similar to: Asterisk running on DMZ (no NAT) PROBLEMS- OPTION message is out of State

Displaying 20 results from an estimated 1000 matches similar to: "Asterisk running on DMZ (no NAT) PROBLEMS- OPTION message is out of State"

2005 Oct 10
3
Billing/SPA-841/CDR Log
Hi list, I have a couple of questions related to asterisk billing and the generation of cdr logs. I've searched the wiki but have not found my answers, hopefully you guys can help. 1) When are asterisk CDR logs _normally_ generated? When the call arrives, when the call hangs up, or both? I have looked at the records created and it seems to only generate it at the time the call is
2005 Oct 05
2
From Database, PHP-Webinterface -> TO flatfileconfiguration
Hi. I've started working on a PHP-project that generates the configuration files i need based on what's in my MYSQL database. I can add, delete and edit users from the web. I can set up exactly the dialplan i need by arranging the users in a firms and groups if needed. I've also set up a java servlet so that i can get asterisk to reload by pushing a button from the web-interface. The
2007 Aug 27
4
Prepaid Billing: A2Billing, AstBill, ASTCC
Hi List; I need to use an prepaid billing system with Asterisk, and I do not know which one is more stable and integrated with Asterisk? A2Billing or AstBill or ASTCC? Also, from where I can download it and ready about its configuration? Regards ITS IP Telephony and Contact Center Engineer Eng. Bilal Ghayad Mobile: 009659849460
2010 Nov 25
4
[PATCH]improve suspend_evtchn lock processing
While doing migration, sometimes found suspend lock file was not unlinked in previous operations, then there is an obsolete lock file in place, which causes the current and later migration cannot get lock. That happens seldomly but do happen. After checking the source code, I found there are some places that potentially cause lock file unlinked, including: 1) in lock_suspend_event() function,
2005 Oct 10
11
Open Source Content Management System - Joomla
There was some discussion in the past about which one is the best Content Management System that can be used in conjunction with Asterisk. Mambo was supposed to be the best out there under GPL. The guys who developed Mambo have a new product now - Joomla. I am using this and it appears to be better than Mambo in many respects. Read the gist about Joomla below. ------------- If you've read
2003 Dec 11
2
Cochran-Mantel-Haenszel problem
Hello, I've tried to analyze some data with a CMH test. My 3 dimensional contingency tables are 2x2xN where N is usually between 10 and 100. The problem is that there may be 2 strata with opposite counts (the 2x2 contigency table for these are reversed), producing opposite odds ratios that cancle out in the overall statistics. These opposite counts are very important for my analysis, since
2006 Dec 18
3
Billing solution
Can anyone recommend a call accounting solution with rating for post paid billing that works well with asterisk using the account code or any other info from the CDR? I don't want the billing software to any phone calls for me, therefore any solution that modifies my extensions.conf is out, nor does it have to allow for customers the ability to log in to check their usage/balances. I have
2006 Nov 30
2
Billing Software
We are looking for an offline billing solution. We have a couple of particular requirements: 1) Since it's offline, we need to be able to import the CDR. 2) A way to support account credits based on referrals. Meaning, that if a member refers a new account, that member would get a free month of service, or similar type credits. 3) Generate invoices in either HTML or PDF format so they can be
2005 Aug 05
3
Realtime IAX
I am using Asterisk CVS from last week and have been using Realtime SIP for a couple weeks now without any problems. Yesterday I decided to turn on Realtime IAX but I am having problems dialing to my long distance providers like Voicepulse, Sixtel or Nufone. I get the following: -- Executing Dial("SIP/2001-3761", "IAX2/password@voicepulse/19566680301") in new stack
2008 Jan 25
2
Asterisk Billing
Hello, I'm checking some Billing Software for Asterisk. In opensource I only know (the name, I haven't used) AstBill. What other software should I check with similar capabilities? Thank you! -- Carles Pina i Estany GPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona
2007 Apr 17
1
Asterisk Billing
Network Configurations Block D, Surrey Park, Barham Road, Westville, 3610 Helpdesk: (086) 163-8266 Tel: (031) 266-1563 Fax: (031) 266-4206 Hi List. I'm in need of something that will allow me to analyze cdr details either via .csv or mysql that will give me call durations as well as call costs. This is so that we can see in what areas/staff are costing what per month/week on outbound
2005 Oct 10
3
country code list
I was wondering if anyone has put together a comprehensive list (that is reasonably maintained) that lists country codes, landline numbers, mobile numbers, etc. The particular requirement is for a dialplan to know what is going to be charged to whom. For example, mobile and landline rates are the same in the US the US has a unified numbering plan of 1NXXNXXXXXX, while the UK has: 441xxx
2005 May 31
1
Built-In Transfer Questions
I've read the Wiki on using asterisk's built-in transfer options (#8 and #6). They work fine but how does one cancle an attended transfer? Example: I have person on phone, I hit #6 to being att-transfer. I enter Sally's extension. I let it ring for a few seconds. Sally never picks up but her voicemail does. How do I hangup her voicemail and resume the previous call? The example on the
2011 Mar 05
3
Prepaid Billing other than A2Billing
Hi All; Any one advise for open source prepaid billing other than A2Billing that can work with Asterisk and it is rich by features (for large business)? Regards Bilal
2007 Mar 06
2
Manager.conf '127.0.0.1 unable to authenticate'
Every few seconds I get the following message: == Parsing '/etc/asterisk/manager.conf': Found == Connect attempt from '127.0.0.1' unable to authenticate I'm trying to track down where it's coming from. I've used TCPDUMP & NGREP to monitor 127.0.0.1, no data's flowing. I've tried loading Asterisk with no modules, tried loading with a naked
2020 Mar 09
2
Manipulating Arch specific code generator state
Hello all on the list, I’m developing a backend for the 65816, however, I need a way to store some state, as processor flags can affect how instructions operate (including the length of some), as well as the calling convention. I need to track for each of these flags (x, m, and e) Set, Unset, Indeterminate. I was wondering if there was a nice way to store this with the MBB, so I can make sure
2006 Feb 27
1
billing - different tarif per phone
Hello, I would like apply different call rate (tarif) per outgoing number (or group of phones, based on prefixes), I'm playing with astpp, but seems, that this feature isn't available here, can you recommend any other open-source billing (A2billing, AstBill?), that this can do? thank you! PJ
2013 Oct 16
1
the print processor does not exist
Hi all, When i try to install one printer shared by a samba server (3.6.6) in a Windows 7 (x64 home premium and professional) clients, i get "The print processor does not exist" and i can not install nothing. All printers are mapped through .bat file, using "net use" command, specifying user/password. I always install like this: in run window (win+R), type the samba server ip
2007 May 10
1
asterisk SIP domain (in LAN or DMZ)?
Hello I want to use Asterisk to implement a SIP Domain allowing my clients to do URI dialing and receive calls from the Internet through URIs and ENUM. My question is, should I put my Asterisk outside the firewall (in the DMZ) to allow connections to the Internet? Or should I have it inside my local network and put a SIP Proxy (like Openser) in the DMZ to implement the SIP domain? Thanks
2008 Apr 23
2
prepaid on the trunks
if i have this setup: [sip users] -- [asterisk] --- [as5300] --- [pstn] asterisk will talk to as5300 using sip. i will use as5300 as a trunk on the asterisk so sip users can call out to pstn. what i would like to is do prepaid on those trunks, not on the sip users. sip users can call any other sip users . i want to do it that way coz i'm trying to build a multi-tenant pbx, and i will use