similar to: ISDN debugging and SIP dial-in issue]

Hi, my setup is quite simple: an asterix CVS of 2003-11-15 on a 2.4.21-debian-5 GNU/Linux box in an internal network (192.168.1.0/24, asterisk is 192.168.1.10). - with a SIP phone configured as 192.168.1.190, and with its SIP server being 192.168.1.190 - with an ISDN AVM c4 i4l card on an ISDN connection with 2 channels. I try to: - dial-in from ISDN, then transfer to the SIP

Hi there, I am trying to setup a configuration that requires me to use SIP and asterisk behind a firewall and over a VPN to a remote office and with some local Phones also. I can't use IAX to my provider because they don't offer it and my handsets ( snom 300 ) also don't support IAX so it's all SIP. The configuration is a follows Asterisk PBX 10.202.17.217/24 ------>|

I recently upgraded from a few month old CVS version of Asterisk to v1.0.1, and dialing out through my SPA-3000 stopped working. Notice right after INVITE, in the old CVS version, it includes the number I'm trying to dial (8019596) which works fine, however in v1.0.1, it doesn't include the number and of course the dial fails. Did a config option change out from underneath me or

Hi! I need some help Well i have this cenario: 1 ip04 running asterisk [A] 1 pc running asterisk [B] I nedd to make calls from A to B, and B to A. Via sip The A-B calls are working. Now I need to configure the dial plan to call B-A either to sip numbers and Fxs. Anyone can help me? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, i know i am asking a lot of questions lately in this forum..sorry about that first of all. :) Ok, here is the deal.. I am trying to make a hybrid system with an ericsson MD110 and asterisk. Internally we have 4 digit phone extensions on ericsson.. and so in asterisk. So, what i want to do is to call pbx side without adding 9 or etc to the begining of the number from asterisk clients.. For

Hi, Trying to figure out why I cannot get access to dell.com Their site is up because I can browse using a different firewall. Trying to find out where the logs are located and what log files it would write to if it were to deny browsing to a website. I can see the [UNREPLIED] when using the shorewall status. Was hoping to know what logfile it is writing it to. Thanks in advance, Elmer

Ivan, I found the same thing after doing an upgrade a couple of months ago and mailed this list. Never got a response. We had to fdisk the shared disks and remount them and restore db from backup. Chris Taylor Sr. Oracle DBA Unique Solutions www.unisolinc.com Office: 336-667-2447 xt. 2242 Cell: 336-262-5545 -----Original Message----- From: ocfs-users-request at oss.oracle.com

I was working with the rmetrics portfolioBacktesting function and dug into the code to try to find why my formula with 113 items, i.e. A1 thru A113, was being truncated and I only get 85 items, not 113. Is it due to a string length limitation in R or is it a bug in the strsplit or gsub functions, or in my string? I'd very much appreciate any suggestions ============Input script:

I am playing videos from my lubuntu computer to my tv through the hdmi interface on my video card. It mostly works. Some sound doesn't and I think it is related to the audio code. A52 Audio (aka AC3) (a52) works. It seems MPEG AAC Audio (mp4a) doesn't work. I checked a video that used to work and it doesn't work now. It has the mp4a codec also. The one that used to work shows this in

Is there a program currently available that will correctly mux a52/ac3 streams and theora streams into an ogg container? I've tried oggzmerge, and it will mux ac3 into ogm and correctly create a new ogm from an existing theora file, but it doesn't seem to put the two together correctly.

On 2018-05-01 06:00 PM, Ilia Mirkin wrote: > What GPU do you have? For video acceleration (i.e. va-api and vdpau), > did you install the necessary firmware? > > On Tue, May 1, 2018 at 5:53 PM, James <bjlockie at lockie.ca> wrote: >> I am playing videos from my lubuntu computer to my tv through the hdmi >> interface on my video card. >> It mostly works. >>

What GPU do you have? For video acceleration (i.e. va-api and vdpau), did you install the necessary firmware? On Tue, May 1, 2018 at 5:53 PM, James <bjlockie at lockie.ca> wrote: > I am playing videos from my lubuntu computer to my tv through the hdmi > interface on my video card. > It mostly works. > Some sound doesn't and I think it is related to the audio code. > A52

On Tue, May 1, 2018 at 6:16 PM, James <bjlockie at lockie.ca> wrote: > On 2018-05-01 06:00 PM, Ilia Mirkin wrote: >> On Tue, May 1, 2018 at 5:53 PM, James <bjlockie at lockie.ca> wrote: >>> I am playing videos from my lubuntu computer to my tv through the hdmi >>> interface on my video card. >>> It mostly works. >>> Some sound doesn't

I am using the script below to simulate a very low bandwidth connection. I found that I could turn the bandwidth knob down to about 4kbit, but below that I didn''t get any traffic through. I''ve had a look at this generally, but couldn''t find an answer. It doesn''t even seem like the first reply packet gets through. I have tried it with much bigger buffers,

Hi, Let's see if any of the questions gets answered or at least I get ponte dto something that can help me. I followed this wiki: http://wiki.samba.org/index.php/Samba4/HOWTO/Join_a_domain_as_a_DC#Getting_ready_for_joining_Samba_as_a_DC_to_an_existing_domain I have my S4 domain running, I compiled and installed another S4 to replicate the first server and joined successfully to the

Hi. I'm experiencing a issue (not big, but important) I have an asterisk installation with a buch of sip phones & analog ones. I have 2 1 sip phone that's outside in the "world", and is nat'ed. I'm using g.729 with it. I wanna use g.729 only for the remote phone, and ulaw for the local ones, since they're on a lan. What happens? when I call the remote phone, g.729

Did anyone have a way to make codec negotiation work with Asterisk? This is something I would love to have working as well. I won't need PSTN -> G729 mixing. Just SIP -> SIP using G729 for calling remote offices via VPN, but everything else use G711. -----Original Message----- From: Brancaleoni Matteo [mailto:mbrancaleoni@espia.it] Sent: Wednesday, July 16, 2003 11:32 AM To:

On an Asterisk 1.2.12.1 system using a TDM400P with two FXO and two FXS modules, servicing two POTS lines: When dialing a number, such as a bank, or pharmacy, where it is required to enter a long series of numbers via the phone's keypad, an unexpected hangup occurs. The hangup does *not* occur when entering the numbers, as one might initially expect, but after the called end begins to

Karl Heyes <karl@xiph.org> writes: > Are you running with realtime privileges, for this you need to start as > root if you want that. Even with realtime privileges there may be odd > cases where the scheduling latency is a bit too high, it all depends on > the drivers and kernel version but the current state is not that bad and > getting better. There are several ways to

I am unable to get re-invite to work on a new system. Also, unwanted transcoding is occurring on PSTN calls. The new system (FreePBX 2.11.0.37, Asterisk 11.9.0, CentOS 6.5) will eventually replace an old system (FreePBX 2.8.1, Asterisk 1.8.7.2, CentOS 5.8) currently in production. Both systems are on VPS with public IP addresses. Goals for the new system include: HD (g722) connections on