similar to: CVS Build from 16-7-2005 Crash! bug or what? ; -D

i get lots of the below from friday 15.7.05 cvs as well ERROR[1171] UTILS.C:509 TVFIX: WARNING NEGATIVE TIMESTAMP -194931. ... ____________________________________________________ Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs

Hi have set the following tbf tc qdisc add dev eth0 root tbf rate 0.5mbit \ burst 5kb latency 70ms peakrate 1mbit \ minburst 1540 I want to add a filter so the ip traffic pass from it.. plz help me __________________________________ Yahoo! Mail Stay connected, organized, and protected. Take the tour: http://tour.mail.yahoo.com/mailtour.html

Hi Friends, I have a problem with my Wine Installation in FC2-Athlon. I was following the PDF File - Configuring Wine Chapter 5 and it says that the "config" file is in ~/.wine/config. I can find the "config" file in the /etc/wine/config. Can you enlighten me on this? Actually, I can run already the "wine winmine.exe" and "wine notepad.exe" Also, I was

After buying some additional lines from my telco, I recently had my phone vendor wire the additional lines from my phone box into an amphenol connector that's plugged into my channel bank (Adit 600). However, although I make the following changes in my zaptel.conf and my zapata.conf files respectively and then reboot, the new channels, when selected for an outbound call, resume a dialtone

I have the following issue: 1) Call comes in from PSTN to Asterisk (IP A) and Asterisk forwards call to a SIP Proxy (IP B) 2) SIP Proxy (SER) forwards the call to a registered user. User does not answer and Call Forwarding is turned on for the user and the number to forward the call is a PSTN number. 3) After a specific timeout, SER has to forward the call to the "forwarded" PSTN

Hi: I need help setting-up multiple calling cards with different prices for the same routes using astcc. All my calling cards' routes now have the same price, but I need to be able to set multiple calling cards with different prices for the same route. I appreciate your feedback of How I can do that. Thanks; Chawki __________________________________ Yahoo! Mail Stay

Hi all, i am running a application which uses ODBC DSN. this is file and connecting with MS sql server. but when i run my program as $ wine c:\\crm\\PMonitor.exe i get fixme:vxd:VXD_Open Unknown/unsupported VxD L"secprov.vxd". Try setting Windows version to 'nt40' or 'win31'. But setting to this versions doesnot help as it provides dsn error like nddeapi.dll could not

--- Dan Pritts <danno@umich.edu> wrote: > Hi all - > > I've just finished building flac in the "darwinports" environment > on MacOS 10.3.9. > > The port maintainer (i've cc'd him) had disabled the xmms plugin > build. > > I wanted that, so I changed the portfile and built locally, yada > yada. > > I've run into three problems,

> All TCP connections have a port number, the default > for http is 80. Thanks. Good point. I was wondering how to handle this in icecast.xml. > As for the source of the relay, it can be shoutcast > or icecast, because > the relay is classed as a listener by the source. What if the source is neither icecast or shoutcast - but another mp3 stream source? If it matters, why?

Hiji wrote: >--- "David F. Colwell" <dfcolwell@dfcolwell.com> >wrote: > > >>Hiji et al, >> >>Couldn't find Half-Life or Counter-Strike in the DB >>yet they returned... >> >>Submitted version rejected >> >> >> >------------------------------------------------------- > > >>The version you

Hi - I've just been testing out the latest CVS HEAD (as of about 10:00a EDT today). I'm getting some weird errors. Calls from one sip phone to another have OK audio in one direction and highly scrambled audio in the other direction. The console shows this error repeated ad nauseum during each call: Jul 18 16:08:03 ERROR[22941]: utils.c:509 tvfix: warning negative timestamp

I've been getting these really weird errors that scroll almost nonstop across the screen when someone attempts to place a call and it starts ringing. Jul 19 17:55:01 ERROR[23246]: utils.c:509 tvfix: warning negative timestamp -536835.-70000 Does anyone know what this is all about and what I can do to fix it? Thanks for any help in advance, this is causing huge problems!

Hi, Updating zaptel gives me this during the make. Any ideas, the searches and Wiki gives me no hints. In file included from /usr/src/linux-2.4/include/linux/fs.h:19, from /usr/src/linux-2.4/include/linux/capability.h:17, from /usr/src/linux-2.4/include/linux/binfmts.h:5, from /usr/src/linux-2.4/include/linux/sched.h:9, from

I started the named server on CentOS and it seems to resolve DNS request OK, but it does not seem to retain the info for very long. From what I can tell using "dig", a domain's ip address is retained for less than 12 hours. So in the morning, it takes 4+ seconds to resolve again the first time. Is there an adjustment somewhere for this or is the caching named support not enabled by

Using pjsip 2.7.2 on Asterisk 15.5 Really struggling to make sense of translating these old 1.8 SIP instructions into a neat pjsip_wizard conf suitable for 2018 http://wiki.sip2sip.info/projects/sip2sip/wiki/SipDevicesAsterisk#Version-18 In pjsip_wizard.conf, I have the following, which seems to get me registered, and it responds to an incoming call, but I always get this: [Jul 28 18:32:29]

Dear all, I have enabled stun module and configured it in asterisk , but asterisk not using stun returned public ip address for any of the sip requests going out of my network. i have done settings as below res_stun_monitor.conf settings: [general] stunaddr = stun.ideasip.com stunrefresh = 30 stun show status Hostname Port Period Retries Status ExternAddr

I'm trying to configure sip2sip, which says: http://wiki.sip2sip.info/projects/sip2sip/wiki/SipDevicesAsterisk "Asterisk, is currently unable to handle more that one result for a DNS SRV lookup, and the Asterisk configuration needed for getting it work with the SIP2SIP service is not trivial" It then gives a complex multi-section workaround in SIP. I remember reading there'd be

Hi, I have to create scenario like following, I have 2 sip soft phone.I configured Asterisk server on local network, on Linux.With two soft-phone , local asterisk sever, i able to communicate.Now i have communicate with other network SIP client.For that i have opened account at @sip2sip.info, they provided me credentials.Then i registered one SIP phone to local Asterisk sever and another to

I've been googling and talking with Libretel about my setup and the fact that incoming calls to my asterisk box through the Libretel number reach my box (I hear the greeting being played) but then don't accept DTMF. Here is a rough diagram of my setup: Asterisk | server | NAT <------------ Libretel | router | Note that there are NO SIP

Unfortunately, Dennis, my source stream is 128kbps and will never go higher. Is Liquidsoap still a good idea? On Fri, 04 Mar 2016 14:48:41 +0100, you wrote: >Great tool to do this: liquidsoap > >Keep in mind that transcoding degrades the quality of tour stream dramaticly. You can avoid this by feeding liquidsoap or stream transcoder with a highquality or even transparant stream and