similar to: Comments on Billing reconcillation with providers

Hi, I realise this might be an insane noob question, but I'm on a huge brain freeze, and I'm trying to decide this: Is Asterisk a SIP Gateway or SIP proxy? -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

Hi all, I am using Asterisk CVS, and I am getting a huge delay in dialing SIP. This Asterisk box is taking calls from a PABX over ZAP, then dialing SIP users. So, a user '0251' dials from his phone, the PABX sends it the my Asterisk box, no delay, then I get a 15 sec delay, before it actually dials the end SIP user. 1 -- Accepting call from '0251' to '0834541083' on

I have installed oh323 channel driver (finaly! :)). I head some problem starting * so I have put the smallest possible oh323.conf file to se what happens. When I don't put available codec's in oh323.conf (*1) Asterisk starts but he also disables h323 channel because there are no available codec's (*2). When I put codec (*3) Asterisk doesn't start (*4). What have I done wrong? I

Hey all, I have read on voip-info.org that to configure MoH asterisk requires the use of mpg123. I have installed mpg123 and restarted asterisk. But, when i put a call on hold i get this error: May 25 14:13:03 WARNING[1872]: res_musiconhold.c:865 local_ast_moh_start: No class: default Can you help, Thanks yusuf

Hi, I am running Asterisk 1.4.4, and needed to setup sip accounts for someone to call my server and place calls. However, he has multiple IP's that he comes from, and since I authenticate him of his IP, I did this, and it works. [vz1] context=outbound type=friend host=x.x.x.x disallow=all allow=alaw canreinvite=no [vz2] context=outbound type=friend host=y.y.y.y disallow=all allow=alaw

Hi all, I have Asterisk 1.2.10, zaptel 1.2.7, spandsp-0.0.3pre22 compiled, and a Sangoma A101, and when I make a call I get this: Jan 8 13:04:06 DEBUG[12252]: chan_unicall.c:2000 unicall_exception: Exception on 19, channel 1 Jan 8 13:04:06 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 <- 1101 [1/ 40/Seize /Idle ] Jan 8 13:04:06 WARNING[12252]:

Hey, does anyone know why i'd be receiving: Aug 28 19:40:04 DEBUG[1875]: ##### Testing 66.27.233.241 with 10.0.10.0 Aug 28 19:40:04 DEBUG[1875]: Target address 66.27.233.241 is not local, substituting externip I get tons of them, usually when the phone is registering/calling/receiving calls. Thanks! Chris

Hi all, I came across a new(to me that is) issue. I want to know from others what they have done to resolve this. I have a 4 port digium card with FXO's, and connected to each FXO is a premicell. When I dial the premicell, after about two seconds is says 'ZAP/1 answered', then it takes a few more seconds for the call to hit the cellular network, before the cellphone starts to ring.

Hello All, I would like to upgrade my recently installed Asterisk 1.2.21.1 to Asterisk 1.4.8? My OS is CentOS 4.5 with Linux 2.6.9-55.0.2.plus.c4smp #1 SMP Fri Jul 6 05:25:07 EDT 2007 i686 i686 i386 GNU/Linux Is there any detail step by step procedure to uninstall the current version and install Asterisk 1.4.8, Zaptel 1.4.4, Libpri 1.4.1, Addons 1.4.2? Cheers, Nitesh

I need some extensions logic assistance, I'm trying to dial out one of multiple SIP trunks, in sequence. I need to detect a busy SIP trunk(I only allow 1 call per trunk) and roll over to a second or third depending on that busy status Here's what I've got for a macro thusfar, but it's not working(fails if the 1st trunk is busy) extensions.conf: [globals] trunk_1 => SIP/trunk1

Trying to get SIP realtime working here... I'm connected to the database... *CLI> realtime mysql status Connected to vox180internal@db1.ipt.XXX.com, port 3306 with username voxadmin for 6 seconds. I can get information for the extension in question... *CLI> realtime load sipusers name 2944093 Column Name Column Value

Hello, I've discovered that in Italy ISDN lines can be programmed to generate a "billing pulse" every n seconds (it dipends from the pricebook). The pulse has these figures: frequency .................................................................... 12 kHz ? 1% level .......................................................................... 200 mVrms on 200

Hmmm, perhaps I am the only one who doesn't trust their telco (I doubt it) but... I have the rates that I currently pay my telco, and would like to extract my CDR's and add an additional field displaying the actual price paid for the call. I would like to do this based on destination phone number, and outgoing channel. However, I have a few difficulties: 1) I pay a different rate for

Can anyone share any links regarding packages to do Call Detail Record (CDR) analysis from the CDR Master file? Login-distance reconciliation, billback, and data presentation are three primary areas of interest. Thanks in advance for your help! --Nick -- Nick Eggleston Consultant Data Communications Consulting, Inc. 6320 Rucker Road, Suite E Indianapolis, IN 46220 317/726-0295 x18

On Sat, Oct 10, 2015 at 03:10:26PM +0300, Ady via Syslinux wrote: > From: Geert Stappers <stappers at nero.gpm.stappers.nl> > > > > com32/mboot/map.c: removed trailing spaces > > > > They were introduced by the patch for ELF64 support. > > > IMHO, the trivial trailing-space cleanup could be included in the same > commit too, instead of adding an

Hi, my name is Yusuf, I just join with this groups. I have using samba PDC with LDAP as backend. I have a problem to change user password from web. I tried using sudo smbldap-passwd, change permission every file so apache can read / execute that file, but I'm still can't change the user password. Is there any way to change the password only with change the LDAP password (using

> Looking around, there appear to be a lot of small places needing some > whitespace cleanup/reconciliation. > Just to be clear, I am not criticizing, but actually asking... Are those white-space characters impacting the binaries being built? Would they affect common users? Are those characters affecting developers? For instance, do they affect some git command? Or, do they make

Hello; I've been playing with a nifty Open Source java based report writer called Datavision (datavision.sourceforge.net) and I've managed to write enough logic to calculate phone bills at different rates from the MySQL cdr's. (cdr_addon_mysql) Eventually I want to have sets of rate structures for each user of the system - so I can bill client A at 3 cents a minute and client B at 2

Hi guys, i've installed asterisk to handle multiple voip accounts. I've looked at CDR configs, and managed to have cdr-csv files growing after each call. It would be easier to check my locak asterisk cdr's than logging into each account and check them at the provider website. i found that if i ring my sip softphone from my ata, bill seconds are counted correctly. however, if i

Hi, i have been using asterisk CVS 19/07/2005 and asterisk-oh323-0.7.2. I now want to use oh323 with Asterisk 1.2.4+. Can anyone tell me what versions of oh323(and pwlib and oh323) they got to work with Asterisk 1.2.4+. -- thanks, yusuf