vicidial.org

Hi Guys,

Could you pls help me to figure out my problem. I have muliple server with loadbalancing. (Scratch installed)

2 Asterisk/Vicidial (Slackware12.1 Linux 2.6.24.5, astguiclient_2.0.4.1rc3, asterisk v. 1.2.27)
1 Mysql/Apache (mysql 5.0.45)

Here is my crontab:
0,3,6,9,12,15,18,21,24,27,30,33,36,39,42,45,48,51,54,57 * * * * /usr/share/astguiclient/AST_CRON_audio_1_move_mix.pl
1,4,7,10,13,16,19,22,25,28,31,34,37,40,43,46,49,52,55,58 * * * * /usr/share/astguiclient/AST_CRON_audio_2_compress.pl --GSM

My extesion recording is 8310 (gsm). Looking /var/spool/asterisk/monitorDONE, no recording at all. I'm doing the recording thru hardisk, later i will do it thru system RAM and convert it to MP3 but for now i would like to confirm if my recording is ok or not.. I'm just testing it using one agent manual call. Later will test predictive dialling.

If i could finish my R&D with this system asap, we are planning to sell the service and difinitely 5% of our income per project will be put to eflo.net.

I hope somebody will help me with my problem.

regards,

P.S. Matt, how much your remote support per hour?

In addition, campaign and agents are configured ALLFORCE recording.

I would suggest recording to WAV format(8309) and then compressing after the call is done.

We have not tested converting from GSM to MP3, but the audio quality will suffer if you do that.

Our standard rates for consulting start at $200/hour and we offer various discounts depending on the number of hours and level of support package if you choose a support plan.

I thank you very much for your offer fo supporting the project, it is very much appreciated.

thanks matt for your prompt reply.

i took your advice, i put it back to wav(8309) and changed the cron scripts into:

0,3,6,9,12,15,18,21,24,27,30,33,36,39,42,45,48,51,54,57 * * * * /usr/share/astguiclient/AST_CRON_audio_1_move_mix.pl
1,4,7,10,13,16,19,22,25,28,31,34,37,40,43,46,49,52,55,58 * * * * /usr/share/astguiclient/AST_CRON_audio_2_compress.pl --MP3

But unfortunately, still no recording files generated in the /var/spool/asterisk/monitorDONE/ORIG and /MP3.

Looking at agc/vicidial.php, it shows the RECORD ID # and the recording file. I check the link below "show conference call call information" while im talking to end party and it shows only 2 channel. I guess, it should be 3 channels, right?

Here is my asterisk cli for your perusal:

-- Executing MeetMe("Local/8600100@default-379f,2", "8600100") in new stack
> Channel Local/8600100@default-379f,1 was answered.
-- Executing AGI("Local/8600100@default-379f,1", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("Local/8600100@default-379f,1", "SIP/64.192.251.11/19148461321||tTo") in new stack
-- Called 64.192.251.11/19148461321
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
Aug 13 15:41:07 NOTICE[30981]: chan_local.c:526 local_alloc: No such extension/context 58600100@default creating local channel
Aug 13 15:41:07 NOTICE[30981]: channel.c:2514 __ast_request_and_dial: Unable to request channel Local/58600100@default
-- SIP/64.192.251.11-082315c0 is ringing
-- SIP/64.192.251.11-082315c0 answered Local/8600100@default-379f,1
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Spawn extension (default, 919148461321, 2) exited non-zero on 'Local/8600100@default-379f,1'
-- Executing DeadAGI("Local/8600100@default-379f,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----26-----24)") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---NODEBUG

Here is my version from VICIDIAL web-client version: 2.0.5-163 BUILD: 80719-1147

From admin.php == VERSION: 2.0.5-136 BUILD: 80719-1351

Additionally, i run the scripts manually without any error:

root@viciserver1:/usr/share/astguiclient# perl AST_CRON_audio_1_move_mix.pl

root@viciserver1:/usr/share/astguiclient# perl AST_CRON_audio_2_compress.pl --MP3

However, if i dialed the extension 8309 directly from softphone..then files were generated.

thanks again in advance.

looks like you are using SVN trunk. You need to read the UPGRADE doc to see that a new extension has been added for recording that you need to put in your dialplan.

thanks matt..

just to clarify, currently i'm using SVN trunk and not the astguiclient_2.0.4.1rc3? If yes, then can I install astguiclient_2.0.4.1rc3 instead of upgrading SVN trunk? Then the extension for recording will be included already?

In astguiclient_2.0.4.1rc3, some experienced a lot of pop up when they made oubound call, is that resolve already?

Hello Matt,

just to clarrify very important thing, according to SVN trunk UPGRADE doc, the latest update is 2.0.5. As mentioned in my third posted message, im using 2.0.5 already meaning i don't have to upgrade it, right?

I just added the exten => _58600XXX,1,Meetme,${EXTEN:1}|Fmq, and now gotchha the recording is working already, also the generated file was their in the corresponding path.

Here is my clarrification and hope u can ans me, after adding that extension, do i have to follow some procedures in the UPGRADE doc or no need since i'm using 2.0.5 already? like run this command no. 3\. /path/from/root/extras/upgrade_2.0.5.sql, and so on...

I hope you can enlighten me for the same.

When did you checkout the code snapshot from SVN trunk?

What is the db schema version as shown in the Admin-> System Settings page?

What is the last db_schema_version as shown in the upgrade_2.0.5.sql file that you have on that machine?

When did you checkout the code snapshot from SVN trunk?

----- 2008-07-24

What is the db schema version as shown in the Admin-> System Settings page?

--- version. 1101

What is the last db_schema_version as shown in the upgrade_2.0.5.sql file that you have on that machine?

--- sorry where can i find the info?

You should be fine. you are using the same version of the database schema that you need.