similar to: Music on Hold

Hi, I'm having trouble getting Dial to exit when the caller hangs up in Asterisk 1.4.21.2. I use a POTS line to call into the DiD given to me by VOIP service provider. When the call comes in, I have the VOIP provider send it to another POTS line. All this works fine however when the caller (me) hangs up, the Dial command does not exit. The callee stays connected (and my billing

Hi all, I need some help, advice or whatever you can explain to me because I haven''t got a clear idea about how to do the following assembly, and I''d be very grateful if I got some help from an expert like you. I''m trying to build a system which represents the following: I''ve got a hosts unit (host1, host 2, ...) which have IP in the network 192

Hello, I'm having trouble getting finding a company that provides DID in Western Canada. More specifically in Edmonton, Alberta. We have tried getting in contact with Link2Voip and Calgary Telecom but neither seems to be answering their phones or email. I would appreciate it if anyone can point me in the right direction. Thank you, Nelson

I have an "outbound-ld" context as follows: [ Context 'outbound-ld' created by 'pbx_config' ] '_1NXXNXXXXXX' => 1. Macro(enumdial|${EXTEN}) [pbx_config] 102. Wait(1) [pbx_config] 103. Set(LINE=${IF($[${LINE}=pots]?link2voip:${LINE})}) [pbx_config]

1. I get the following error when I run “puppet device’ err: Could not request certificate: Could not write /var/opt/lib/pe-puppet/devices/certname/ssl/private_keys/certname.pem to privatekeydir: Permission denied - /var/opt/lib/pe-puppet/devices/certname/ssl/private_keys/certname.pem Any thought? Thanks, -- You received this message because you are subscribed to the Google Groups

Hi All, I am using Asterisk for one of my projects in OpenBTS. I am having the age old problem of "extension not found" when try to make a call from one registered SIP phone to other registered SIP phone (two mobile phones connected to Asterisk via OpenBTS). The exact error thrown on Asterisk CLI is *"chan_sip.c:20039 handle_request_invite: Call from [IMSI310410270465840] to

Each register is a 4-component (namely, r, g, b, a) vector register. They are actually defined as llvm packed [4xfloat]. The instruction: add_sat r0.a, r1_bias.xxyy, r3_x2.zzzz Explaination: '.a' is a writemask. only the specified component will be update '.xxyy' and '.zzzz' are swizzle masks, specify the component permutation, simliar to the Intel SSE permutation

Hi there Looking at the archives I didn't find a solution to the following problem we have here: Printing from our linux-server (wagner) to an intel printserver (PS652D8F) doesn't work. Here's the stuff we know/tried: wagner:~ # smbclient -L //PS652D8F -N added interface ip=10.0.0.10 bcast=10.0.0.255 nmask=255.255.255.0 Got a positive name query response from 10.0.0.40 ( 10.0.0.40

Hi, I'm having trouble getting feature codes to work in Asterisk 1.4.21.2. Features.conf contians this: blindxfer=## atxfer=*2 automon=*1 disconnect=** I'm really most interested in getting disconnect to work so that I hear "Goodbye" when I press ** during a call connected this way in my dial plan: exten => 1,n,Dial(SIP/14168724765 at 6135551212-sw1|120|gtT) exten =>

Good evening all, hope everyone in the US had a nice Thanksgiving! On one of our internal servers, I decided to make the leap from 1.4.2x to 1.6.2.0-rc6 so I could start learning about the changes and new features that have been implemented. I upgraded all the configs, removed all the deprecated stuff, etc -- well went well. However, I noticed after the upgrade, when dialing into an

need help trying to figure out why calls hang when using multple ports on Sangoma card. we have 1 quad card with 3 T1 ports configured, Port1 acts as connection to teleco (to our T1 PRI) port 2 connects second system and routes calls to port1 port 3 is Asterisk pbx calls all go in and out properly but sometimes we get a call hang on when both sides hangup. this causes all calls to fail until

Actually the problems that Tzu-Chien Chiu are encountering are similar to what should be done for generating SSE code in the X86 backend and also other SIMD instruction sets. I think LLVM neeeds to add instructions for permuting components, extracting and injecting elements in packed types. If the architecture has instructions which can do permutations for each instruction (for example

Incoming calls ring SIP users who have |Ttr in their dial plan, but outgoing calls are done through a macro as follows: [macro-diallink2voip] exten => s,1,Dial(SIP/${ARG1}@link2voip-sw2,120) exten => s,2,Goto(s-${DIALSTATUS},1) exten => s-ANSWER,1,Hangup exten => s-CONGESTION,1,Dial(SIP/${ARG1}@link2voip-sw1,120) exten => s-CONGESTION,2,Goto(ss-${DIALSTATUS},1) exten =>

Hi all, i would like to find the best way to resolve the following problem. Suppoose i have a vector x of length N with k different elements. length(x)=N u<-unique(x) length(u)=k I would like to get a matrix M with k rows and N columns such that: in each line i (i=1,...,k), which(x%in%u[i]) is equal to 1 and 0 else. Thanks for your help. Olivier --

I have to register asterisk/sip with a sipproxy that does not support authentication, I do not know how to tell Asterisk not to send authentication request? # sip.conf [general] insecure=very permit=207.148.115.10/255.255.255.0 [myproxy] type=friend host=217.118.115.10 context=from-sip # Logging: <--- Reliably Transmitting (NAT) to 207.148.115.10:5060 ---> SIP/2.0 407 Proxy

Sorry to all bothering you with a trivial question: I am using R (rw0630 for Win32) and am simply unable to read-in the *.txt data. I've tried to copy the file (prum.txt) to different directories of rw0630 but get uniformly the answer "object "prum.txt" not found". Which is the default directory for read.table() ?? Thanks! Z. Skala ++++++++++++++++++++++ Zdenek Skala

Hi I am using R65.1 in Windows 95 I have a CSV file from Excell > a<-read.table("c:/heberto/mgc/tst.csv",header=T,sep=",") > attach(a) > a manolo fvcpp fevpp fvvcpp tlcpp rvpp rvtlpp plmaxpp 1 1 99.28 97.67 98.38 91.14 102.9 111.25 117.64 2 1 86.97 68.56 78.89 94.60 112.34 118.53 159.20 3 1 81.12 71.76 88.37 89.16

SetUp: - Asterisk behind a NAT, - Red Hat 9.0 - Asterisk 1.2.14 My Asterisk box is behind a NAT and I have a DiD from an ITSP. I have my dial plan set up so that when outside callers dial the DiD, the call is answered by my auto-attendant. The caller can then select who they'd like to speak to and the call is transferred to the external line associated with that person (usually a mobile

I have an Asterisk server it runs great with SIP phones, soft SIP phones (twinkle) and a soft SIP phone app on my Android phone but I am having problems getting two ATA boxes working. I have a Linksys PAP2T, it is unlocked and I have used them before with no problems. I was able to receive calls with from any local SIP phone or from my Link2VoIP connection via the Internet but it could not call

Hi, I have a quick question regarding estimation of a truncation regression model (truncated above at 1) using MLE in R. I will be most grateful to you if you can help me out. The model is linear and the relationship is "dhat = bhat0+Z*bhat+e", where dhat is the dependent variable >0 and upper truncated at 1; bhat0 is the intercept; Z is the independent variable and is a uniform