similar to: Asterisk SIP realtime and realtime DB tools

All, I am looking at a little support on this, as I haven't found it on google yet. I have had this work on Callweaver, but am moving to Asterisk for a variety of reasons. My dial plans, and everything else transferred perfectly, though I am not sure they are 'correct' for Asterisk 1.6.1, with simple things like SIP users outlined in the sip.conf file, not in the users file,

All, I have a Cyberpower UPS that I have been working with for about a year. I have used NUT in the past with good results, but as of late have been seeing issues with nut talking to the UPS. I will start with the information on the server that is running NUT, and has the UPS connected VIA an RS232 cable. O/S : Fedora 9 Kernal (uname -a): Linux haruhi 2.6.26.5-45.fc9.x86_64 #1 SMP Sat

All, I have a Cyberpower UPS that I have been working with for about a year. I have used NUT in the past with good results, but as of late have been seeing issues with nut talking to the UPS. I will start with the information on the server that is running NUT, and has the UPS connected VIA an RS232 cable. O/S : Fedora 9 Kernal (uname -a): Linux haruhi 2.6.26.5-45.fc9.x86_64 #1 SMP Sat

Ok, First I want to start off with this isn't the first icecast I have cloned, but it is the first where I have not only jumped Fedora release versions, but arch types as well (i386 - x86_64) and I am having strange problems. The Icecast versions are the same between the old and new servers. The biggest being is that the server binds to whatever port it feels like instead of the bind port

Karl Heyes wrote: > Seann Clark wrote: > >> The biggest being is that the server binds to whatever port it feels >> like instead of the bind port specified. The rest of the issues I have > > a random port bind is a new issue. Can you show us the > netstat -tnlp | grep icecast > for the xml provided? Thu Jul 03-13:54:17-root at haruhi-new:~> netstat -tnlp | grep

All, I just noticed this in my logs, and am rather lost as to what module it pertains to. I would assume pseudo-realtime priority for the process, but I am looking for a little confirmation from the group: [Apr 28 12:28:36] WARNING[20773] asterisk.c: The canary is no more. He has ceased to be! He's expired and gone to meet his maker! He's a stiff! Bereft of life, he rests in

All, Recently my area got slammed by very bad storms (something among the lines of 120 MPH winds and lots of hail) which knocked out a quarter of the power in my city. When I arrived back to my home office, I naturally found everything powered off and silent, but what I did notice is my UPS still had full battery. While booting the systems back up after power was restored, I didn't

All, I just had a power failure that caused me a little drama. This spurred me into trying to get NUT up on my cyberpower 1500 again. For about 6 months I tried to get it to work with the new power panel drive with no luck. Today I removed all the old files for NUT and installed the newest version I could find (2.4.3) and it works perfectly now with my UPS. Hopefully now I

All, I am toying with an idea of using an AGI to be able to 'call' my phone, or phones, in case of severe weather warnings. I have been tinkering with a script that reads from weather underground for the forecast, based off a PHP version of a weather AGI I found on the net. It seems rather trivial to have the AGI as a script, that does nothing unless a condition is met, and

All, I have been fighting with my dialplan for hours now, and google searches talk lots but offer nothing in terms of explication for this. I have my SIP peer set up and working with Broadvoice: [sip.broadvoice.com] type=peer user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=5555551234 secret=password defaultuser=5555551234 insecure=port,invite context=broadvoice

Hi Guys, Just did an Asterisk 1.6.x (repo install) and FreePBX (source install). When trying to dial a number, I get this: tel*CLI> Use of uninitialized value in hash element at /var/www/html/panel/ op_server.pl line 3367. Use of uninitialized value in concatenation (.) or string at /var/www/html/panel/op_server.pl line 3372. Use of uninitialized value in pattern match (m//) at

Does someone here configured RealTime Extensions using ODBC connecting to Oracle DB? Im having a problem in dialplan patterns, it doesnt work. Pls. help! -Chris __________________________________ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com

Hi All, I'm testing different ways to implement a LCR/OCN tabe to shift calls to multiple carriers for better rates. I'm using realtime mysql database access, asterisk 1.2.9.1 with mysql 3.23. Scenario 1: I send outgoing calls with with a Goto statement into the context with the realtime switch to dip into the mysql lcr database, currently I have ~13K records/routes. The database

All, I am looking into open source idea's for something I play with on the closed source side. What I am thinking is to get two Asterisk PBX's behind a single SIP proxy to load balance calls inbound, and potentially outbound to an external sip provider, with the potential of multiple provider type lines (SPA3102, and a sip provider) that allows the call to persist. What I am

Hello all, I use asterisk 1.6.0.13 with realtime DB. when a phone (SIP) receive a new voicemail, asterisk update with a NOTIFY the number of waiting message to the phone, so all is ok . if I use the voicemail application to consult and to delete voicemail, asterisk again update correctly the number of message to the phone. now, if I use a web application or script on aastra phone to delete a

Hi, Asterisk 1.4.31 here. I have a queue with members defined, and those member have member names "member 1", "member 2", etc. They are in a realtime DB. When I modify those member names (column membername) the changes aren't reflected in the queue status ("show queues" from cli. They aren't reflected when a new call comes in, or when I reload the

Hello list, I'm using asterisk 1.6.2.11 with realtime SIP (mysql DB). I notice that when the SIP peer registers, the fields 'fullcontact', 'ipaddr', 'port', 'regserver', 'regseconds', 'lastms' are filled with values. But after a while, these fields become empty. Asterisk CLI shows : asterisk*CLI> sip show peers Name/username

Hello list, scrambling different pieces of info together I've come with the following : I want to have my "register =>" statements in a MySQL-database, so I've made the following table. table ast_config : id 1 cat_metric 0 var_metric 0 commented 0 filename sip.conf category general var_name register var_val username:password at sip.provider.net In ext_config

OK, I have just discovered what may be a conceptual flaw in realtime extensions. DB schema as follows: uniqueid, context, exten_id, priority, application, application_data Primary key is uniqueid with an compound index of context and exten_id data looks something like this: xxxxx, sip, 5555551212, 1, SetVar,ext=${EXT} xxxxx, sip, 5555551212, 1, Goto, officevm|2|1 The database is MySQL

Ok, sorry about the top post. Compiled and installed 3.2.3, found a few stilled borked configs with my MPD, and fixed that, and both Icecast (after playing with ownerships that were deleted when I de-yum'ed the icecast RPM from the Fedora repo's) and icecast is working. I will clean up the config to remove some of the things that were mentioned though. Looking at the interface for the