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...ult = builder.CreateBinOp(llvm::Instruction::BinaryOps::Add, llvm::ConstantInt::getSigned(functionType, 40), llvm::ConstantInt::getSigned(functionType, 2)); builder.CreateRet(result); llvm::verifyModule(module, llvm::PrintMessageAction); std::string errorInfo; llvm::raw_fd_ostream fileStream("test.ll", errorInfo); llvm::PassManager pm; pm.add(llvm::createPrintModulePass(& fileStream)); pm.run(module); And here is the result: ; ModuleID = 'test' define i16 @foo() { __entry__: ret i16 42 } (Somehow I am beginning to get the fe...

...ruction::BinaryOps::Add, > llvm::ConstantInt::getSigned(functionType, 40), > llvm::ConstantInt::getSigned(functionType, 2)); > builder.CreateRet(result); > > llvm::verifyModule(module, llvm::PrintMessageAction); > > std::string errorInfo; > llvm::raw_fd_ostream fileStream("test.ll", errorInfo); > > llvm::PassManager pm; > pm.add(llvm::createPrintModulePass(& fileStream)); > pm.run(module); > > And here is the result: > > ; ModuleID = 'test' > > define i16 @foo() { > __entry__: > ret...

...g-originate-rec-dev2:1] AGI("SIP/upps-ccm-tq01-0000003f", "agi://4574/ajpbxtest.agi?path=uploadrec&callid=") in new stack -- <SIP/upps-ccm-tq01-0000003f>AGI Script agi://localhost/ajpbxtest.agi?path=uploadrec&callid= completed, returning 0 == MixMonitor close filestream (mixed) == End MixMonitor Recording SIP/upps-ccm-tq01-0000003f Unfortunately I get two different calls to the h extension, but this I can cope with. The one without called is not interesting. The uploading will fail since the MixMonitor is still on when I try to upload the file. The file will no...

I am trying to code a Web Service in rails that can handle file uploads. Now as i read, SOAP 1.1 doesn''t support this yet..so i will have to use SOAP4R, right? This is fine from server side...but will this API will be compatible to .NET clients, which will be using standard SOAP API perhaps. Any idea, whats the way to go here?

Hi I'm using asterisk 1.8.3.2 (with a couple of patches) I have the following scenario... SIP call comes in and gets answered by extension A (MixMonitor is executed as part of this inbound dial plan of the number being called) Extension A puts call on hold and calls extension B Extension A then does an attended transfer of incoming call to extension B I'm finding that the recording

...--------------------------------------------- 02 - Forget Me.flac Error Message: Could not find file 'F:\FLAC\ReFLAC\Claire Martin - He Never Mentioned Love (200 7)\02 - Forget Me.flac'. Stack trace: at System.IO.__Error.WinIOError(Int32 errorCode, String maybeFullPath) at System.IO.FileStream.Init(String path, FileMode mode, FileAccess access, I nt32 rights, Boolean useRights, FileShare share, Int32 bufferSize, FileOptions o ptions, SECURITY_ATTRIBUTES secAttrs, String msgPath, Boolean bFromProxy) at System.IO.FileStream..ctor(String path, FileMode mode, FileAccess access, FileShare...

...39;UnpauseMonitor()' with code '#01' -- Feature Found: pauseMonitor exten: pauseMonitor -- Executing [h at xxxxxxxxx:1] System("SIP/xxx.xxx.x.xxx-0000000c", "php agi-bin/process-call.php 0 "2013-07-12 15:09:30" xxxxxxxx inbound") == MixMonitor close filestream == End MixMonitor Recording SIP/213.166.5.185-0000000c I've tried executing it on self and peer with the same result. Any thoughts? Thanks in Advance Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: ish at pack-net.c...

...Linux. It is GPL) http://robur.slu.se/jensl/jellog.tar.gz Excerpt from headerfile: /* Functions for setting the destination of logger_facilities. - filename is the file to append log entries to. If file does not exist it is created with umask 0666 (modified as usual by open() call). - filestream is the FILE stream to use. - fd is the filedescriptor to use. - priority is the syslog priority to use as used in the syslog() call. - timestamp is used as a boolean. True means begin line with a timestamp. */ int logger_set_file(logger_facility facility, char *filename, int timestamp); int l...

I MAY be able to derive something out of the present example given here : http://blog.codeinmotion.com/index.php/2006/12/22/pdf-generation-in-ruby-on-rails/ but this deals with filling out forms. Is there a simpler example that just allows you to talk to itext , send it some plain text and get back a pdf and then send that pdf to the user as downloadable / renderable data? -- Posted via

...angup, cause 27 [Mar 18 17:04:15] VERBOSE[27353] pbx.c: == Spawn extension (incoming-pri, sw-30-218543080, 11) exited non-zero on 'DAHDI/189-1' [Mar 18 17:04:15] VERBOSE[27353] chan_dahdi.c: q931_hangup: other hangup [Mar 18 17:04:15] VERBOSE[27354] app_mixmonitor.c: == MixMonitor close filestream [Mar 18 17:04:15] VERBOSE[27353] chan_dahdi.c: NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null, hold-state Idle [Mar 18 17:04:15] VERBOSE[27353] chan_dahdi.c: NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null, hold-state Idle [Mar 18 17:04:15] VERBOSE[27353]...

