search for: 5064

...logics[i] = 1; } #pragma omp parallel for schedule(dynamic,1) reduction(&:bit_and) for (i = 0; i < LOOPCOUNT; ++i) { bit_and = (bit_and & logics[i]); } } //----------------------------------------------------------- Examination with valgrind produces these results: ==5064== Thread 2: ==5064== Use of uninitialised value of size 8 ==5064== at 0x40074E: main.omp_fn.0 (in test113) ==5064== by 0x4C44FE1: gomp_thread_start (team.c:115) ==5064== by 0x3A1EA07760: start_thread (in /lib64/libpthread-2.12.1.so) ==5064== by 0x3A1E2E14FC: clone (in /lib64/libc-2.12.1...

...bled on the Grandstream IP-telephone, then the local Asterisk-server CLI shows this (when SIP debugging) : --- [Oct 10 18:09:21] Really destroying SIP dialog '0d8ad63d6c1940a11ff1ee23728468ba at 192.168.1.77' Method: OPTIONS [Oct 10 18:09:31] Reliably Transmitting (no NAT) to 192.168.1.100:5064: OPTIONS sip:nslu at 192.168.1.100:5064;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.77:5060;branch=z9hG4bK769c6ff5;rport From: "asterisk" <sip:asterisk at 192.168.1.77>;tag=as1cad824f To: <sip:nslu at 192.168.1.100:5064;transport=udp> Contact: <sip:asterisk at 192.168....

https://bugzilla.samba.org/show_bug.cgi?id=5064 Summary: compile error - SCO 5.0.6a Product: rsync Version: 3.0.0 Platform: x86 OS/Version: Other Status: NEW Severity: normal Priority: P3 Component: core AssignedTo: wayned@samba.org Repor...

...piranha on same vip on 9004 port like below. 192.168.23.3:9004 ---->>192.168.17.16:9004 192.168.23.3:9004 ---->>192.168.17.17:9004 But i am getting below error on my syslog ################################################################################ Jul 21 20:25:38 testlvs nanny[5064]: starting LVS client monitor for 192.168.23.3:9004 Jul 21 20:25:38 testlvs nanny[5065]: starting LVS client monitor for 192.168.23.3:9004 Jul 21 20:25:43 testlvs pulse[5040]: gratuitous lvs arps finished Jul 21 20:25:47 testlvs nanny[5065]: READ from 192.168.17.16:9004 was too short Jul 21 20:25:4...

...') WARNING: app_voicemail.c:4922 vm_authentication: Unable to read password. I read in previous posts that this may be to do with the dtmf settings and have set both (asterisk and BT100) to info. This has not helped. My phones register with SER (port 5060) and use Asterisk for voicemail (port 5064). My configs are below along with my BT100 settings: ;Grandstream BT100 SIP Server: x.x.x.x:5060 SIP User ID: 2092 Authenticate ID: 2092 Name 2092 SER then forwards to port 5064 of Asterisk. ;sip.conf [general] bin...

Under asterisk info: Sip registry 12/12 76.xxx.xxx.xxx D N 5066 UNREACHABLE 11/11 76.xxx.xxx.xxx D N 5064 UNREACHABLE 10/10 76.xxx.xxx.xxx D N 5062 UNREACHABLE All these IP phones are behind NAT. What could be the problem? Thanks in advance. Jarga -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asteris...

...-------------- ---- 2nd port ------------------------------------------------------------------------ ---- --snip-- <-- SIP read from 194.208.44.44:34560: REGISTER sip:194.208.44.43 SIP/2.0 Via: SIP/2.0/UDP 194.208.44.44:34560;branch=PPC998202.z9hG4bK97220000265fffff,SIP/2.0/UDP 194.183.145.211:5064;branch=z9hG4bK97220000265fffff From: <sip:435572949012@194.208.44.43;user=phone>;tag=f8e70000a6870000 To: <sip:435572949012@194.208.44.43;user=phone> Contact: <sip:PPC998202@194.208.44.44:34560;user=phone> Call-ID: e62dffffcd4dffff@194.183.145.211 CSeq: 100 REGISTER Expires: 3600...

...') WARNING: app_voicemail.c:4922 vm_authentication: Unable to read password. I read in previous posts that this may be to do with the dtmf settings and have set both (asterisk and BT100) to info. This has not helped. My phones register with SER (port 5060) and use Asterisk for voicemail (port 5064). My configs are below along with my BT100 settings: ;Grandstream BT100 SIP Server: x.x.x.x:5060 SIP User ID: 2092 Authenticate ID: 2092 Name 2092 SER then forwards to port 5064 of Asterisk. ;sip.conf [general] bin...

...ant to use 5062 port for SIP protocol. I made the below modifications in my server to use 5062 port. Polycom phone: port=5062 Trunk settings: port=5062 sip.conf: bindaddr=5062 Extension configuration details: 5062 Our VoIP provider told me that they are allowing the SIP traffic through 5060 to 5064. I observed on my server console that my server is registered with our VoIP provider with 5062 port. But, I am unable to make outgoing calls. Do I need to modify any other settings in Asterisk? Look forward to your response. Thank you. Regards, Chandra. --------------------------------...

