similar to: Manager API Originate CDR Problem, all is NO ANSWER

Hi, I've noticed that one of my new servers (new mobo) if drifting slowly backwards in time (in aprox. 24 hours, system time drifts back 5 minutes). I have an ntpd process which is supposed to sync with a lan time server but it's not quite working. So I'm launching a manual ntpdate or ntp-client once an hour and that seems to work. However, suppose I update system time at every hour

Is there any program Asterisk users use to calculate ASR and ACD ?? Thanks for any comments. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090910/af1f9656/attachment.htm

Hi all, I have a question: how can I see hints of a remote Asterisk in IAX2 trunk?? I want to set BLF on my phones to look state of other phones also in other Asterisk server. Someone have any idea or solution? I use Asterisk 1.4.24. Thanks all Marco -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have Asterisk 1.4.24 en a Digium TDM410 with EC and with 3 FXO modules. When I plug one PSTN-line into a FXO-port I am able to receive calls on this line and I can also make calls from an internal SIP-phone to the external PSTN-network. Still I am bothered about something that appears on the CLI when I do a reload chan_dahdi.so : asterisk*CLI> reload chan_dahdi.so -- Reloading module

hi,all i have just set up asterisk 1.6.0.7 rc1 with spandsp 0.0.5 pre4 to ReceiveFAX, link to a E1 (DE410P) using dahdi this can receive the fax from E1 successfully, but i see many error message in the log like this: [Mar 16 09:24:38] ERROR[23540] channel.c: ast_read() called with no recorded file descriptor. when i receive a 5 pages fax, i will see this error message over 200 lines..... it

Hi All, I have been running a environment with asterisk 1.4.20.1 for some time now with no issue but have recently added some extra functionality (enabled call recording via MixMonitor) and ran into some deadlock issues which seem to be well documented with earlier 1.4.x releases so have decided to take the plunge and upgrade. I decided to start testing with 1.6.2 but have run into a couple

Hi all, Just ran into some issue with the originate AMI command. It seems that there is a limit of around 120 calls I can place with the originate command simutanously. By that I mean sending Asterisk a lot of originate command very fast. Anyone know if there is a limitation? Thnx. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I am migrating a PABX system based in Asterisk 1.4 to Asterisk 13, with success. I have an application that sends an action Originate to AMI for calling, it's working well, but when i see to Asterisk's CLI, i see 2 calls for just one originate: pftestes40copiabh*CLI> core show channels verbose Channel Context Extension Prio State Application

Has anyone else had issues with Originate returning the wrong error code? According to the docs, the following errors are supposed to be returned: 0 = no such extension or number 1 = no answer 4 = answered 8 = congested or not available Now in Asterisk 1.4.23 I get some error code 5's but since they're so few I tend not to worry. But what is concerning is the number of Error 0's I

I'm using the Originate Action on the Asterisk Manager to place calls between two extensions in async mode. Is there any way to cancel the Originate Action before I get the OriginateResponse action? I'm unable to perform a Hangup because I can't know the channel name before I get the response... thanks in advance! -- santiago aguiar *netlabs* / Palmar 2548 Montevideo, Uruguay Tel.

Hello. I've been trying to use the AMI to originate phone calls. I'm trying to call the SIP phone 'zoiper' with the SIP phone 'yziquel'. So, the AMI interaction is: > Action: originate > Channel: SIP/zoiper > Exten: yziquel > Priority: 1 > Timeout: 30 > Context: internal > > Response: Error > Message: Originate failed > > Event:

I have written an asterisk manager client which creates an outbound call using Asterisk manager API's Originate action. when the call is connected I run 3 applications on it. 1)read a dtmf digit from user 2)A customized application which I have written,(It plays something to user) 3)Hangup If user hangs up while app 2(see above) is executing I get a 'Event Hangup' from asterisk in my

Hi, In current (6140c3177a0330f42411618c3fca28930ea02a21) samba's ctdb/tools/statd_callout_helper I find this comment: notify) ... # we need these settings to make sure that no tcp connections survive # across a very fast failover/failback #echo 10 > /proc/sys/net/ipv4/tcp_fin_timeout #echo 0 > /proc/sys/net/ipv4/tcp_max_tw_buckets #echo 0 >

Hello, I'm using an AMI Originate command to send a fax. The fax is sent by a script, and I'd like my script to send the fax, wait until it has succeeded or failed, then exit with an appropriate error code (it is driven by a mail system, so the exit code will tell the mail system whether to retry the fax later). The script works great if the fax succeeds, or if the line is busy or

Hi all, I have a strange situation on a Asterisk 1.2.17 with FreePBX 2.2.1 Doing a select in the CDR table I noticed there are some calls with billsec greater than duration, duration is always 0 in those calls. How can this happens ? Am I missing something ? Tnx in advance Regards Edoardo Serra WeBRainstorm S.r.l.

Hi, I'm stuck on a silly thing. I need to get the "billsec" CDR value after a call. But I'm finding its always 0. Here's my test code: exten => *244*,1,Dial(Local/test@custom-tests/n,,g) exten => *244*,n,Noop(after dial duration is ${CDR(duration)} billsec is ${CDR(billsec)}) exten => *244*,n,Hangup [custom-tests] exten => test,1,Answer exten =>

Hi, i am recording a message, but it is always cut off at 40 secs. There are no time out configured. Gabriel -- The educated person is not the person who can answer the questions but the person who can question the answer.

Hello, I want to create php rate script and I'm using Deadagi. But I allways get billsec 0 , or nothing. Can you help me to solve this problem... My extension.conf: exten => _123,1,DeadAgi(rate.php) exten => _123,2,hangup And my simple test php script rate.php #!/usr/local/bin/php -q <?php include_once (dirname(__FILE__)."/phpagi.php"); $AGI = new AGI();

This is more of mysql question then asterisk :D . Most voip providers use 6 second rounding for costing . My asterisk server stores call cdr's in mysql properly with billsec field containing number of billed seconds . I want to know some function to round this to 6 seconds ( or any custom valud like 30 seconds ) ..Suppose if billsec field is 3 seconds then it should round to 6 seconds , if

Hi All, I'm using the asterisk 1.4.39.2 with phpagi 2.20 I have setup a dial plan: [callback-outbound] exten => _00.,1,Macro(callout|${EXTEN}) [macro-callout] exten => s,1,AGI(getchannel.php|${ARG1}) exten => s,2,Dial(Local/${OUTBOUND}@from-internal/nj||tr) exten => s,3,Hangup() but for some reason i am not receiving the argument: Executing [s at macro-callout:2]