf configuration "canreinvite=no"... think
this may be a problem: since A
Sorry but I've ran out of ideas...Anyone else out there with a successful Polycom g729 pass through-only experience?Alyed
Return-Path: <[EMAIL PROTECTED]> Thu Sep 21 11:27:21 2006Received: from nz-out-0102.google.com [64.233.162.206] by maila11.webcontrolcenter.com with SMTP;
Not an expert at reading Polycom config files, but guess g729 and ulaw are both preference 1 isn't it?
hey... you have in your sip.conf configuration "canreinvite=no"...
think this may be a problem: since Asterisk will always stay in the
path of the RTPs, I think it might need to have the pr
Hi, I enabled sip debug and i get the following when i am trying to
call a polycom phone with the same sip.cfg I sent before (with g729 as
the primary codec):
--- (15 headers 9 lines)---
Using INVITE request as basis request - [EMAIL PROTECTED]
Sending to 10.14.34.130 : 5060 (NAT)
Found user '202
Still having the same problem. i modified the sip.cfg in order to make
g729 the first choice:
Cheers,
Santiago
On 9/19/06, Alyed Tzompa <[EMAIL PROTECTED]> wrote:
Make sure the codec used by the Polycom will be only g729 via the phone's
web interface, as far as I remember Polycom w
Make sure the codec used by the Polycom
will be only g729 via the phone's web interface, as far as I remember
Polycom will try always to use ulaw or alaw first unless it is
configured to use only or as first choice the g729 codec.Alyed
Return-Path: <[EMAIL PROTECTED]> Tue S
