similar to: Timeout Gives T in cdr.

If '#' is pressed during a call the CDR that is written at the end of the call contains '#' in the dst / destination field rather than the number that was originally called. How do I avoid losing that original number so that I can use the CDR for billing? I've tried not having a '#' target in extensions.conf and I've tried calling ResetCDR(w) in the

Hi, Have a problem in my extension.conf: I have: [sip] exten => _333.,1,wait,3 exten => _333.,2,Answer exten => _333.,3,AbsoluteTimeout,7 exten => _333.,4,Hangup I wanted to test if * is executing this dial plan by calling 3335254255 for example. The problem is as follow: It waits, it answers but it does not seems to see the Absolutetimeout: call goes forever. What's wrong? Am

Is there a way to get the que logs to go to a postgresql database? And/or are they any documents that explain the layout and field meanings? The logs look like they will be very helpful. Especially to show how long calls are held and who is taking them. I would like to make simple web reporting tool to show a daily summary status of queue activity. Also, there is not a manager command to do

Hi, I had compiled support for MYSQL_FRIENDS and it works for SIP, but when use tiwh IAX2 I have some problem, I can register with a client, but when I try to make a call I got this error: Jul 17 12:52:03 NOTICE[229387]: chan_iax2.c:5183 socket_read: Rejected connect attempt from <IP-ADRRESS> When I google'ed this problem I can see other users also found this error (bug ?) But no-one

Hi, I just wanted to point out that I have added a patch to Bug #1420 (http://bugs.digium.com/bug_view_page.php?bug_id=0001420), which adds Manager Events and Queue Log records for the AgentCallbackLogin application. Since the Bug Summary starts with "[Request]" and the Severity is "Trivial", both of which I can't change, this patch might be easy to overlook. Would

Hello I have contacted my line provider who is saying that in order to get my 0845 or 0870 number to id as the incoming number on a landline that i may call i need the following. User must provide - NPI set to E.163/E.164 User must provide - TON = "national or international I have had a good search around and can't seem to find a good answer to this. Does anyone have any idea where i

hi all i have an asterisk working and i need to add a mettme public service. for example i need to download a soft (sjphone) and without any configuration call to 509@asterisk.mydomain.com (meetme) and join a conference but when i do that i received an error saying nomber do not exist. but if i call a extension is work propperly. in the extensions.conf have exten => 411,1,Answer exten

Let me lay it out for you.... Call comes in over a T1 - Signal is em_w. The extension is seen as *<callerid>*<last 4 digits of number being called>*. Which is fine in it self. I have my extension.conf file set up as follows... [did] ; Receive call as *<calling>*<called> exten => _.,1,Answer exten => _.,2,Cut(CALLING=EXTEN,*,2) exten =>

Hi, Sometimes my asterisk server stops. (after a day or two) Last output from CLI is: -------------------------------- -- Registered SIP '000b82017eb7' at 213.237.12.125 port 11620 expires 120 -- Channel 0/26, span 1 got hangup -- Hungup 'Zap/26-1' voip1*CLI> Disconnected from Asterisk server Executing last minute cleanups Asterisk cleanly ending (0).

Hi all I am having problems with atxfer if I do the extact same thing with blind xfer it works fine when I hit press #2 (defined in conf for atxfer) i get "transfer" I dial the number I want and i get the following on the console -- Playing 'pbx-transfer' (language 'en') -- Executing Dial("Local/18005558355@jesnjer-f97a,2", "/18005558355")

Hello- I am currently testing with a carrier that seems to be having some trouble around toll-free (800 number) access. While a problem, its the resulting behavior that I'm finding disconcerting. When I dial an 800#, I get the following response: -- Executing Macro("SIP/2700-e10b", "carrier-out|18005558355|70|r") in new stack -- Executing

Btw Dirk, can I help package interflex for apt? Please be welcome to refer me to guidelines/a manual for doing so if you can imagine it happening. Johan Den ons. 7. jun. 2023 kl. 08.37 skrev Johan Andresen < johan.andresen at gmail.com>: > I can understand why it changes the game. > > It would be sweet if we could do something like r2u for Debian, but for now >> we

Is there a particular way to run a specific alias analysis that is already implemented on a .ll file produced by clang -S -emit-llvm? On Mon, May 14, 2018 at 7:36 PM, Soham Sinha via llvm-dev < llvm-dev at lists.llvm.org> wrote: > Date: Mon, 14 May 2018 14:26:13 -0400 >> From: Kenneth Adam Miller via llvm-dev <llvm-dev at lists.llvm.org> >> To: llvm-dev <llvm-dev at

> > Date: Mon, 14 May 2018 14:26:13 -0400 > From: Kenneth Adam Miller via llvm-dev <llvm-dev at lists.llvm.org> > To: llvm-dev <llvm-dev at lists.llvm.org> > Subject: [llvm-dev] Andresen Algorithm > Message-ID: > <CAK7rcp_yXA79UuLvcD6LPMQBvi9tTpOGAcuWZ > Ux1tFKTyvuh2Q at mail.gmail.com> > Content-Type: text/plain; charset="utf-8" >

11-10-2004 there was a subject: Re: Where did USE_SIP_MYSQL_FRIENDS go?: on asterisk.user list. >All db specific code has been removed from the code in favor of the >currently-in-development "RealTime" method of configuration from >database. >You are most likely not using the 1.0 stable branch. >You need to use the new RealTime configuration method. And currently,

Usually you need to add the -analyze option, to dump the analysis results. On Wed, May 16, 2018 at 9:09 PM, Kenneth Adam Miller via llvm-dev < llvm-dev at lists.llvm.org> wrote: > Actually I figured out that it is: > > opt -cfl-anders-aa *.ll > > But now I noticed that I need the graph that it computes, and it doesn't > seem to emit that. > > On Wed, May 16,

Hi, A'm about to set up a asterisk for 5000 users, and the customer had a 64bit server - can asterisk compile on that ? I will use a digium X100P for timing use will that do on a 64bit ? (I'm using SUSE91 kernel 2.6) What else ? Is it posible to have only one server for 5000 users ? I gues that it will be 5-700 sim. users only talking sip, and IAX2 to my PSTN-Gateway. The system is

Got this error: Printing analysis 'Inclusion-Based CFL Alias Analysis': Pass::print not implemented for pass: 'Inclusion-Based CFL Alias Analysis'! On Thu, May 17, 2018 at 12:14 AM, Alexandre Isoard < alexandre.isoard at gmail.com> wrote: > Usually you need to add the -analyze option, to dump the analysis results. > > On Wed, May 16, 2018 at 9:09 PM, Kenneth Adam

Actually I figured out that it is: opt -cfl-anders-aa *.ll But now I noticed that I need the graph that it computes, and it doesn't seem to emit that. On Wed, May 16, 2018 at 5:19 PM, Kenneth Adam Miller < kennethadammiller at gmail.com> wrote: > Is there a particular way to run a specific alias analysis that is already > implemented on a .ll file produced by clang -S

I can understand why it changes the game. It would be sweet if we could do something like r2u for Debian, but for now > we can't. Switching between Debian and Ubuntu is not that onerous though. Do you mind clarifying what you do here? Do you operate Ubuntu when working in R? Or do you somehow operate Ubuntu to manage R packages and dependencies, and then switch back into Debian to carry