similar to: AMI input streams limit?

Hello List, I have been working on a little PHP software that uses AMI's UpdateConfig command in order to modify some of it's config files. I was working with 'Asterisk 1.4.22.1' and everything was working. After upgrading to 'Asterisk 1.4.23.1' I receive a lot of errors of the type: ERROR[11505]: utils.c:966 ast_carefulwrite: write() returned error: Broken pipe

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, 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

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, 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 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, 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

Hi, I'm seeing weird behavior with AMI where no events are output until some input is detected (can be an empty line), at which time all the buffered output is spewed out at once. I am maintaining multiple Asterisk installations, and with one installation I have run into a weird buffering problem with AMI. The version is 1.6.1.11 in this particular case, which I am running at multiple

I'm using a perl daemon i wrote to connect to AMI and perform actions. The daemon connects to asterisk via AMI at start up. Is there anyway to check if the AMI connection is still alive, for example every 2 seconds. if the connection is not alive, re-connect to AMI? Also, does AMI timeout after a certain amount of time of not sending commands? -------------- next part -------------- An HTML

Hi, i'm currently using Originate command on AMI, i can call a certain channel like a SIP user SIP/1000 then once 1000 is answered it dials out to amobile or landline. Would just like to know if i can use AMI to dialout to a mobile or landline first (instead of SIP user) and once answered, dial another mobile or landline again. If not is it possible to call a macro from the AMI? i think

Hi Is there a way I can stop Manager logged on/off messages from going to the console/logs without losing all the other information I need? Regards Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062

Hello. I've been trying to use the AMI to originate phone calls. I'm trying to call the SIP phone 'zoiper' with the SIP phone 'yziquel'. So, the AMI interaction is: > Action: originate > Channel: SIP/zoiper > Exten: yziquel > Priority: 1 > Timeout: 30 > Context: internal > > Response: Error > Message: Originate failed > > Event:

Hi Using AMI how can I get the presence feature.Below are the requirement. --> List of all users in the PBX including analog and SIP including registration status. --> Status(BUSY or available ) of all users both analog and SIP Please help on this.. Thanks Nikhil

Hey Gang, Trying to figure out how I can do the following (have each part working individually but drawing a blank on combining) 1) click on-screen which sends an AMI originate (works fine) 2) the originated call is to an internal extension that looks up the number to be dialed (works) 3) then via Perl, adding in a SIPAddHeader for answer-after=0.. (works separate from the above) What I

Hello, I'm using an AMI Originate command to send a fax. The fax is sent by a script, and I'd like my script to send the fax, wait until it has succeeded or failed, then exit with an appropriate error code (it is driven by a mail system, so the exit code will tell the mail system whether to retry the fax later). The script works great if the fax succeeds, or if the line is busy or

hello all, I have installed centos 5.5 ( linux text) and I have updated it with # yum install bison bison-devel================?ok # yum install ncurses ncurses-devel==========?ok # yum install zlib zlib-devel===============?ok # yum install openssl openssl-deve=======?ok # yum install gnutls-devel============ ==?ok # yum install gcc gcc-c++============?ok # yum install newt

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, Maybe I'm mistaken, but, shouldn't the 'fname_base' variable of 'Monitor' application affect the file name generated through 'automon' feature? I initialized this variable with a value as follows: Set(fname_base=auto-${CALLERID(num)}-${EXTEN}-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}) a. Should I use 'fname_base' in uppercase (FNAME_BASE)?

Hi All, I've just upgraded our CRM and it has an Asterisk Integration Module that I would like to test out. The CRM is running on one of our hosted servers in the cloud. The Asterisk server is running in my office. I am running Asterisk 1.4.21.2~dfsg-1ubuntu3. Reading the page http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20manager.conf got me a little concerned

Hello, I am using Asterisk 1.4 version. How to dial multiple numbers per second through asterisk manager???? Thanks and regards