Displaying 6 results from an estimated 6 matches for "sw6".
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2004 Jun 24
1
ZyXEL Prestige 2000W and DTMF
...400]
type=friend
context=from-sip
username=400
secret=verysecret
disallow=all
allow=g729
dtmfmode=rfc2833
host=dynamic
nat=yes
qualify=300
canreinvite=no
My phone is set to use DTMF 'outband'
any ideas?
Dominique
--
taridium.communications
dominique kull, partner
the old lodge, london sw6 6ee uk
t: +44 207 731 1562
f: +44 207 900 6564
v: fwd 268167
w: http://taridium.com
e: dk@taridium.com
2004 Jun 18
2
C7960 g729 question
I have multiple voiceage g729 licenses installed on a RH9 box, and have
a remote C7960 configured to use it (low bandwidth). In calls like:
Remote C7960 -> g729 -> asterisk -> g711 -> C7960
the audio is oftentimes rather choppy. Changing the remote 7960 to use
g711 seems to eliminate/reduce the choppyness. Any ideas on what might
be behind this?
2004 Jun 24
2
R: R: R: How to force G729
> "If" I understood your initial objective correctly (and I may not have),
> the user's phones are negotiating the codec to be used for each rtp session.
>
> Asterisk parameters can be used to dictate rtp sessions between the sip
> phone and asterisk, but that won't influence the next step in which the sip
> phone negotiates a new rtp session directly with the
2004 Jun 26
2
ZyXEL Prestige 200w - should I return it ?
Hi all
I have just got a P2000w and experience several problems. Hopefully there is
someone out there that has got it working. I saw it on Cebit and the person
demonstrating it there told me that it was connected to an Asterisk server
on the stand -so it should work.
Problem 1: it does not register correctly
It get lots of messages like this:
Jun 26 19:45:19 NOTICE[1107585968]: chan_sip.c:5630
2004 Jun 02
5
ZyXEL Prestige 2000W SIP hangup fails
Does anybody have any experience with the ZyXEL Prestige 2000W? I am
having problems with the line tear down when I call another extension.
If nobody picks up at the other end when I hangup the 2000W, the other
extension continues to ring. Is there any way to hangup a SIP call if
there is no more traffic? Asterisk seems to think that there is still a
connection open. This is pretty annoying
2009 Jul 23
1
[PATCH server] changes required for fedora rawhide inclusion.
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