similar to: 5 second latency sip to oh323

* 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 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 fail. netmeeting works ok... here is the debug, thanks for any info ~kelvin H323 debug enabled --

hi michael, here are the core dumps. only kphone works when 0.5.5 and * cvs. audiocodes and msn messenger all cause seg faults when calling ccm thru * (or vice-versa) ~kelvin [chan_oh323.so] => (OpenH323 Channel Driver) == Parsing '/etc/asterisk/rtp.conf': Found == Parsing '/etc/asterisk/oh323.conf': Found 0:00.004 OpenH323 Wrapper OpenH323 Wrapper

Thanks Dragos, I assume I will be setting those parameters during initialization of encoder right? Question is, if connection gets too lossy, how will opus adapt to it? Can it automatically shift bitrate down to minimize impact? Mark from IRC suggests that the app has to be aware of the losses and change it on the fly. Has anybody on the list tried this? Kelvin Chua On Wed, Mar 4, 2015 at 5:53

hi guys, do you have any suggestions on how to connect 2 TE410P via E1? (for simulation and testing purposes) asterisk1 --> TE410P ----> ? ---------> ? ---->TE410P -->asterisk2 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030907/698cd499/attachment.htm

I am using libopus for my implementation. I wonder if anybody in the list have any experience on how to make libopus dynamically adjust its bitrate? On Mar 3, 2015 10:42 PM, "Benjamin Schwartz" <benjamin.m.schwartz at gmail.com> wrote: > It sounds like your software isn't adjusting the opus bitrate in response > to network conditions. For example, many WebRTC

hi guys, have anybody tried using audiocodes sip fxs against asterisk? how's the device fairing? ~kelvin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030725/ae4b2f25/attachment.htm

hi guys! i'm going to share a workaround for authentication 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 hope the realm can be parsed from extensions.conf in the next release... ~kelvin =) -------------- next part

hi guys, i'm encountering one way audio on cvs using netmeeting and chan_h323.so is there a quick fix or workaround for 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 -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have a .csv table named mailing.csv as below. It consist a receiver, subject and sender. Receiver subject sender 1 Adrian Cole RE: [WHIRR-117] Composable services Tom White 2 Adrian Cole RE: [WHIRR-117] Composable services Tom White 3 Adrian Cole RE: [WHIRR-117] Composable services Adrian Cole 4 Adrian Cole RE: [WHIRR-117]

Thanks. I already request as below. https://bugzilla.samba.org/show_bug.cgi?id=11923 -----Original Message----- From: Jeremy Allison [mailto:jra at samba.org] Sent: Thursday, May 19, 2016 2:54 AM To: Kelvin Yip <kelvin at icshk.com> Cc: samba at lists.samba.org Subject: Re: [Samba] Completely Disable NTLM on Samba4 On Wed, May 18, 2016 at 05:15:40PM +0800, Kelvin Yip wrote: > Dear

hi guys, have any of you guys managed to use msn 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 send a pre-defined realm, and also edit the Contact: field. after this, asterisk would send a 401 to the

Hi, I have a table with data, as below: dput(table): structure(list(Adrian.Cole = c(0L, 0L, 0L, 0L, 0L, 0L), Alison.Wong = c(0L, 0L, 0L, 0L, 0L, 0L), Andrei.Savu = c(0L, 0L, 0L, 0L, 0L, 0L), Bruno.Dumon = c(0L, 0L, 0L, 0L, 0L, 0L), Edward.J..Yoon = c(0L, 0L, 0L, 0L, 0L, 0L), Eugene.Koontz = c(0L, 0L, 0L, 0L, 0L, 0L), Jakob.Homan = c(0L, 0L, 0L, 0L, 0L, 0L), Kelvin.Kakugawa = c(0L, 0L,

Thanks Kelvin I'm a bit confised tho, is this patch already avaiable? if yes, what is the parameter that disable ssl into the smb.conf? Maybe the guys from Enterprise samba have already included the patch into their releases so it's just a maatter of enabling the flag. I'm using sernet-samba-4.2.2 Thanks!

Dear R Users Thanks for the help in advance and lets get straight to the problem: I have a 400 year long temperature time series and I am looking for decades that show a linear trend decrease of approximately -0.1 Kelvin or degrees. --> What I would like to program: A loop/function / command line that prints the values of all the trends (can also be overlapping) into a matrix that could have

Hi (cc'ed to r-devel where further discussion should probably take place) Thanks Lee. I see that problem. There is a "+ 1" in the Cairo device code for setting the clipping region (https://github.com/wch/r-source/blob/ba600867f2a94e46cf9eb75dc8b37f12b08a4561/src/library/grDevices/src/cairo/cairoFns.c#L156) Remove the "+ 1" and the problem goes away (for your example

On Tue, Mar 09, 2010 at 07:14:59PM -0500, Charles Lepple wrote: > On Mar 9, 2010, at 5:54 PM, Kelvin Ku wrote: > >> However, if I try to run usbhid-ups as user "nut", I get: >> >> $ sudo -u nut ./usbhid-ups -a usb > [...] >> Whereas if I run it as root: >> >> $ sudo ./usbhid-ups -a usb -u root > > Did you pass "--with-user=nut"

Hi Geoff, I'm a bit confused... After reading the docs, I was under the impression that icecast2 does not do on demand relaying? I did a google search, found this (well, it's a year ago) http://lists.xiph.org/pipermail/icecast/2004-February/006555.html Also, in the relaying section of the icecast documentation: http://www.icecast.org/docs/icecast-2.2.0/icecast2_relay.html There was no

Dear all, May I know if there is any way to completely disable NTLM and NTLM V2 on samba4 ? I need to ensure if someone bring their own workstations back to office and they cannot connect to samba4 server using their password. On Windows, there are a Security Settings to do this (Local Policies -> Security Options -> Network Security: Restrict NTLM: Incoming NTLM Traffic) Already

It is my understanding that a secure ldap connection should only send encrypted data yet my configuration is sending plaintext The following strace output from a smbpasswd results in the following: connect(4, {sin_family=AF_INET, sin_port=htons(636), sin_addr=inet_addr("<ldap server>")}}, 16) = -1 EINPROGRESS (Operation now in progress) ...snip... write(4,

Hi guys, I'm trying to move away from meetme to loose the dependency on dahdi. ConfBridge seems to be a good fit but I can't get it going. The document sounds like an easy to use app. Am I missing any bridge_ modules? Asterisk 1.6.2.0~rc2-0ubuntu1.2 -- Executing [521 at outbound:1] Answer("SIP/109-b877a8c8", "") in new stack -- Executing [521 at