I'm pretty sure the info has been posted to the mailing list several
times and should be in the searchable archives.
On Wed, 2003-09-10 at 14:28, Peter Pauly wrote:
> On Tue, Sep 09, 2003 at 02:38:01PM -0500, Eric Wieling wrote:
> > That would be reinvite= and canreinvite= in the user entry for ea
On Tue, Sep 09, 2003 at 02:38:01PM -0500, Eric Wieling wrote:
> That would be reinvite= and canreinvite= in the user entry for each SIP
> endpoint. Asterisk will allow the endpoints to talk directly to each
> other if both those settings are = yes (the default, I think) AND both
> endpoints use th
From: "Steven Critchfield" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, September 10, 2003 12:01 PM
Subject: Re: [Asterisk-Users] Endpoint-to-Endpoint RTP Packets
> On Wed, 2003-09-10 at 10:42, Olle E. Johansson wrote:
> > Eric Wieling wrote:
> &
On Wed, 2003-09-10 at 10:42, Olle E. Johansson wrote:
> Eric Wieling wrote:
>
> > That would be reinvite= and canreinvite= in the user entry for each SIP
> > endpoint. Asterisk will allow the endpoints to talk directly to each
> > other if both those settings are = yes (the default, I think) AND
Eric Wieling wrote:
That would be reinvite= and canreinvite= in the user entry for each SIP
endpoint. Asterisk will allow the endpoints to talk directly to each
other if both those settings are = yes (the default, I think) AND both
endpoints use the same protocol (SIP) AND the same codec.
I tried
On Tue, 9 Sep 2003, Eric Wieling wrote:
> It would have to do some kind of trascoding,
Forgive my ignorance, but why? PSTN is delivering 8 bit 8 KHz
ulaw samples. G711 is delivering 8 bit 8 KHz ulaw samples over
SIP. Aren't the two data streams identical down to the bit
level?
--
Mike Ci
Hi all,
I'm interested in using asterisk WITHOUT codec support: I work in a LAN, with no
bandwidth, delay, ... problems; I use a Cisco GW as PSTN interface and when I
use asterisk the overall delay is to high and the quality drops.
In particular, I'm interested in using asterisk as h323 to sip
It would have to do some kind of trascoding, but it's a non-issue since
G729 is not involved and the CPU overhead is minimal.
On Tue, 2003-09-09 at 15:26, Mike Ciholas wrote:
> On Tue, 9 Sep 2003, Eric Wieling wrote:
>
> > Transcoding would be required for access to ANY of the asterisk
> > sound
On Tue, 9 Sep 2003, Eric Wieling wrote:
> Transcoding would be required for access to ANY of the asterisk
> sound files, voicemail and PSTN via Zap interfaces.
If you are using G711 ulaw from the SIP phones, and that is what
you are getting from the T1 PSTN link, would * have to transcode
that?
Codecs are always an issue. Best to put disallow=all and
allow=whatevercodecyouwant in each [sipuser] entry. You can't have
Asterisk do codec translation (transcoding) bewteen g729 and some other
codec unless you have the g729 licenses (US$10/channel from Digium).
Transcoding would be required f
Subject: Re: [Asterisk-Users] Endpoint-to-Endpoint RTP Packets
> That would be reinvite= and canreinvite= in the user entry for each SIP
> endpoint. Asterisk will allow the endpoints to talk directly to each
> other if both those settings are = yes (the default, I think) AND both
> en
At 02:38 PM 9/9/2003 -0500, you wrote:
That would be reinvite= and canreinvite= in the user entry for each SIP
endpoint. Asterisk will allow the endpoints to talk directly to each
other if both those settings are = yes (the default, I think) AND both
endpoints use the same protocol (SIP) AND the s
That would be reinvite= and canreinvite= in the user entry for each SIP
endpoint. Asterisk will allow the endpoints to talk directly to each
other if both those settings are = yes (the default, I think) AND both
endpoints use the same protocol (SIP) AND the same codec.
On Tue, 2003-09-09 at 13:04
t; <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, September 09, 2003 2:40 PM
Subject: Re: [Asterisk-Users] Endpoint-to-Endpoint RTP Packets
> I wonder if your company took your name off of the support contact page
> because you do not know how to properly use your e
I wonder if your company took your name off of the support contact page
because you do not know how to properly use your email software yet.
Can you justify your abuse of the reply button where you only changed
the subject line and didn't even remove the non relevant message below?
Stupid, 'tar
I have seen this asked in the archives several times, but do not see a
definitive answer anywhere. Is there a way to tell the Asterisk to act like
a "normal" SIP Proxy, handling only the SIP messages, and letting the RTP go
point-to-point?
Sean
___
Sean
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