. when the
other party hangs up, BYE will be sent to the new IP.
in my setup, asterisk still sends BYE to the old IP.
Is this something we can already do? or possible to add?
Kelvin Chua
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wild guess would be a conflict on host= setting.
there might be another entity on your sip.conf which have type=friend
and host=callcentric.com or host=204.11.192.161
Kelvin Chua
On Mon, Apr 14, 2014 at 8:01 AM, Sean Darcy seandar...@gmail.com wrote:
On 11.9, trying to set up a callcentric peer
Hi guys,
does res_jabber support realtime? if not, are there any plans to?
Kelvin Chua
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New to Asterisk? Join us for a live introductory webinar every
has anybody made waitfordialtone in chan_dahdi.conf work outside the UK?
callprogress=yes
progzone=us
waitfordialtone=1000
i keep getting:
WARNING[3859]: chan_dahdi.c:7111 dahdi_read: Never saw dialtone on channel 1
i'm using asterisk 1.8.4.4
Kelvin Chua
as to avoid detection of false hangups. i think
this is not too hard as Newman Ventures implemented
something like this before.
Kelvin Chua
On Fri, Aug 26, 2011 at 11:51 AM, Shaun Ruffell sruff...@digium.com wrote:
On Fri, Aug 26, 2011 at 10:58:10AM +0800, Kelvin Chua wrote:
I should clarify
waitfordialtone=yes on chan_dahdi.conf is supposed to be the perfect
solution, but does it work on on UK lines?
Kelvin Chua
On Fri, Aug 26, 2011 at 4:44 PM, Kelvin Chua kel...@gmail.com wrote:
i am comparing the experience of using an analog span to a T1 for example:
if i have a 3 quad port
i think this also happens with cisco callmanager way back using h323.
this is fixed (as far as callmanager is concerned) by a patch submitted
to the mailing list a few months back by marian durkovic (search the
archive). i don't think that patch reached the cvs though... or did it?
On Thu,
uhm, strange but does this work on your setup? even with permit and
deny, if a user is not matched in the conf, it is allowed access to the
default context stated in the conf.
On Wed, 2004-04-28 at 16:12, James H. Thompson wrote:
I think the problem is that using permit= alone does nothing.
this is something i just recently noticed.
have you found any info on how to manage incoming calls through
chan_h323? it doesn't seem to match any entity you define, it always
uses the default context...
On Sat, 2004-01-24 at 02:39, Fran Boon wrote:
Some of you may remember seeing my issue using
why not make * call straight to your cisco?
in my implementation, i made * call sip straight to a 3640 and
everything's ok
On Thu, 2004-04-15 at 01:42, Kyle Stone wrote:
I've got an Asterisk to H323 bridge working... but I'm having a few
problems..
I got everything working by setting up
this patch however worked for me, all calls through the patched
chan_h323 are ok, hold, transfer, etc works perfectly. except that there
is no music on hold, while in fact asterisk shows that it is playing,
yet there is no audio heard on the callmanager side.
so i had test on both oh323 0.5.10
- Original Message -
From: Michael Manousos [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, October 09, 2003 9:25 PM
Subject: Re: [Asterisk-Users] 5 second latency sip to oh323
How do you transfer the call?
Michael.
Kelvin Chua wrote:
hi guys,
i'm using sept 30 cvs
hi guys,
i'm using sept 30 cvs and oh323 5.5
i'm having 5 second latecy(on only 1 audio path)
when a call is transferred
the scenario is this:
sip-asterisk-h323:operator (who
then transfers the call)
h323:destination
* compiled from cvs, i am trying callip
phones in callmanager 3.2
10.17.0.2 is my callmanager
i noticed from network dumps that instead of
sending rtp to the ip phone,* sends it to 10.17.0.2!
thereby causing no audio from* to ip phone.
audio from ip phone to* is ok.
only callmanager calls
jeremy
FYI, people have reported that asterisk-oh323 works fine
with CCM (haven't tested that myself).
Michael.
Kelvin Chua wrote:
* compiled from cvs, i am trying call ip phones in callmanager 3.2
10.17.0.2 is my callmanager
i noticed from network dumps that instead of sending rtp
Manager, because I
haven't been able to dedicate enough time to make native bridging work.
