search for: 6ee

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: