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Telephony logging.

Offline arne

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Telephony logging.
« on: October 30, 2007, 11:28:13 PM »
Hello !

I used to have 3 boxes, the linux firewall gateway, the sme 7.2 server and the Astlinux asterisk server.

I then desided to build it all into one box, using the sme server for it all.

I downloaded and installed one of the versions of asterisk for sme server rpm's and due to some nostalgy I preferd to keep my old manually edited configuration files for the Asterisk server (I guess I do not use Sail, but the asterisk module made for the sme server.)

Everything works just perfect now, sound quality it exelent and everything is good, exept one detail...

There is no logging of telephony traffic. I have looked into the standard log files. There is an amount of technincal information, but not this small little info: Who did call me when I was away. At least I am not able to find it.

If anybody should now som parameter setteing or some command to make it just do a log entry when somebody calls me, I would be greatly thankfull.

Arne.
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Offline SARK devs

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Re: Telephony logging.
« Reply #1 on: October 31, 2007, 12:31:36 PM »
Hello Arne,

I'm not sure I fully understand what you are asking.  However, you can set many different logging options in /etc/asterisk/logger.conf.

Usually an inbound call which isn't answered will/can be routed to voicemail.  Do you use this at all?

Any inbound call will usually be recorded as a CDR (Call Detail Record) in /var/log/asterisk/cdr-csv/Master.csv

Hope this helps

Kind Regards

S



Offline arne

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Re: Telephony logging.
« Reply #2 on: October 31, 2007, 04:29:01 PM »
Thanks a lot and shame on me  8-)

Of cource it was there: /var/log/asterisk/cdr-csv/Master.csv (I did look into this directory but for uknown reasons I did not open this file.)

Voicemail .. I used it before, but then I got the idea that I would rather practical use the external voicemail of Skype. So then if I'm not there the call it sendt to Skype and then to Skype voicemail, that I can receive and pick up from anywhere. It works quite allright exept for one weakness: When the call is resendt to Skype voicemail, I can only see my own telephone number as ID, so I have to play the msg before I know who was calling. (And thats one of the reasons that I like to know where to find the log file on the server.)

One other thing - I am using the Zoiper telehony client on the laptop configured to use IAX on UDP port 53. This really works well and usually logs into the server in a fraction of a second when Windows boots up. Using UDP port 53 makes it work witout a problem trough most outbound firewalls. (IAX port 53 and SIP with all ports can not be compared at all when it comes to "connectivity on different locations"). While Skype will use some seconds to come trough an outbound firewall, and a SIP client will not do it at all, Zoiper with IAX2 and UDP port 53 pass trough in a fraction of a second.

Arne.
« Last Edit: October 31, 2007, 04:32:23 PM by arne »
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