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I'd like to have an extension "join" a call. That is, C can join A and B, just as if it were an analog extension phone. ChanSpy works, sort of. The problem is that once A or B hangs up, the channel is gone. With an analog extension, C would remain connected with B if A hung up. Can I throw A and B into a confbridge and then add C? Create a new channel that grabs the A

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi OpenSSH'ers, I am emailing you to ask is it possible to record failed passwords attempts and log them to syslog? Are there patches available for this? Has anyone managed to do this before? Are there alternitive methods? Many Thanks, A - -- Alan Neville, Postgraduate Education Officer, DCU Students' Union 2009/2010, BS.c Computer

Hi, newer lvm2 releases don't have have uncommented "filter" lines, so the current way to edit lvm.conf doesn't work anymore. Instead, switch to augeas (with a "custom" len) for a cleaner and working way to set the lvm filter. Pino Toscano (3): daemon: add add_sprintf daemon: move AUGEAS_ERROR to the common header daemon: lvm-filter: use augeas for setting the

I'm trying to install and run Asterisk 11 on Ubuntu 12.04.2 running over Oracle VM VirtualBox (v 4.1.8). So far I have tried it following two guides. The first is the one from "Asterisk: The Definitive Guide" 4th edition (http://ofps.oreilly.com/titles/9781449332426/asterisk-Install.html) and the one from Billy Chia "How to Install Asterisk 11 on Ubuntu 12.04 LTS" (http://blogs.digium.com/2012/11/14/how-to-install-asterisk-11-on-ubuntu-12-4-lts/). I'm able to install Dahdi, Libpri and Asterisk with no errors bu...

...converting them to a suitable variable name for MySQL. > sqlSave(channel, zz, addPK = TRUE, verbose = TRUE) Query: CREATE TABLE zz (rownames varchar(255) PRIMARY KEY, NAME varchar(255), ID varchar(255), GROUP varchar(255), OAC_RT double, OUC_RT double, OAT_RT double, OUT_RT double, OFN double, OFP double, OFP_RT double, TAC_RT double, TUC_RT double, TAT_RT double, TUT_RT double, TAD_RT double, TUD_RT double, TFN double, TFP double, TFP_RT double, OFP_PERC double, OFN_PERC double, OTD_PERC double, TFP_PERC double, TFN_PERC double, TTD_PERC double, SEX varchar(255), AGE double, EDUC double, EM...

I made myself look a little silly recently in a talk regarding asteriskdocs.org. I didn?t realize the 4th ed. of the Definitive Guide was apparently actually out (http://shop.oreilly.com/product/0636920025894.do), because I went by asteriskdocs.org?s claim that it was being worked on in OFPS (now retired, apparently.) Is the 4th ed. available to read online like the 3rd ed. was? Is someone on this list able to update asteriskdocs.org with current info regardless? -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-...

I'm having annoying errors trying to get configure working. tar xvzf /usr/local/src/asterisk-1.8.6.0.tar.gz cd asterisk-1.8.6.0 ./configure I get complaints related to pwlib / ptlib... checking for openr2_chan_new in -lopenr2... no checking /root/pwlib/include/ptlib.h usability... no checking /root/pwlib/include/ptlib.h presence... no checking for /root/pwlib/include/ptlib.h... no checking

Hello I'd like to display CID information on users' monitor running Windows. I know I can run a script through the dialplan to send a datagram that is picked up Impulse Technology's free NetCID (www.imptec.com), but I'd rather use an open-source solution. An alternative would be to use a Windows application that would connect to Asterisk's AMI. I don't know if multiple

Hello! First of all a question: When you make the encoder tables (ie: mode_e.h) do you use the mapping0_forward function? Because you made the encoder tables, and after this you did a minor correction in the final beta4 ... (this correction was: additional[0]=fabs(additional[0]*scale); in the mapping0_forward function) (I see this was a bug, and the modification was correct, but this correction

Dear All, I want to play music in my AGI while i am searching for a field in DB. Actually during some processes in AGI i need to play music . Thanks in advanced. Regards, Mehdi

There does not seem to be very much info out there about using LDAP to create asterisk configurations. Does anyone have some information that they would suggest I start with? Thanks.

...ne without this certificate. So why should I use it and how to be sure it's used ? When trying to use it, I cannot see any client certificate request from the server in Wireshark. Should it be set somewhere in the conf ? I couldn't find anything in sip.conf or in Asterisk 1.8 doc<http://ofps.oreilly.com/titles/9780596517342/ch08.html#Voicemail_id272814)>about it. Thanks for your help, Gilles -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110129/4b20823f/attachment.htm>

Hi, I'm trying to work out an issue with call queues. I need the calls that are in a queue to be kicked out if all members are unavailable (for example if all SIP members are having network problems). I tried leavewhenempty = yes but that only seems works when all queue members specifically log out of a queue. I've looked at autopause, but we need it to automatically un-pause once it

Hi, I'm trying to connect two asterisk instances using the method described here.. http://ofps.oreilly.com/titles/9781449332426/asterisk-OutsideConn.html under the section "Connecting two Asterisk systems together with SIP" I have an user named venu in serverA and vijay in serverB the serverA ip is 192.168.0.35 & serverB is 192.168.0.36 Here are the details of the config...

After a bunch of cumbersome testing I was able to transcode a Quicktime MOV file to OGG with theora and Vorbis. The method's a bit fumbly, but it works... It is automatable except for the initial audio decoding. Software used: Xine, oggenc, ffmpeg, ffmpeg2theora, oggzmerge Step 1: Get the video out of Quicktime ffmpeg -i infile.mov -b 5000 -f mpeg2video outfile.mpg Step 2: Get the video

Hello. i am running debian 6 with asterisk 11.4. The system has exim4 to send to email the voicemails. i would like to get rid of the analog fax machine and use asterisk to send/receive faxes. I do have a PSTN line with a SPA3102 adapter to interface it to asterisk. The number of the PSTN line is dedicated to faxing only. So i would like to: -receive faxes to asterisk and then send it as PDFs

Hello, I have a current asterisk 1.8.13.0 asterisk-addons 1.6.24 asterisk-sounds 1.2.1 dahdi-linux-complete 2.6.1+2.6.1 libpri 1.4.12 and asterisk-gui 2.1.0.rc1 (not trying to use the gui, want to do everything by hand) with a TDM410 with 2FXO and 2FXS. I have my POTS (PTNS) line plugged into port 1 (FXO) and a analog phone connected to port 3 (FXS). I compiled asterisk with asterisk

How to get rid on following.. why its Invalid ? holler*CLI> queue show queue1 queue1 has 0 calls (max unlimited) in 'rrmemory' strategy (0s holdtime, 0s talktime), W:0, C:0, A:0, SL:0.0% within 0s Members: Agent/7201 (Invalid) has taken no calls yet Agent/7202 (Invalid) has taken no calls yet No Callers -------------- next part -------------- An HTML

Is this command the best way to access a MySQL database - MYSQL(Connect connid dhhost dbuser dbpass dbname) ? I thought I heard that using ODBC was a bit more stable. Anyone have any experience? Thanks, Matt

Hello, ffmpeg2theora (with encoder mmx patches) is available on konvalo.org. You can run it on Linux (2.6.x), FreeBSD (5.x) and NetBSD (2.x) on Intel x86. ffmpeg2theora may be considered the recommended Theora encoder. The author of the program is Jan Gerber @ v2v.cc Those of you who already use Konvalo.org do not need to do anything, the program is in your PATH. Otherwise see