similar to: Odd asterisk behavoir

I have a user who is off my system with IAX. When he calls and goes out my long distance provider my asterisk switch seems to be bridging the two calls. As a result I loose all accounting information. All I get is the call setup time (15 or 20 seconds). How can I either make asterisk not bridge the call, or keep correct tabs on the call accounting for me? Mar 23 21:55:24 VERBOSE[18185]

Hi, I have 2 systems (A and B). I have an 800 number... when someone calls the 800 number it goes: IAX2-->A---IAX---B--->SIP PHONE However.. if the user calling the 800 number is a SIP user that is registered to A it goes: SIP--->A---IAX---B--->SIP PHONE This is the problem... when a call comes in from the IAX2 800 provider, things work fine... however if a SIP user registered to

Hi all, Have been experimenting with agents and queues instead of placing calls direct to a user's phone extension, but I've run into problems with calls to both the agent and the extension which creates a zombie and double records calls abandoned etc. We're using a unique queue for each agent (only a handful of users) to try and get some agent/queue information to see what the

Hi, Our Asterisk server has started core dumping more often, which seems to be related to when it releases the clone lock, and it stops masquerading the channel. So, I wondered if anyone had an idea what causes these masquerading attempts. Our system is running: Asterisk SVN-branch-1.2-r31555M. Full-log: Oct 20 11:09:37 DEBUG[57839] chan_sip.c: Assigning Replace-Call-ID Info

Originally posted on asterisk-dev with no response for 5 days, so posting it to the wider audience now. Asterisk Release 1.6.1.1 Scenario:- 1. 2 SIP peers (Zoiper softphone, if it matters) registered as 901 and 902 2. Using AMI, 901 is Originated 3. When 901 answers, it is Redirected to an extension "exten => dial,1,Dial(SIP/902)" 4. 902 rings, then answers 5.

On 1.2.x I have a random problem when a Zap/x channel flashes to transfer or make a three way call. The Zap/x-2 channel is created and the transfer or three way proceeds, but on hangup the Zap/x-1 channel fails to destroy the old bridge and asterisk goes crazy logging the problem. Here is an example debug log. This happens only once a day or so, with 100 or so users transfering and three

Hi, I'm currently running CVS-HEAD 2005-09-03 I do plan to upgrade to the newest version, but need to do some testing with it first. In the mean time... does anyone know what these messages below are about? I've never seen it before, but when it happened it locked Asterisk up pretty good. Feb 10 10:16:51 DEBUG[14917] chan_zap.c: Echo cancellation already on Feb 10 10:16:57

I tried to add a couple of SIP phones (polycom 601s) to my existing asterisk installation. I can successfully make a call from the SIP phone to any other phone (inside or outside), but I can not make any calls to a SIP phone. Attached are the pertinent parts of sip.conf and extensions.conf. The log starts off normal with: Mar 26 09:51:15 DEBUG[4885] chan_zap.c: DTMF digit: 2 on Zap/55-1 Mar

Hello to all, I post again (last time subject: Line drops strange problem(got event On hook) because i have caught in debug a situation where i get a call and the line drops and i get a call from the same caller and the line works well and the call normally closes by both parties. The only differences i find are underlined. If someone can understand the reason why the line drops from the debug

Hello to all, I have a strange problem with my asterisk. Line drops while i am in a call and without a reason.The line drops no matter if it is a incoming or outgoing call and it happen while i am talking on the phone (no silence detection problem). I tried to debug the situation and the only strange thing is the "got event On hook" (i guess..). I am thinking that it is a problem

Hello, I've problems with following - ----- --- --- PSTN | --- isdn --- | A | ----- iax2 ------ | B | ----- --- --- On [B], there is unconditional call forwarding set back via [A] (dialparties.agi is used) to PSTN. So, call from PSTN is routed via [A] to [B] and than back again into PSTN.

On Thu, Nov 29, 2012 at 11:03 AM, Dmitry N. Mikushin <maemarcus at gmail.com>wrote: > Hi Dan, Sean, > > Thanks for replies, > > So does it mean user is not expected to use any other stream for stdout, > but outs()? Although there is a comment "If you don't want this behavior, > don't use outs().", outs() is a static instance which always exists and

hi I have problem with mod_authz_ldap with this setup: <Directory /var/www/html/weby_2005> AuthzLDAPServer "ldap:389" AuthzLDAPUserBase dc=group,dc=cz AuthzLDAPBindDN uid=ds,ou=People,dc=group,dc=cz AuthzLDAPBindPassword XXXXX AuthzLDAPUserKey uid AuthzLDAPUserScope subtree AuthzLDAPLogLevel debug # needed for user auth

Hey everyone, I'm mirroring a lot of directories from one machine to another. The source machine is running Rsync 2.5.7 and the destination machine (the machine holding the mirror) is running Rsync 2.6.9. I'm using the --backup, --backup-dir, and --delete-after options. I tested this with two machines running Rsync 2.6.9, and noticed that files deleted in the mirror (because

Hi - The problem is with two abline(s). Attached are: 1) jnk.r to run program 2) jnk.rin the data 3) jnkps.eps the postscript output 4) jnkscreen.bmp (from photoshop after bmp copy to clipboard) in jnk.zip the eps and bmp are different on my machine (windows 2000) any suggestions appreciated bob >>> Diego.Kuonen at epfl.ch 03/18/00 08:24AM >>> "Robert L.

hi listees- Running windows 2000 and r 0.99 or 1.00; I get different lines on screen and print out: > j<-read.table("jnk.rin",head=T,sep="\t") > attach(j) > #gives lines I want > qqnorm(Dog,pch="o",main="Cumulative Frequency Log Dog Size",log="y",sub="Figure 2",x =c(-3,3),ylim=c(2000,1000000)) >

Hi Dan, Sean, Thanks for replies, So does it mean user is not expected to use any other stream for stdout, but outs()? Although there is a comment "If you don't want this behavior, don't use outs().", outs() is a static instance which always exists and creates problems, even if not used. - D. 2012/11/29 Dan Gohman <dan433584 at gmail.com> > On Wed, Nov 28, 2012 at

On Wed, Nov 28, 2012 at 10:54 PM, Dmitry N. Mikushin <maemarcus at gmail.com>wrote: > Dear all, > > Consider there is a program that writes to stdout using two different > raw_fd_ostream-s: raw_fd_ostream does buffered I/O, so it's not really meant to be used like this. > #include "llvm/LLVMContext.h" > #include "llvm/Module.h" > #include

Hello, I came across a strange behavior of expand.grid (or at least strange to me). For certain values of one of my input variables - created by seq() - I have to use strings (e.g ==".6") to select a row of the object created by expand.grid(), for other values numerical (e.g. ==.8) and for some both work. Please find an example below. #Example x<- seq(0,1,1/10) y <-

Hello, Below is some example code that should reproduce an error I'm encountering while trying to create a tiff plot with two subplots. If I run just the following bit of code through the R GUI the result is what I'd like to have appear in the saved tiff image: x<-seq(0:20) y<-c(1,1,2,2,3,4,5,4,3,6,7,1,1,2,2,3,4,5,4,3,6) plot(x,y,type="l",las=1,ylim=c(0,12))