similar to: Dial option limit call duration

the gdb of the core taken from the asterisk as the time of crash is as below I run asterisk-1.2.5 on fedora core 3 with chan_ss7 can someone help out? #0 ast_var_name (var=0x1) at chanvars.c:71 71 if (var->name[0] == '_') { (gdb) bt #0 ast_var_name (var=0x1) at chanvars.c:71 #1 0x0808934e in pbx_builtin_getvar_helper (chan=0x0, name=0xf5bc2d46

Hi, I?m playing with the CVS head time limiting at Dial application, it just works fine but the only problem is that the caller isn?t hearing the warning message. I?m using a Cisco 7960 as the caller and a Polycom 500 as the callee. The audio is passing through Asterisk: -- Executing Dial("SIP/8992-9712", "SIP/8988|20|L(10000:2000)") in new stack -- Limit Data: --

Dear All, I tried to use 'L' option on my dial command. I set the x to 65000(65 seconds), y to 60000(60 seconds), z to 30000(30 seconds). The max calltime should be 65 seconds, and it will play "beep.gsm" at 60 seconds left. And repeat the beep at 30 seconds left. But the call will be hangup by system at 60 seconds left. In another word, when it plays warning file, the call

Hello All, I manage to configure Asterisk RealTime and now it loads the SIP users/peers from MySQL DB. The table I am using is of A2Billing DB "cc_sip_buddies". Now the only problem I am facing is incoming calls are failing... The ATA which is assigned this DID number is behind NAT and according to Olle's explanations he said "*there's no support for NAT keep-alives

Hello all, We have a customer who upgraded from Asterisk 1.6 to 1.8, and pickup groups which previously worked fine have stopped working. Can anyone advise if there has been a change in how pickups work? Here is an example where 1000101 is trying to pick up a call to 1000103: <SIP/product-local-00000005>AGI Rx << EXEC Dial "Local/1000103 at product-pickup

Here i am doing a dial command from a php agi... EXEC DIAL H323/123456789@xx.xx.xx.xx:1720|40|HL(585000:61000:30000) But asterisk is not disconnecting the connection after 585 secs... the result is ... answered time is 1926n but thing is time out is working some time and some time not.... LOG: 2005-06-28 20:26:13 VERBOSE[19094] logger.c: callcard.php: string(111) "app_callingcard:

Hi all, Recently, I have upgraded the asterisk as following. asterisk-1.4.13 asterisk-addon-1.4.4 libpri-1.4.1 zaptel-1.4.5.1 Usage of the server: inbound and outbound call, queue, mixmonitor, meetme, moh After upgrade, the server get segfault randomly and asterisk crash and restart itself. I got 2 core dumps of the segfault. Based on the core dump, we can't figure out the root cause to

Hello. I have set up ASTCC properly, calling it like this: DeadAGI(${ACCOUNTCODE},${EXTEN}) It seems to be working correctly, but I have two questions: - Although the cards' credit seems to be maintained correctly, I cannot see the call details in astcc-admin. When I try to view information on the card, it's just blank. Any ideas? - When does the 2nd, 3rd and 4th trunk get used? I have

Hi, I'm having problems using a TDM400P Card. I put my SIM card in a Nokia Premicell and connected it to a TDM400P card but when I make calls to the number, it hangs up automatically. The card also can't call out. Any ideas? I've searched the archives without much success. I appreciate all your help. Details: I'm using Asterisk 1.2.17 on Fedora Core release 5 (Bordeaux). On an

Hi All, I have a data frame: Time1 Time2 1 176:46:10 41:48:06 2 171:28:57 61:19:10 3 178:25:15 34:05:35 4 74:04:20 25:01:55 5 136:11:20 37:59:32 6 138:17:17 30:22:27 7 183:04:48 29:25:02 8 179:35:01 19:29:44 > str(df) 'data.frame': 8 obs. of 2 variables: $ Time1: Factor w/ 583 levels

Anyone has idea of what would be the best way to track connection time some a particular user to the internet ? Imagine a wifi network where the users will connect to the system via DHCP ( there is no PPPOE session involved ). If there is a need to track internet usage based on connection time ( to the internet ), what would be the best way to track it ? Appreciate any input or ideas.

Hi I am using Asterisk-1.6.1.6. Recently I had converted billsec and duration fields in my postgres 8.4 database from integer to numeric. I wanted to have better accuracy in calculating duration of the call. When the call is answered I can see better precision (for example 298.758421 sec), The problem is when the call is not answered (disposition is no answer) the duration is close to expected

In the last 10 days I've had 4 calls be charged by my provider at exactly 12 hours longer than the call actually lasted. I think this as all after upgrading from 1.6.17.1 to 1.6.17.2 but I have no way to be sure and it's so random. 2 calls 20 minutes apart last week and 2 more today many hours apart. Is anyone else seeing anything like this or have any idea where I might look. They

Is there a way to find out what the computation duration time was to complete executing a code chunk?

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Thank you very much for your help, it worked that way. Mathias ----- Urspr?ngliche Mail ---- Von: "xiphmont@xiph.org" <xiphmont@xiph.org> An: Conrad Parker <conrad@metadecks.org> CC: Mathias Kunter <mathiaskunter@yahoo.de>; vorbis-dev@xiph.org Gesendet: Donnerstag, den 20. M?rz 2008, 12:49:16 Uhr Betreff: Re: [Vorbis-dev] Determining the duration of an ogg vorbis

Dear R-listers, I am trying to compute simultaneous confidence intervals with simint from the package multcomp. 230 measures (abundance) have been taken in 23 sites (factor) of a data.frame (donnees: a file can be sent on request, saved with save(donnees,file="donnees")). I would like to get all pairwise comparisons with : mc<- simint(ren~ID,type="Tukey",data=donnees) I

Hello All, I am interested in estimating a duration model (also known as survival analysis or event-history analysis). I use an economic dataset. In economics terms, the model is "duration model with sample selection (or selectivity)." The time spell variable is only observed for a sample that meets certain requirements so the sample is nonrandom. Does anybody know any R package that

Hi. I am new to asterisk. I have an ivr application with asterisk and voiceglue. I make a call from asterisk (say to A) and when callee press a button voiceglue transfer the callee to another number (say to B). When I look cdr records the billsec between A and B always 0 and billsec with A shows the billsec for A and B. I am confused. Is there a reliable way to get the real call durations? Best

Hi all, I've got a question on regarding how to find out the duration of an ogg vorbis file. Sorry if it should already be mentioned somewhere in the docs, but I couldn't find out how to do it the best way. So, I can read the [bitrate_nominal] field from the identification header (if it is set), find out the size of the audio data (by subtracting the size of the three headers from the