search for: regal

Hi I was interested to see how Mongrel uses Lex/Yacc to parse the HTTP requests using a Regal generated parser. I downloaded the source but do not see the lex and yacc files...

Hi, please forgive me for this (hopefully) simple question. I cannot seem to find an answer or solution while searching around. I want to be able to call in to my server using my cell phone and be able to set call forwarding for my extension and enter a phone number and also be able to call in to that extension and disable the call forwarding. I see I can do this through the ARI web interface

Hi, As of v 1.6.1.1, can anyone tell me what the current possible values for DIALSTATUS could be? I found http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS but believe it is outdated since there is no FAIL or FAILED in this list. Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I am attempting to make about ten calls simultaneously and intermittently get 'SIP/voipprovider is circuit-busy' followed by 'everyone is busy/congested at this time" I am not sure if this is related to my bandwidth to my voip provider, a configuration issue or something else. Has anyone seen this before and have any suggestions. Thanks in advance. --------------

...d is, "what code would you never write?" Standard library functionality is at the top of my list. Why would anyone replace rigorously tested code with something not as widely tested that one has to maintain oneself? It reminds me of an interview lunch at which the developers constantly regaled me with tales of their wonderful custom data structures library. And then I asked about where they spend most of their time debugging. Guess where? Suffice to say, I didn't take that job. :) Again, I'm not dismissing libc++. I'm making a general point about reusing hardened code....

Hi All, I am relatively new to Asterisk. I have CDR enabled and successfully writing to MS SQL server. In my cdr table I am setting the userfield value with a line in my dialplan. If a call is placed to an invalid number (e.g. 12125551212), I see a cdr record created, however, my userfield value never gets set since the call never made it into the context of my dialplan. I am using AMI with the

Hi, I am attempting to implement Answering Machine Detect and have also played with using BackgroundDetect instead. Does anyone recommend one over the other? Here is the code I am using for the BackgroundDetect method (from voip-info.org). Thanks. [detect] exten => s,1,Set(MACHINE=0) exten => s,2,Answer exten => s,3,BackgroundDetect(silence/5, 1000, 50) exten =>

Hi, I am using a call file formated like this: Channel: local/12125557891 at outbound/n Callerid: 12125551212 Context: detect Extension: s Priority: 1 This sends the call into the dialplan at the [outbound] context. In [outbound], I have: [outbound] exten => _1.,1,Dial(SIP/${EXTEN}@flowroute,43) If the call is answered, it move on to the [detect] context. When using this method, it appears

Hi, My Dial() is set to the following, but always stops about 30 seconds into the call even when I set it to try for 60 seconds. exten => dialnumber,1,Dial(${DialInfo},60) I am running on 1.6.1-r199820. Is there some other setting that is overriding mine? Or an issue with this release? Thanks for the help. JR -------------- next part -------------- An HTML attachment was

Hi All, My client makes manual sales calls to prospects. He is often sent to voicemail on the prospect's side. If he finds himself having to leave a message, he would like to be able to press a key and let a pre-recorded message play into the prospect's vmail box. This is so he can maintain consistency in his message. Can anyone offer suggestions of how I could accomplish this

Hi, I just deployed new Aastra 9480i phones and when I attempt enter digits on other systems, like host pin in a GoToMeeting, the servers on the other end do not get my entries. I am assuming this is a DTMF issue but do not see anything in this phones config other than turning on the display of the digits. I have the DTMF method set to "SIP INFO". I am using AsteriskNow w/FreePBX.

Hi all, I am trying to make decisions in my dialplan based on {DIALSTATUS}. I am creating calls using AMI (rawman with parameters via URL) with action:Originate. I am using SIP and an outside voip provider for the calls. If I define the number to call in the Channel parameter (e.g. SIP/15555555555 at myvoipprovider, the call gets placed before entering the context that I defined. I understand

On Tue, May 11, 2010 at 5:40 PM, Andrew Sutton <andrew.n.sutton at gmail.com>wrote: > > What's driving libc++? >> >> The possibility of being a superior solution. >> > > I thought "to support Apple applications" from the previous post was > sufficient motivation :) Either way, I'm excited about a new library. Plus, > it looks a little

Hi Nicolas, I'm running it on Kubuntu. Now that you mention it I did have a problem during vmkit setup. After running the ./configure file with all the options I ran the make file it was giving this error: make[3]: *** [/.automount/root/exports/transient/nataraj/Project/vmkit-0.25/lib/Mvm/Runtime/Release/EscapeAnalysis.o] Error 1 " So I went into Makefile.common and modified GNU

Hi All, LLVM now includes a C++ standard library, written by Howard Hinnant. You can read about it here: http://blog.llvm.org/2010/05/new-libc-c-standard-library.html Or get the code here: svn co http://llvm.org/svn/llvm-project/libcxx/trunk libcxx If you have questions or comments, please direct them to one of the clang mailing lists. Thanks! -Chris

We should prefer `const struct pci_device_id` over `DEFINE_PCI_DEVICE_TABLE` to meet kernel coding style guidelines. This issue was reported by checkpatch. A simplified version of the semantic patch that makes this change is as follows (http://coccinelle.lip6.fr/): // <smpl> @@ identifier i; declarer name DEFINE_PCI_DEVICE_TABLE; initializer z; @@ - DEFINE_PCI_DEVICE_TABLE(i) + const

We should prefer `const struct pci_device_id` over `DEFINE_PCI_DEVICE_TABLE` to meet kernel coding style guidelines. This issue was reported by checkpatch. A simplified version of the semantic patch that makes this change is as follows (http://coccinelle.lip6.fr/): // <smpl> @@ identifier i; declarer name DEFINE_PCI_DEVICE_TABLE; initializer z; @@ - DEFINE_PCI_DEVICE_TABLE(i) + const

...sers] ...is circuit busy message To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Message-ID: <alpine.BSF.2.00.0905202112310.46773 at phoenix.jeff.net> Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed On Wed, 20 May 2009, John Regal wrote: > Thanks for the reply and apologize for the double post. My original post > landed in another thread and thought it may have been missed... > > I questioned my voip provider before posting and they told me they have > other asterisk customers that are making hundreds of simu...

Hi All, I know this was a bug a while back. I am using Asterisk 1.6.0.8 and logging cdr's to MySQL. All records show FAILED for the disposition regardless of what really happens. Please let me know if there is something more I could post to help figure this out. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, I have the latest AsteriskNow installed (1.5) and after a couple of months with system in production I have a need to use the Paging/Intercom features. I have the module installed and I am able to successfully intercom with individual phones using *80xxx (extension number) but if I create a paging group it does not work. I receive a message that it is an invalid conference number. I