Displaying 8 results from an estimated 8 matches for "6ee".
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2004 Jun 24
1
ZyXEL Prestige 2000W and DTMF
...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
2007 Apr 30
0
[LLVMdev] Boostrap Failure -- Expected Differences?
...6e8 <__FUNCTION__.22568>:
> +000006e8 <__FUNCTION__.22481>:
> 6e8: 63 61 6e arpl %sp,0x6e(%ecx)
> 6eb: 6f outsl %ds:(%esi),(%dx)
> 6ec: 6e outsb %ds:(%esi),(%dx)
> 6ed: 5f pop %edi
> - 6ee: 74 72 je 762 <__FUNCTION__.22568+0x7a>
> - 6f0: 75 65 jne 757 <__FUNCTION__.22568+0x6f>
> + 6ee: 74 72 je 762 <__FUNCTION__.22481+0x7a>
> + 6f0: 75 65 jne 757 <__FUNCTION__.22481+0x6f>
&g...
2007 Apr 27
2
[LLVMdev] Boostrap Failure -- Expected Differences?
The saga continues.
I've been tracking the interface changes and merging them with
the refactoring work I'm doing. I got as far as building stage3
of llvm-gcc but the object files from stage2 and stage3 differ:
warning: ./cc1-checksum.o differs
warning: ./cc1plus-checksum.o differs
(Are the above two ok?)
The list below is clearly bad. I think it's every object file in
the
2005 Dec 21
9
question about changejournal
Hi,
I''ve got a newbie question--sorry if this is covered elsewhere, I parsed
through the archives for awhile and didn''t see it.
I''d like to listen for whenever a file is renamed (e.g. foo.txt -> foo.old)
and then magically change it back. This sounds odd, but I''m working with a
stubborn application and this will actually make things work nice.
So, if I do: