similar to: Always forward an extension?

I am trying to find the app ChanSpy for Asterisk v1.0.7. I have tried looking on VOIP-info.org's ChanSpy page (http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+ChanSpy)and also referred to the link regarding bug 3836 (http://bugs.digium.com/bug_view_page.php?bug_id=0003836). I downloaded the attachments and tried to use the patch and compile the source. However, it seems that

Any users of the VocTel VOIP service? (Canadian) How have you found the quality (Choppy / smooth audio)? Any problems registering? (I have been unable to register for hours) After reading about the collapse of a big USA VOIP provider, I'm curious Thanks, OCG -------------- next part -------------- An HTML attachment was scrubbed... URL:

IAX2/guest@switch-3.asterlink.com/996 at 1pm CDT on Thurday the 30th. If you have any topics that need to be covered please direct them to me. Thanks, /b --- Anakin: ?You?re either with me, or you?re my enemy.? Obi-Wan: ?Only a Sith could be an absolutist.?

I'm having a little problem. I have a dial-plan with a lot of SetVar's and loops, and under certain circumstances (reproducible) it makes asterisk crash. Wanting to debug this, I compiled using "make valgrind". But doing so, I eliminated the crashes and the dial-plan works perfectly. Now from what I understand, valgrind removes compiler optimisation to ease debugging. What kind

Lately when I issue a 'reload' from the CLI, I find that it will sometimes hang forever, completely locked up. I can press enter and see the CLI prompt move, but no commands are taken. "top" shows asterisk eating everything up: PID USER PRI NI SIZE RSS SHARE STAT %CPU %MEM TIME CPU COMMAND 20669 root 25 0 10068 9.8M 5392 R 88.4 1.9 1:02 0 asterisk

I have an incoming 800-number over IAX from Teliax and I'm experiencing the large packet loss on connection. When a call comes in there is no ring tone and the first few words of the welcome message are cut off, regardless of the delay I set. Standard call (not 800-number) coming over IAX with the same provider works just fine only the tall free number. So it seems there are some packet loss

>1- Call comes in without callerid >2- AGI script answers line >3- AGI script asks to record name >4- Park the call and get the parked extension number >5- Ring all the phones in the house (exec Dial) >6- If phone is picked up, play recorded name >7- Wait for DTMF to accept or decline call >8- If accepted, bridge parked call and current call. Mike, I am wanting this

Hello all! My basic problem is that we haven't been able to get realtime to use ODBC to store configuration data. Here are the details: We've installed Asterisk on a CentOS machine as follows: 1. Downloaded, compiled, and installed FreeTDS 0.63 2. Downloaded, compiled, and installed unixODBC 2.2.11 3. Downloaded, compiled, and installed Asterisk, Asterisk-Addons, and Zaptel from CVS

Okay, first a little background - I've been with Packet8 since a month after they started. I found that we were outgrowing their services and decided to move to an asterisk box in the office. I found a service provider that offered me a reasonable rate. After a fair ammount of testing I decided to stick with their services and port my 3 primary DID's from Packet8 to the new service.

I have a TDM02B to bring in two POTS lines for my incoming calls; I need to point each line to a different IVR... is there somewhere that can I can look to get this setup working? Basically, each line is for a different business. I know that for a DID the routing is simple but I'm not seeing where I can match up a DID with a Zap channel. I'm currently looking into the zapata.conf file

After doing an update from SUSE 9.2 to 9.3 and Checking out the latest from CVS, Asterisk crashes on startup with an apparent MySQL (res_config_register) error: # asterisk -vvvgc > asterisk_startup_error1.log asterisk: symbol lookup error: /usr/lib/asterisk/modules/res_config_mysql.so: un defined symbol: ast_cust_config_register The log is shown below. I've seen the posts

I ditched the idea of using Asterisk straight through my router, there was too much to set up in too little time for me. I've found a spare machine and installed Asterisk@Home on it. Things run smoothly except for connecting to my IAX provider; It doesn't even look like packets are going out at all. Here's my config: ANY Cable ------> Firewall/Router

Hi, I have a couple of questions about MeetMe and call queues. I'm still pretty new to Asterisk, but already having to write a Service Center call manager for it (which I might add, our director has agreed to make open source!). MeetMe: How can I get MeetMe (does it even do this) to ask the user to speak their name first, and play that as the new member announcement. It seems

Hi, FYI, Red Hat released an advisory today about a vulnerability in Ruby. So far it doesn't appear in the VuXML, but am I correct in presuming it will soon? https://rhn.redhat.com/errata/RHSA-2006-0604.html http://cve.mitre.org/cgi-bin/cvename.cgi?name=CVE-2006-3694 cheers, -- Joel Hatton -- Infrastructure Manager | Hotline: +61 7 3365 4417 AusCERT - Australia's national

Hi, On behalf of one of our users, I installed R 1.8.1 on HP-UX11.0 by compiling the source using the gnu compilers and all was fine with make check On 1.9.0, I get an error from 'make check' when running reg-tests-1.R which gives error exit code 1. Commands used are: ./configure --prefix=/apps/global/Gnu_R/R-1.9.0 --with-x R_BROWSER=mozilla (all on one line, of course) make make

Dear All, I'm trying to draw a TeX histogram with the following pair of commands, pictex(file = "realhisto.tex") hist(Peaklist$V3,xlab="Height $z/\\ut{mm}$",ylab="Probability density $\\phi{}(z-z_0)/(1/\\ut{mm})$") However, in the resulting file realhisto.tex, I get, for example \put {Height \$z/\ut\{mm\}\$} [lB] <0.00pt,0.00pt> at 136.13 9.17 when

Hi all, I have situation in that we are renting out a room in our office building to another company, including the provision of phone lines via our Aterisk box. To keep billing simple, we would like them to be billed separately for all their calls. We think we can achieve this by them leasing their own outbound IAX trunk (which our providers have said is no problem) and having all their

Hi, I'm having trouble getting an answer to the following problem on -questions - I hope someone here has done something similar and can help. I'd like to compile support for FreeBSD OPIE into sshd. Presently I have to use PAM to achieve one-time password support. On a 4.x system I have in /etc/ssh/sshd_config ChallengeResponseAuthentication yes and in /etc/pam.conf sshd auth

I think they are being vague to give people a time to upload to the latest version. Cheers, Dean > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Brian West > Sent: Thursday, 23 June 2005 11:45 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re:

Ok I have to get a vote of all the people that are going to come to Cluecon so we order the beer keg's for the developers board room. Anyone have any preference? (if you haven't registered for ClueCon now is the time to register!) Choices... choices... choices... I want Red Bull on tap! /b --- Anakin: ?You?re either with me, or you?re my enemy.? Obi-Wan: ?Only a Sith could be an