...cuting [s at macro-agntanserd:2] MixMonitor("SIP/1119-0000001c", "testmixmonitorB.wav,W(4)") in new stack [May 5 17:26:40] == Begin MixMonitor Recording SIP/1119-0000001c [May 5 17:26:46] == End MixMonitor Recording SIP/1119-0000001c [May 5 17:26:46] == MixMonitor close filestream [May 5 17:26:46] == End MixMonitor Recording SIP/31-0000001b Any idead why is Asterisk not creating recording for Mixmonitor() application used in macro? Has anybody faced similar issue, or is a bug? Asterisk version- 1.8.3.2 I couldn't get chance to test on other Asterisk versions. Than...

...lement the last solution above, as it's the smallest change - if there's a preference for a different approach I'm happy to modify accordingly. The global rename is perhaps the better solution long-term, but it would require changes to the current naming scheme. (eg CallSite CS, FileStream FS) 2. The output from Solaris nm is just sufficiently different from GNU nm to break GenLibDeps.pl - modified the latter to be slightly more permissive. An alternative fix would be to ensure that a GNU nm is located by configure, but this seems to be the simpler fix for now. Outstan...

...tension (from-internal, 0149XXXXXX, 7) exited non-zero on 'SIP/306-000000b8' -- Executing [h at from-internal:1] Hangup("SIP/306-000000b8", "") in new stack == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/306-000000b8' == MixMonitor close filestream (mixed) == End MixMonitor Recording SIP/306-000000b8 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150325/074177f9/attachment.html>

...-- SIP/6005-00be6220 is ringing* -- SIP/6004-00c23270 answered DAHDI/77-1 -- Stopped music on hold on DAHDI/77-1 == Begin MixMonitor Recording DAHDI/77-1 == Spawn extension (queues, 6501, 12) exited non-zero on 'DAHDI/77-1' -- Hungup 'DAHDI/77-1' == MixMonitor close filestream == End MixMonitor Recording DAHDI/77-1 I want to input the ringing agent into my database.Anybody can help?I appreciate it a lot. Best Regards, Krishna __________ Information from ESET NOD32 Antivirus, version of virus signature database 4743 (20100104) __________ The message was checked by...

Hi, I've got a MSAccess database on a linux server (fedora core 5, samba 3, 192.168.0.90) which a Win XP client machine needs to use (192.168.0.50). However, Access refuses to open the database because "it's not on the LAN so is a security risk". How can I configure Samba so that Windows recognises 192.168.0.90 as a local machine, please? Jim Donovan Office +61+2-8923-5208

On Thursday, December 19, 2002, at 06:10 am, Monty wrote: > Het rillian: How to get HTML output from the XML in order to roll > changes back to the website? Or do you plan to replace the HTML > on-site too? Sorry, I got distracted before I finished everything. If you have the tools installed (xsltproc, passivetex) 'make' should generate monolithic pdf and html

...IP/mydevice/00441111111 WARNING[13053][C-00000002]: channel.c:6164 ast_channel_make_compatible_helper: No path to translate from SIP/mydevice-00000005 to SIP/hardphone-00000004 == Spawn extension (default, 00441111111, 4) exited non-zero on 'SIP/fixedline-00000004'. == MixMonitor close filestream (mixed) And when I try to try to initiate a call with a manager script, I receive an authentication error from the script. How might I find more info to help diagnose either or both of these issues? -- Eric Smith

...) exited non-zero on 'SIP/1010-00000162' == Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/1010-00000162' in macro 'dialout-trunk' == Spawn extension (from-internal, 7190000000, 6) exited non-zero on 'SIP/1010-00000162' == MixMonitor close filestream == End MixMonitor Recording SIP/1010-00000162 == Extension Changed 1010[ext-local] new state Idle for Notify User 1004 On inbound calls I see: == Using SIP RTP CoS mark 5 -- Called SIP/1010 -- Connected line update to SIP/ BVTrunk -00000160 prevented. == Extension Changed 1010[ext-l...

...XXXX, 7) exited non-zero on > 'SIP/306-000000b8' > -- Executing [h at from-internal:1] Hangup("SIP/306-000000b8", "") in new > stack > == Spawn extension (from-internal, h, 1) exited non-zero on > 'SIP/306-000000b8' > == MixMonitor close filestream (mixed) > == End MixMonitor Recording SIP/306-000000b8 > The verbose output states why your call is congested: -- Got SIP response 556 "No address found" back from 217.195.XX.XXX:5060 The far end came back with a 556 response to the outbound INVITE request. It doesn't t...

...rnal:1] Hangup("SIP/5547741200-000051cb", "") in new stack == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/5547741200-000051cb' == Spawn extension (from-internal, 6001, 42) exited non-zero on 'SIP/5547741200-000051cb' == MixMonitor close filestream (mixed) == End MixMonitor Recording Local/9991416445 at from-internal-0000017b;1 However, when I check the CDRs in the database, I see only the following: +---------------------+------------+------------+------------+---------------+-------------------------------------------+-----------------...