search for: sw6

Displaying 6 results from an estimated 6 matches for "sw6".

Did you mean: sw
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.
...>v=&CxQdE at xWf=Oq<%%9Psr2`!_^x$43e|OW=6X44u9H*U zXc#ROOt;2Ja1_M_uA*qol4;>0xI!)rtPpEJ%Vb&eWx?DhF6va)qXsWY$|8PeZSY3x z#^`6f<VGoeBQ!1IJ$xQ^39%pU$6=s^&qL{Q3J|^^{FH~~@OLV5j;O9%N>-4?Ge3~U z5ig4r3MR4;FaKT^Xf5&vM%3s9b+X-TD at 2yjAIS2smt_$X8RGGS*+tag1w9e{PWNBX zrSP})O1M(ik=+hSw6pL!1gFdX>2ys>O)`2>?@^nK){)PM8l=c7iuTdlwoOIUk|IsC z%6u8aIYcH;M4jOX7nKxeauteKXNOQq1m0rr&V-USy8|s`whxxWJkb5+k`x`S7s=Bv z3mt>d8Toh6nrC__ at 3Dt+N@woMQ$Kg*$?Z6a3=3nGOcuJW at vjy=tUncau}lVK=J1ch zUU^^7*f;7*L&|W5oKFkIP)F_YM9BE)pZPH$<HH+O_zcaE3u7~6m$|hW%&ig3(T?ib za$vSt7hNlOEJ%-n%...