similar to: Setting 'fname_base' variable doesn't affect 'automon' result file.

Hello Guys, After the Wiki was updated to the 3.5.X version, my username is no loger available: user: khratos mail: jpe at slackware-es.com I had some documents on my personal space. Is there a way to recover the account? Regards, -- Jose P. Espinal http://www.eslackware.com IRC: [OFTC|FreeNode] Khratos @ #slackware | #asterisk/-doc/-bugs

Hello List, A few days ago I installed ViciDial on a server, and while looking to the default 'extensions.conf' file, I saw this line: exten => _010*010*010*015*.,1,Dial(${TRUNKTESTast}/${EXTEN:16},55,oT) Can someone point me out to the Asterisk documentation part where explains how to use server IP's as extension number? I could not see it in the ATFOT2 book, and I would

Hello List, I have a little issue with calls placed to a provider declared on sip.conf, because of a not clear (*for me*) behavior of 'remotesecret' parameter. Before continuing, this is my environment: Asterisk: 1.6.2.16.1 OS: CentOS release 5.5 (Final) 2.6.18-194.32.1.el5 Details: I have this block on sip.conf ----- start ---- ... register => john:j0nhp4ss

Hello Folks, What could be producing the following warnings on console, after an installation from source (Asterisk 1.4.41): [May 12 21:36:54] WARNING[15344]: loader.c:434 load_dynamic_module: Error loading module 'res_musiconhold.so': /usr/lib/asterisk/modules/res_musiconhold.so: undefined symbol: cap_set_proc [May 12 21:36:54] WARNING[15344]: loader.c:434 load_dynamic_module:

Hello List, To whom it might concern: I have been working in some SlackBuilds (script for making Slackware Packages) for my personal use, but thought they might be useful for someone else here. Beside of the exceptional distributions used so far (CentOS, Debian, Ubuntu, etc.), you might want to test Asterisk on a Slackware Linux box, as it offers outstanding stability and flexibility as

Hello list, I'm facing a little issue with dahdi attempting to load the OSLEC echo canceller into my current kernel. After compiling dahdi 2.3.0.1 with OSLEC support, I get the following error when set 'oslec' as the echocanceller: DAHDI_ATTACH_ECHOCAN failed on channel 1: Invalid argument (22) - Similar errors are *NOT* present using other echo canncelers. - I tried adding the

Hello List, Is it possible to select ODBC_STORAGE without entering to 'menuselect'? I'm currently building a package for my distro with a little script, and would like to set this option without entering manually to 'menuselect' I know that I could make the script to change the 'menuselect.makeopts' var from: MENUSELECT_OPTS_app_voicemail= to:

Hello, After installing Asterisk from source in Slackware 13.1, I get the following error: Error loading module 'res_config_odbc.so': /usr/lib/asterisk/modules/res_config_odbc.so: undefined symbol: ast_odbc_clear_cache Then a core dump. If I change the /etc/asterisk/modules.conf in order to preload the 'res_odbc.so' module, then the error dissapears, *but* still crashes

I have been trying for a very long time to get asterisk to detect and utilize dtmf tones from my sip clients within my dial scripts. I have set automon=>#9 in my features.conf, I have Dial(....,gWw) in my dial scripts. I have Set(DYNAMIC_FEATURES=automon) as the first script in my extension. I can see the dtmf tones on the wire as SIP INFO packets. Using the Read() app I have verified that * is

Hello List, I was writing something in PHP that connects to AMI and sends a data stream ( example of it: http://slackware-es.com/ami-input.txt ), but the file (voicemail.conf , in this case) does not get fully written. I tried pasting the stream directly through telnet to AMI, and everything *appears* to be OK, but the file is not being completely written. No errors on CLI No errors on AMI

Hello List, I've been trying to compile Asterisk with H.323 support and, after correctly installing PTLib and H323plus (OpenH323), the Asterisk configure script still doesn't detect the dependencies as installed. I know they are correctly installed because after going into "[asterisk-source-directory]/channels/h323" and issuing a 'make opt', it correctly builds

Hello List, I've been working with a set of SlackBuilds that might be helpful for people that want to install Asterisk 1.4/1.8 from source, but, maintaining the possibility to easily (painlessly) update/upgrade/uninstall/patch/recompile Asterisk. The Scripts are located here: https://bitbucket.org/jespinal/slackbuilds/overview You just need to do: $ hg clone

Actually the docs for the Queue application say: 'w' -- allow the called user to write the conversation to disk via Monitor 'W' -- allow the calling user to write the conversation to disk via Monitor couldn't get these to work tho. Does this mean I can do one touch recording with agents, or does it mean I can use the monitor() command? Very confusing... Doug.

Guys. How are filenames determined for automon and queue recordings enabled on queues.conf? I see the names have some tomestamps or something but is there a way to predefine the filenames to use? Thx!

Hi list, Can someone please explain how to get one touch recording (automon) to work reliably? I'm using Asterisk 1.4.14 on a Debian etch system. My current configuration includes the following settings: In /etc/asterisk/sip.conf: [2000] ; Siemens Gigaset S675 IP wireless SIP phone. type=friend secret=1234 context=phones-j dtmfmode=rfc2833 qualify=yes

There is option automon => *1 in features.conf As I understand when *1 pressed during conversation => recording should begin. But unfortunately it doesn't work for me. I use CVS-HEAD-01/27/05 Does anyone has that feature working? Thanks -- Best Regards VladK

I'm using Asterisk 1.4 but my "one touch recording" is not working: feature.conf automon => *1 extension.conf [globals] DYNAMIC_FEATURES=>automon exten => 117,1,Dial(SIP/117,30,jrwW) When I press "*1" on incoming call asterisk is not recording anything. Did I miss any setting? -- Joseph

Hello, Just in case someone hasn't upgraded yet, and is using IAX2. -------- Original Message -------- Subject: AST-2009-006: IAX2 Call Number Resource Exhaustion Date: Thu, 03 Sep 2009 17:47:35 -0500 From: Asterisk Security Team <security at asterisk.org> To: bugtraq at securityfocus.com Asterisk Project Security Advisory - AST-2009-006

Hi there, We used to record all the calls with the Monitor function. Now, I haveimplemented on-demand recording with automon instead... Everything is working fine apart from the generated filename, which as per all docs, should be auto-epoch-caller-callee....but in my case, it is auto-epoch-who_started_record-the_other_end. Thank you all in advance. Regards, Marta -------------- next part

I've got automon up and recording calls on demand from information I found in the list archives, however instead of ending up with one monolithic file, I've got a -in and -out version of the files in my monitor directory? Anyone have suggestions how I could end up with a monolithic file that does what I want?