similar to: CDR problem

Hi folks, I've just updated my * and I noticed that from the update the uniqueid field into mysql, is not written and ASTPP do not charge the calls. I got an eye to cdr_mysql.c and I found that at line 212, into one insert query, uniqueid is missing. But I can be wrong. In any case, somebody got same problem? Any suggestions? Thanks to all. -- .:FaberK:. -------------- next part

Does anyone have experience setting up ASTPP? I have an Asterisk server in my office that I also give access to some friends and family that live outside Mexico so they can make local calls. I want to keep track of the costs and I only need to use ASTPP to rate the calls, not for calling cards or anything else. I found the documentation a little vague on the details so after setting up and

does anyone get a hold of the SPA-941 Provisioning Guide? i tried call Sipura's tech support, seems like none of them heard of the term "remote provisioning". they kept refering me to their web site which i've check thoroughly, and could not find any documentations on the SPA-941. finally they gave me a phone number to call, which appears to be a fax machine. that's when i

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

Good Day, I'm finally getting around to officially announcing ASTPP. For the last 6 months, I've been working on converting ASTCC into a decent billing package for asterisk. I'm just finishing up fixing a few bugs before the 1.0 release and would appreciate if there would be a few who would be willing to do some testing on the software. Here is a list of features: Provide call

our * crashed twice in a month with segmentation fault & a core dump. here's the stack trace: #0 0xb7e11965 in mallopt () from /lib/tls/libc.so.6 #1 0xb7e10c43 in malloc () from /lib/tls/libc.so.6 #2 0xb7e17090 in strdup () from /lib/tls/libc.so.6 #3 0x08057ada in ast_verbose (fmt=0x0) at logger.c:879 #4 0xb7b29009 in moh_files_release (chan=0x9455ca0, data=0xb657be48) at

so what should "astpp db" be exactly, where can i find its name? what should i write there? Thanks again.. > The Database field should contain the name of the astpp db, something > along the lines of "astpp" is what I would put in there. Here is a fixed > version of the script. It did not post properly to the wiki: >

hi. i connected Asterisk to an NEC NEAX system with a crossover T1 cable and the Digium TE405P using E&M wink signaling. the connection's ok. however when dialing from the NEC to the Asterisk. most of the time the Asterisk only sees the first digit of the dialed number(which is 4 digits). some time if i dialed the 4 digits very fast it might get through. seems like there's a timming

Hello, I'm setting up AST Post Paid application, is there anybody who set up astpp ? I followed the directions, i visited the astpp admin page in my web browser. But i couldn't setup the brands and routes etc. "Database unavailable -- please check configuration" appeared on the top of the page, so i went to "configure" section, I filled in the blanks according to my

hi folks. i have the following puzzle: when i call certain cell phone# using a regular phone & POTS. the called cell phone co. usually return a message such as phone travel out of range or phone is busy etc. if the phone is unreachable. now when i have the following setup: sip phone -> asterisk -> PRI -> phone co. i call the same cell# and if it's unavailable. the PRI return

hi folks. we recently deployed 10 Cisco 7960G w/ SIP firmware 7.3 on our network and we start having problems of dropping calls (actually the calls wasn't dropped it just the sound was muted for about 5-10 seconds, but most users will think the call dropped and hangup/redial). i've check the console output. there was a lot of messages like the following: Sep 28 15:00:49 NOTICE[8182]:

i recently bought an SP4000 conference phone but having problem provisioning it using ftp, every time it just hangs at "Updating initial configuration..." screen. when i switch it to tftp, it'll work fine. i though it was bootrom/firmware issue so i upgrade it to bootrom 3.2.2/sip 2.0.1 but it makes no difference. any thoughts? p.s. i'm using debian sarge proftpd 1.2.10 and the

hi folks. we've experienced some weird problems lately. we have about 600 SIP phone on a single system running *1.4.26.2 for about a month. recently there was massive UNREACHABLE messages like this one showed up: chan_sip.c: Peer '2699' is now UNREACHABLE! Last qualify: 1252 then they all became reachable again in a few seconds. sometimes it last for couple minutes. but sometimes

I am copying the Master.csv file to another server and importing to mysql. I am looking for a simple billing application that will produce a bill for a give account code for a give period, based on a rate table. Is this available? -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271 Cell: 264-235-5670 Yahoo IM:

hi folks. i'm experimenting with iaxmodem + hylafax using DID to determine where to send the fax to it's final destination. however i have difficulties passing the DID information from iaxmodem to hylafax. in extensions.conf: exten => _XXXX,1,Dial(IAX2/iaxmodem0/${EXTEN}|20|r) exten => _XXXX,n,Dial(IAX2/iaxmodem1/${EXTEN}|20|r) exten => _XXXX,n,Busy exten => _XXXX,n,Hangup

Hello All, I am having trouble finding a specific * piece of software so I thought I would see If you guys can help me get my terminology clear. First off let me premise this with "no, this is absolutely not for doing call marketing". I need to make my Asterisk box call a group of people and play them a message. My company deals with education so we need to do follow ups if students

hi folks. recently some of our customers complained about bad voice quality on the phone system. i looked at the logs and found a lot of these: [2012-11-03 08:26:38] NOTICE[11305] chan_dahdi.c: PRI got event: HDLC Abort (6) on D-channel of span 1 [2012-11-03 08:26:45] NOTICE[11305] chan_dahdi.c: PRI got event: HDLC Abort (6) on D-channel of span 1 [2012-11-03 08:26:54] NOTICE[11305]

Hi all, I have installed this package onto my Debian and placed the files i want the Cisco 7960 phone to get from the tftpdboot directory. But it doesn't seem to work. Are ther any special settings I should do to this server? Many thanks for all your help, Christian

I'm running Asterisk 1.4.18 and having a problem with the clearglobalvars option. I have a NIGHT_SERVICE variable which I initially set equal to off. I then have an extension they can dial which will toggle that variable. My problem is when you enter the CLI and type "reload", it resets to "off" again. I've tried setting the clearglobalvars=no as well as just

Greetings all- During some recent testing and debugging, I wanted to use the 'fxstest' application. However, I found it hasn't been built when doing the standard 'make, make install' shtick with dahdi-linux-complete-2.3.0.1+2.3.0... Can anyone tell me how to build fxstest? Thanks! --Tim