similar to: SIP contexts being confused

how to input multiple .txt files? A data folder has lots of .txt files from different customers. Want to read all these .txt files to different master files: such as: cust1.xx.txt, cust1.xxx.txt, cust1.xxxx.txt,.............. to master file: X.txt cust2.xx.txt, cust2.xxx.txt, cust2.xxxx.txt,.............. to master file: Y.txt Thanks! [[alternative HTML version deleted]]

So there are a couple parts to this question. I am trying to implement the rpart/random forest algorithms on a transaction lists. That is to say i am trying to train models in order to deduce what are the most predictive transactions within a customers history in order apply this model to future test data and identify accounting irregularities(ie. this account had x and y so they should have also

*** If anyone has a better way of doing this, please post to the list. I hadn't seen anything on this list or in channel.c/chan_local.c - which prompted this email *** I'm not sure how many VoIP providers out there are using Asterisk as a service platform like we do, but I thought I'd share an experience with call looping that was racking my brain with the list. One of the

Should be mailbox = 7752365815@vpbx-wpti Best Regards, Ben Bawkon --------- Original Message --------- From: Bruce Komito To: <asterisk-users@lists.digium.com> Subject: [Asterisk-Users] voicemail notify problem on sip extension Sent: 5/19/2004 4:27:51 PM I'm having a problem with the voicemail notify feature. Although I have the voicemail box configured for the sip extension, the

I try to change something in ASTCC, but I am now totally blind, .... I hang on one line now. I changed: vpbx:/var/lib/asterisk/agi-bin # diff astcc-original.agi astcc.agi 22c22 < # exten => _00XXXXXXXXX,1,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN}) --- > # exten => _00XXXXXXXXX,1,DeadAGI(astcc.agi,${CALLERIDNUM},${TARIFF},${EXTEN}) 35c35 < # exten =>

vpbx*CLI> sip show peers Name/username Host Dyn Nat ACL Port Status 621/621 192.168.250.76 D N 5060 OK (65 ms) 626/626 192.168.250.109 D N 5060 OK (180 ms) 616/Ronald Softphone (Unspecified) D N 0 UNKNOWN 615/Ronald office 192.168.250.103 D N 5060 OK (41

I tried to add the databases from the localhost to the database server and changed the every /etc/asterisk/*.conf from host=localhost to host=192.168.10.10 (my dababase server) When I restart asterisk, I do not get any errors, but after a phone call I see: Jun 7 18:11:56 ERROR[7877]: cdr_addon_mysql.c:400 my_load_module: Failed to connect to mysql database cdr on 192.168.10.10 Or if I try

I am having a problem with the Vmail Web app and was hoping someone has some advice. I have been able to install vmail.cgi. I am able to bring up the vmail web page and login with out problems. However, I never see any vmails listed even when there are vmails present in the directory [root@linux-home INBOX]# ls -l total 40 -rwxrwxrwx 1 root root 255 Dec 2 14:45 msg0000.txt

I tried to install astguiclient and it gives me for each follow page: Object not found! Looking into the apache log file I find: [Sun Feb 27 16:18:30 2005] [error] [client 192.168.250.108] File does not exist: /srv/www/htdocs/astguiclient/method=POST, referer: http://vpbx.elmit.com/astguiclient/phone_stats.php?extension=gs102&server_ip=192.168.250.20 I have uncommented the ### If you have

I have now a very strange situation. Asterisk dies exactly every hour at hour:09 !!! crontab has the entry: 10 3 * * * root /usr/sbin/asterisk-restart >/dev/null 2>&1 It seems that there is a time difference between mail server and asterisk server so that it might be "synchronized" to the crontab entry. *CLI> show version Asterisk CVS-HEAD built by root@vpbx on a

Hi, all. I've been searching google, bug reports and forums and have looked in all the asterisk-users list archives back to 2003 but haven't seen an answer to this, so thought I'd post here. The problem seems to be that Asterisk 1.6.0.5 is sending backslashes (needed to escape commas and so forth in 1.4.21.2) as *literal* backslashes to Mysql, so that Mysql gives a syntax error

I have added in my zone file; _sip._udp.elmit.com. IN SRV 20 0 5060 vpbx.elmit.com. As I understand it should mean that any sip connection to <anyname>@elmit.com should go to the udp port 5060 at the host vpb.elmit.com. In Asterisk's extensions.conf I have in the context [default] exten => ronald,1,Dial(${PHONE_615},60,tr) exten => ronald,2,Voicemail,u615@office exten =>

I have a problem that is almost certainly nat-related, but I can't figure out what's happening. Since moving the Sipura behind a NAT server (Linksys), I am no longer able to call between the two lines on the same Sipura. When I dial one extension from the other, it rings, but immediately after I pick up the ringing phone, the call is uncerimoniously dumped. I can tell the call

I try to install astcc. Make install gives me: DBI connect('database=astcc;host=localhost','astcc',...) failed: Unknown database 'astcc' at ./astcc-admin.cgi line 67 DBI connect('database=;host=','',...) failed: Access denied for user: 'root@localhost' (Using password: NO) at ./astcc-admin.cgi line 60 Ignoring that (!!! ;-) ) and going to the

Below is my extensions.conf for the fax part [incoming_28345474] ; ;******************************************************************** ; BEGIN - Inbound call handlers ;******************************************************************** ; exten => 8862100,1,NoOp(${CALLERID(num)}) exten => 8862100,2,Background(if-u-know-ext-dial) exten =>

Am 2016-02-24 um 14:15 schrieb Sketch: > On Wed, 24 Feb 2016, Stefan G. Weichinger wrote: > >> I set up a test VM now, same OS and software, with >> >> idmap config CUST:range = 10000-99999 >> idmap config CUST:backend = rid >> idmap config *:range = 2000-9999 >> idmap config * : backend = tdb >> >> This *seems* to work fine

I got a new motherboard and upgraded Asterisk, ... However, I am not lucky enough to get it running again. To compile went ok, besides this remark at the end of asterisk: WARNING WARNING WARNING Your Asterisk modules directory, located at /usr/lib/asterisk/modules contains modules that were not installed by this version of Asterisk. Please ensure that these modules are compatible with

Hi, I'm working with this issue for a while, Now I already solve the dialplan issues, but I still have a question about the Callgroups, I read at www.voip-info.org that , there is a 63 limit of callgroups. And I'm wondering why?? and if the 1.2.0beta version supported more than 63 Groups?? (I did'nt find any Changelog for 1.2) or If not There is any unoficial patch for that ? Thanks

I'm working on building a pbx that will allow us to use our cellphones as extensions (to some extent) The dialout is working fine. What I would like to do is have an inbound cellphone call appear as if it were an extension. So right now if I call in from cell #9995551212 the caller id is 9995551212 but if I dial extension 30013 it will call cell #9995551212. I would like to change the caller

Using Dahdi/PRI, I end up with channel names like 'DAHDI/i8/9995551212-4d6B', but when I do a 'core show channels' it cuts off those names to only 'DAHDI/i8/9995551212-'. This is the same for AMI. Is there a way to get the full channel name within AMI? I'm using asterisk 11.7.0 Thanks, -Justin -------------- next part -------------- An HTML attachment was