Hell, maybe chan_skinny is the best way to interface CCM to Asterisk.
Only if I had a non-production CCM to play with and more time.
Jeremy
Kelvin Chua wrote:
* compiled from cvs, i am
guys,
what firmware version of audiocodes mp104 fxs are
you using with asterisk?
i'm having sip stack problems.
~kelvin
our university is going to roll-out 1000 lines in the next few months. we
are going to deploy either quintum, audiocodes, vg248 or ata-186 around
campus (and soon maybe grandstreams and cisco). we have a cisco callmanager
to do the call routing and asterisk for voicemail and protocol conversion.
hi guys,
do you have any suggestions on how to connect 2 TE410P via E1? (for
simulation and testing purposes)
asterisk1 -- TE410P ?-? TE410P
--asterisk2
, September 08, 2003 12:52 PM
Subject: Re: [Asterisk-Users] how to connect 2 TE410P
On Sun, 2003-09-07 at 23:24, Kelvin Chua wrote:
hi guys,
do you have any suggestions on how to connect 2 TE410P via E1? (for
simulation and testing purposes)
asterisk1 -- TE410P
i also encountered this problem
i'm not too sure either but i don't think codec has to do anything with it
for i tried mix and matching but to no avail.
so for the meantime, try adjusting the tos for oh323 and i think you could
live with it
by the way, are you running cvs?
- Original
]
To: [EMAIL PROTECTED]
Sent: Monday, August 18, 2003 11:30 AM
Subject: Re: [Asterisk-Users] Chan_h323 one way audio
not sure what you mean by 'are you running cvs'?
What does the TOS setting do?
Regards,
Steven Thomas
Kelvin Chua
hi guys,
i'm encountering one way audio on cvs using
netmeeting and chan_h323.so
is there a quick fix or workaroundfor
this?
compiled using
openh323 1.12
pwlib 1.5
i also saw this in earlier version of openh323 and
pwlib
thanks for any info
~kelvin
, August 08, 2003 2:02 PM
Subject: Re: [Asterisk-Users] h323 and cvs one way audio
I used the exact versions listed in the readme for chan_h323 and it works
fine. Slackware and RH8 and 9.
bkw
On Fri, 8 Aug 2003, Kelvin Chua wrote:
hi guys,
i'm encountering one way audio on cvs using
there's this one way audio problem using h323
(CVS)with cisco callmanager?
has anybody encountered this problem? oh323 works
ok though...
or is there any workaround for this?
thanks
hi guys,
i'm in need of several 24port or higher fxs device
which supports sip, aside from mediatrix and audiocodes (cisco's vg248 doesn't
support sip), do you have any idea who else manufactures such
device?
yes, i agree, we never really felt the need to use unity, *'s vm is
functionally ok with callmanager
(except for the message waiting indication, or is there?) can *'s vm send a
MWI to the callmanager?
- Original Message -
From: Siggi Langauf [EMAIL PROTECTED]
To: Asterisk user list
hi guys,
have anybody tried using audiocodes sip fxs against
asterisk? how's the device fairing?
~kelvin
do you have any technical specification of this dlink sip phone? or
pictures? links? i can't seem to find any related literature on this. thanks
- Original Message -
From: Greg Renouf [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, July 17, 2003 5:18 AM
Subject: Re:
windows xp have msn messenger by default, and it supports sip.
- Original Message -
From: lists [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, July 14, 2003 6:23 AM
Subject: Re: [Asterisk-Users] Question #3
Voice only would be fine,
I don't care about the protocol, I just want
hi guys! i'm going to share a workaround
forauthentication from msn messenger, you have to change two lines in
chan_sip.c
msn messenger is known to look for the correct
realm in authentication, therefore, change the realm in chan_sip.c, line 2061
and line 2910 (release 0.4.0)
i hopethe
hi guys,
have any of you guys managed to usemsn
messenger to authenticate with asterisk using its DNS name? based on my
experience with other sip proxies, msn will not authenticate if it sees a
different realm than the realm of the client. one workaround i did was to edit
the chan_sip.c to
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