similar to: Change OS from CentOS 6 to 7

Hi everyone, I have a very annoying character encoding problem. Have a look to this: # ls -l M*mo-1.* -rw-rw-rw- 1 root root 8417218 6 sept. 2013 Mémo-1.aif -rwxr--r-- 1 hope hope 8417218 6 sept. 2013 Mémo-1.aif -rw-rw-rw- 1 root root 363175 6 sept. 2013 Mémo-1.m4a -rwxr--r-- 1 hope hope 363175 6 sept. 2013 Mémo-1.m4a Yes, it looks like two files have exactly the same name, but

I had to setup a new Samba server as a PDC in my job (Samba 3.0) to replace and existing Samba server PDC (Samba 2.2), the old server was running ldap as backend, the new is running tdbsam. I have a lot of bizarre problems when I put both in the same workgroup with other machines. I will give you a list of them: 1) The new server is configured as local master browser and domain master browser,

hi, i have strange problem with no rtp packets from asterisk after dns query. see pcap below centos6/asterisk 13.9 + chan_sip 172.23.0.3 - asterisk 172.23.5.1/2 - voip phones any ideas/hints? 1170 25.028206000 172.23.0.3 -> 172.23.5.1 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x334508F6, Seq=49318, Time=1442112256 1171 25.045556000 172.23.5.1 -> 172.23.0.3 RTP 214 PT=ITU-T G.711

On Thu, Apr 26, 2018 at 07:29:41PM +0200, Emmanuel Florac via samba wrote: > Hi everyone, > > I have a very annoying character encoding problem. Have a look to this: > > # ls -l M*mo-1.* > -rw-rw-rw- 1 root root 8417218 6 sept. 2013 Mémo-1.aif > -rwxr--r-- 1 hope hope 8417218 6 sept. 2013 Mémo-1.aif > -rw-rw-rw- 1 root root 363175 6 sept. 2013 Mémo-1.m4a >

Hello list, We're currently facing some issues concerning T.38 gateway faxing. This is a device used almost exclusively for receiving faxes. Calls are incoming to asterisk on a SIP trunk (sangoma netborder) using G711A. Gateway mode is activated in the asterisk dialplan towards a Cisco SPA 112 running firmware 1.3.5. We are using asterisk 1.8.13.0 with the T.38 gateway patch applied (I know I

Hi LLVMers, there were a gcc ICE problem discussed in current mail list. Chris was right here: http://mail.cs.uiuc.edu/pipermail/llvmdev/2003-December/000693.html saying that the PR 12544 is not really the corresponding issue :) The correct one is PR 13392: http://gcc.gnu.org/bugzilla/show_bug.cgi?id=13392 Interesting fact is that -O2 (or -O3) goes somehow around this

Hello. Is someone here using VoIPmonitor? I am using just the sniffer and I found some pcap files that contain some odd streams. For example, I have a file with 3 streams, but the weird stuff is that 2 streams are the same (e.g., have the same source address and port and same destination address and port). Example: "Source Address","Source Port","Destination

If this has been covered before - I appologize. We use some Sipura SPA-2000's with the g711 codec and all seems fine (except for the occasional failure to register errors in my asterisk logs - but I will save that for another post). g711 call quality is on par with our Cisco 7960's. However, when using the g729 codec, the call quality on the Sipura device goes downhill on the PSTN side

hey all, can any body know what this parameter stands for i got RTPAUDIOQOS while i have SIP channels but could not understand then what this parameter tell * ssrc=254186206;themssrc=2024901615;lp=0;rxjitter=0.020917;rxcount=150;txjitter=0.000000;txcount=83;rlp=0;rtt=14818.715000 * if any one know plese help me to or give any documentation link regards Dhaval -------------- next part

Hi Everyone, We're routing calls through Asterisk (dialing in via sip and then dialing out via SIP). We've noticed a curious behavior in chan_sip that doesn't persist with chan_pjsip. When examining the packet capture, we're seeing the SSRC changing constantly on the call. At first it happens over a variable interval (15s 6s etc) but eventually it ends up changing exactly every

Hi, I'm getting messages like res_srtp.c:395 ast_srtp_unprotect: SRTP unprotect failed with replay check failed (index too old), retrying == SRTP unprotect failed on SSRC 576693764 because of authentication failure 10 == SRTP unprotect failed on SSRC 576693764 because of authentication failure 160 [...] ... after a couple minutes during voice calls after which the connection is being

Hello I am trying to set up webRTC video calls from my Chrome webbrowser (Fedora) to my Chrome webbrowser (Windows 10). There is local video input (I can see myself), but never video on the receiving side. This is the case in both directions (so it makes no difference which peer is calling which peer). Both webRTC SIP peers have opus and H264 codec in their peer definition :   Video

Building PowerPC.td register information header with tblgen Included from PowerPC.td:22: Parsing PowerPCInstrInfo.td:53: Variable not defined: 'GPRC'! make[3]: *** [PowerPCGenRegisterInfo.h.inc] Error 1 make[3]: Leaving directory `/pool/tmp/ssrc/llvm/lib/Target/PowerPC' maybe I just have to "make clean" and/or ./configure BTW, would it be nice to put Depend, Release and

Reclocking of NVAA/NVAC is substantially different from NV50+, enough to justify a separate clock implementation. This code is a forward-port of reclocking code that has been sitting in a branch for a while, and has been tested on my NVAC. Traces show no significant reasons why this shouldn't work on NVAA, but testers are always welcome. And since these are IGPs without dedicated RAM to

Hi list, I know * generates it's outgoing RTP stream based on the incomming RTP stream, i'm having some audio problems after i recieve an rtp reinvite from my carrier. Situation: Phone -- Asterisk A --- Asterisk B --- Carrier --- PSTN Asterisk A: reinvite = no Asterisk B: reinvite = no If i dial out on phone via asterisk A, Asterisk B relay's the INVITE to the carrier, after the

Evnin... Googled around for this strange error meesage with no helpful results at all... Does somebody has any idea what this means? "Forcing Marker bit, because SSRC has changed" At the same time I only get inbound audio but other side can't hear me...sometimes I just hear my echo and nothing from other side... Asterisk version 1.2.9 and both participants with public IP

Corresponds with GT215. Don't rely on the lock test logic being unconditionally enabled, and disable test logic when done (presumably to save power). v2: Remove warning, nvkm_msec already warns on time-out Signed-off-by: Roy Spliet <nouveau at spliet.org> --- drivers/gpu/drm/nouveau/nvkm/subdev/clk/gf100.c | 8 +++++++- drivers/gpu/drm/nouveau/nvkm/subdev/clk/gk104.c | 8 +++++++- 2

Hi, Is there a variable that contains the Sip Peer name? I was using ${CALLERID(num)} for outgoing calls, but when a call is being transferred, that variable contains something else. I need a variable that is always set to the SIP Peer's name. Thanks Dan -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Has anybody some information on on the "Redundant audio data" header in the speex rtp payload? Is this header always present? Is its value always the same? Can it be modified through some speex_*_ctl function? Thanks, Emmanuel -- ------------------------------------------------------------- Emmanuel Wauters Tel : (+32) 11 30 13 30

Hello, any idea why the Asterisk CLI gets flooded by these messages ? How can the SIP peer /vita3/ cause this flood ? [Apr 2 11:45:48] VERBOSE[17029] app_dial.c: [Apr 2 11:45:48] -- SIP/vita3-000010af requested media update control 26, passing it to SIP/708708-000010b3 [Apr 2 11:45:48] VERBOSE[17029] app_dial.c: [Apr 2 11:45:48] -- SIP/vita3-000010af requested media update control 26,