similar to: agi debug in Asterisk 1.6?

Hi all, This issue is giving me a lot of grief with my customers. I have 5 asterisk servers running in production, each one with almost 70 simultaneous calls at peak hour. Most of my customers complain that their calls drop after 20 seconds or so. After running through my cdr's, I see that the number of 20 second calls is MUCH larger than any other number. (see below) billsec count(*) 1 924

I'm trying to get my asterisk server to reinvite. I have two asterisk servers with public IP's. My users (behind NAT) register on one server (I'll call it server 1), and for some calls they are transfered over to the other server (server 2), because that server has the E1's. I want server 1 to be in the signaling path for billing reasons, but handling the media stream is killing

Hi all, I've been trying to add a custom mysql field to my CDR's, but I must be doing something wrong. I am using asterisk 1.4 and asterisk 1.6, in extensions.conf I add: exten => h,1,Set(CDR(q931)=${HANGUPCAUSE}) This extension is executed, I can see it in the asterisk console. I have added a new column in my MySQL database called q931. However, the new field does not show up in

Hi all, I am having a curious problem. I use two cdr backends, csv and MySQL. These are my settings: Call Detail Record (CDR) settings ---------------------------------- Logging: Enabled Mode: Batch Log unanswered calls: Yes * Batch Mode Settings ------------------- Safe shutdown: Enabled Threading model:

Hello everyone. I'm using Asterisk 1.4.31 and A2Billing 1.7.0 to manage a small wholesale operation, so I configured A2Billing for not to answer the call nor play any greetings or balance notifications to the caller. I'm authenticating each customer by it's IP address, and each customer has it's own context, in which I set the following: ;=====in extensions.conf======

Hello all, I have a pretty much standard installation of an Asterisk 1.6.1.6 with no PRI cards of any type (full VoIP). Occasionally (it happens every 2 weeks or so), it just stops running. I was using safe_asterisk but it seems that safe_asterisk did not restart it. I do have the core dump file at /tmp/core.myservername-2009-10-20T18:36:20+0200 but it seems it's from an earlier crash. When

Hi I am currently using 1.2.x and 1.4.x behind OpenSER. One of the things I do on INVITES is to re-authenticate the user from OpenSER. Then when the INVITE gets passed to Asterisk I capture the AUTH to a variable in the dialplan and pass to an AGI script. I am now trying to set the same thing up in 1.6 However because the argument delimter in 1.6 has changed from pipe to comma this breaks as the

Hello everyone. I'm using Asterisk 1.4.31 and A2Billing 1.7.0 to manage a small wholesale operation, so I configured A2Billing for not to answer the call nor play any greetings or balance notifications to the caller. I'm authenticating each customer by it's IP address, and each customer has it's own context, in which I set the following: ;=====in extensions.conf======

Dear, I have asterisk 1.6.2.13 in a small hardware PBX (512 GB RAM and Celeron), and last days when I call from one extension to another of the same PBX after I dial the number the rings sound after 20 seconds. In the CLI log, when I debug the AGI, I see always goes good until dialparties.agi, and after that there are 20 seconds without any log, and so the ring sound. I've read

Hello! In my agi-debug i get the following error-message: AGI Rx << Jan 16 14:45:52 WARNING[18299]: asterisk.c:788 ast_set_priority: Unable to set normal priority AGI Tx >> 510 Invalid or unknown command AGI Rx << SET VARIABLE MODCLI 00434345452 the agi i call is a very simple shellscript that simply removes wrong charakters: #!/bin/bash modcli=`echo $1 | sed -e

I'm wondering if anyone knows a good, stable C AGI library (* v. 1.4 and 1.6 compatible). I've taken a look at CAGI and QUIVR but their latest code releases date back to 2006. I've also seen a more recent project (wildpbx) dated 2009: http://github.com/comradeb14ck/wildpbx/tree/master/libraries/agi/c/ Any suggestions/recommendations for a C AGI library? Thanks, Vieri

Hi all, I am having trouble getting cdr_adaptive_odbc to work. I have correctly configured the odbc drivers and dsn (I have tested this by connecting directly using isql). I have also configured /etc/asterisk/cdr_adaptive_odbc.conf like so: [test-asterisk] connection=test-asterisk-odbc table=cdr I have tested the ODBC connection test-asterisk-odbc and it works correctly However when I try to

I am currently suffering various SIP attacks. I am using the following extension to record the caller's IP address: exten => h,n,set(CDR(srcip)=${CHANNEL(recvip)}) However, in recent attacks, this IP address is not correct, and I believe that they are spoofing it. I am using asterisk 1.6.2.15. Does the CHANNEL(recvip) variable record IP show in the SIP header instead of the real, UDP

Hello everybody, When I execute the "sip show peers" command in the asterisk console I always get the following notice, repeated 15 times after the sip show peers output. [Dec 21 03:38:31] NOTICE[12693]: utils.c:1074 ast_wait_for_output: Timed out trying to write This happens on a freshly installed 1.6.1.12 and a 1.6.1.6 box that I am running. Both of them use Debian Linux (lenny) on

Hello everybody. I have a provider that has 3 asterisk boxes which I must balance my calls against. At the moment, I route different destinations to different boxes but this causes lots of problems. Without resorting to OpenSER or other proxies (as my provider also uses IAX), is there a way I can load balance outgoing channels in Asterisk? For example an IAX peer like: [iax_provider] type=peer

Hello list members. I would like to know how to playback an audio file to the caller, and while it's played asterisk to continue executing the next priorities on extensions.conf That's not the case when using "playback" command, because the next priority is executed until the audio file ends playing. I want to evaluate some variables while caller hears the audio file. Any

Hello list, what is the correct syntax ? exten => s,n,Queue(${queuename},,,,${timeout},cleanpickup.agi^${CHANNEL}) [Sep 13 10:23:58] WARNING[23551]: res_agi.c:886 launch_script: Failed to execute '/var/lib/asterisk/agi-bin/cleanpickup.agi^SIP/329909007906-0000017a': File does not exist. Kind regards, Jonas. -------------- next part -------------- An HTML attachment was

Hi, I am developing asterisk support for our application using the Async AGI and Asterisk-Java. One thing I haven't been able to implement is how to stop playing a sound. Something similar to StopIO for Dialogic GlobalCall or DivaStopSending for Eicon. Is there any way to achieve this today which I have missed? Or could someone give me hints on how I could implement this in the res_agi.c The

hello i am trying to develop perl application for asterisk with radius accounting how can i debug that weather callback is working when call is stoped. how can i check this syslog('info', 'hello Asterisk!'); thanks Kamran ____________________________________________________ Sell on Yahoo! Auctions ? no fees. Bid on great items. http://auctions.yahoo.com/

Dear Sir, Can someone help me please to find a free ebook talking about AGI scripting through asterisk? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090218/a59fc299/attachment.htm