...') WARNING: app_voicemail.c:4922 vm_authentication: Unable to read password. I read in previous posts that this may be to do with the dtmf settings and have set both (asterisk and BT100) to info. This has not helped. My phones register with SER (port 5060) and use Asterisk for voicemail (port 5064). My configs are below along with my BT100 settings: ;Grandstream BT100 SIP Server: x.x.x.x:5060 SIP User ID: 2092 Authenticate ID: 2092 Name 2092 SER then forwards to port 5064 of Asterisk. ;sip.conf [general] bin...

Hello Aaron, Anton, LLVM-dev, While working on http://llvm.org/PR13676#c6 I found out that whenever I compile code with class methods returning structures it get generated incompatible with MSVC. Looking at lib/Target/X86/X86CallingConv.td, I found out that CC_X86_32_ThisCall maps SRet to EAX but in fact it should push the address of the return temp on stack. The following patch fixes the issue

...As far as I remember, there is a difference - "simple" (probable POD-like stuff) are returned in the regs, while classes with non-trivial ctors, etc. are passed / returned on stack. It's frontend responsibility to emit proper IR in this case. See http://llvm.org/bugs/show_bug.cgi?id=5064 and around. This seems to be the correct description of what's going on. -- With best regards, Anton Korobeynikov Faculty of Mathematics and Mechanics, Saint Petersburg State University

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Guenther Boelter <gboelter at gmail.com> schrieb: > Don't use Port 5061, your SIP-port should be always even like 5060, > 5062, 5064 or 5066. Could you please explain why? I see in /etc/services, that 5060 is the port for SIP and 5061 for SIP-TLS, but I don't find anything for the other ports... Thanks Luca Bertoncello (lucabert at lucabert.de) -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.10 (GNU/Linux) iEYEARECAAYFA...

...0.0003 4040.0004 All lines is registered in 5060 port so when someone call to 4040.0001 the call arrive on asterisk but arrive to last number registered 4040.0004becouse it is listening on same port as all others. How i make each number register in one different port, like 5060,5061,5062,5063 and 5064 ?? Thanks. -- Frederico Madeira fmadeira@gmail.com www.madeira.eng.br -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061109/8354d0dd/attachment-0001.htm

Hi, I found some suspicious code in X86TargetLowering::getRegClassForInlineAsmConstraint, but I don't know if it's a bug or my poor understanding of what the code does. This is the code in question: (lib/Target/X86/X86ISelLowering.cpp:5064) if (VT == MVT::i32) return make_vector<unsigned>(X86::EAX, X86::EDX, X86::ECX, X86::EBX, 0); else if (VT == MVT::i16) return make_vector<unsigned>(X86::AX, X86::DX, X86::CX, X86::BX, 0); else if (VT == MVT::i8) return make_vector<unsigned>(X86::AL, X86::DL, X86::CL,...

...fter a router with NAT: * SPA-1 and SPA-2 are on the same network, we have a pat 5060 => SPA-1 and 5061=> SPA-2 on the internet router * SPA-3, we have a pat 5062 => SPA-3 * SPA-4, we have a pat 5063 => SPA-4 * SPA-5, we have a pat 5064 => SPA-5 * SPA-6, we have a pat 5065 => SPA-6 On the Asterisk Sip conf, we have nat=yes and dynamic host. The problems are SPA-1 and SPA-2 can call to all other SPA except SPA-3 with SPA-3, i speak, it's good, spa-3 have the sound, but spa-3 speak i don't have the sou...

...gistered indication country 'fr' -- Registered indication country 'de' -- Registered indication country 'nl' -- Registered indication country 'uk' -- Setting default indication country to 'us' ........WARNING[16384]: File chan_iax2.c, Line 5064 (set_config): Ignoring port for now ...WARNING[16384]: File chan_zap.c, Line 577 (zt_open): Unable to specify channel 21: Device or resource busy ERROR[16384]: File chan_zap.c, Line 4781 (mkintf): Unable to open channel 21: Device or resource busy here = 0, tmp->channel = 0, channel = 21 E...

...st, using asterisk 1.4.30 I have the strangest problem that some SIP accounts can register to my Asterisk and others not. I see no connection between all those that can register or all those that can't. It's not a firewall problem as all register to port 5060 and the range 5060 --> 5064 is open. It's just very strange that some can register and other not. Any input/suggestions ?! Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100624/17f2c46e/attachment.htm

...t; POD-like stuff) are returned in the regs, while classes with > non-trivial ctors, etc. are passed / returned on stack. Sort of. > It's frontend responsibility to emit proper IR in this case. Isn't it what's SRet is supposed to be? > See http://llvm.org/bugs/show_bug.cgi?id=5064 and around. This seems > to be the correct description of what's going on. FTR, http://llvm.org/bugs/show_bug.cgi?id=5058 seems to be more up-to-date. Thanks for your reply!

...nothing about this on the mailing list, I apologize if I missed any > important information. I would suggest avoiding any complicated calls from LLVM-generated code into Visual Studio-generated code. For example, http://llvm.org/bugs/show_bug.cgi?id=5058 , http://llvm.org/bugs/show_bug.cgi?id=5064 . Not sure about the particular issue you're running into; looks like a bug at first glance. The file that mostly controls this sort of thing is lib/Target/X86/X86CallingConv.td . -Eli