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Showing posts from October, 2012

Heartbeat dan DRBD

Dalam sebuah implementasi saya harus mengganti implementasi vrrpd (virtual router redundancy protocol) dengan heartbeat+drbd disebabkan adanya penambahan database dalam server yang digunakan. Service awal pada mesin ini hanyalah web server statis, named dan dhcpd yang relatif statis dan file-filenya saya sinkronisasi dengan rsync. Tetapi dengan adanya penambahan database (mysql) dibutuhkan sebuah mekanisme dimana data yang disimpan dalam satu mesin primary dapat secara langsung ditulis juga ke mesin backup. Untuk hal yang terakhir ini vrrpd saja tidak mencukupi karenanya saya harus mengganti vrrpd dengan heartbeat (baca hartbit, bukan hertbet :-) )sedangkan untuk menjamin mekanisme clusternya saya menggunakan drbd.

Implementasi heartbeat saja sangatlah mudah. Cukup mendownload, mengkompilasi dan mengkonfigurasi tiga buah file /etc/ha.d/ha.cf, /etc/ha.d/authkeys dan /etc/ha.d/haresources. Untuk drbd bisa download tarball dan jangan lupa untuk membaca dokumentasinya, karena drbd harus d…

Asterisk 1.6.1 on openSUSE 11.1 (Part 5)

I will explain a bit more deeper about Asterisk configuration in this post, some trick and useful configuration that I found really helpful in configuring asterisk instalation. Asterisk developer really did a good job to make a complete PBX, they give the best tools to us and now it is our job to configure it.

One thing I found really annoying is the echo if we connect asterisk to PSTN line. I use digium TDM 410P and leave the card without tune it will give annoying echo. In my earlier post (Part 2) I explain that by running /usr/sbin/dahdi_genconf dahdi will automatically create /etc/dahdi/system.conf file that already contain information about hardware echo canceller. First thing you should remember if you have the budget is buy a card with hardware echo canceller. It will let the card to manage the echo without give the processor too much task to reduce it. After that you should tune the card. Luckily Digium give the best tools to tune the card named fxotune. To tune your card first…

Asterisk 1.6.1 on openSUSE 11.1 (Part 4)

medwinz note:

I got a bunch of email from Indonesian gentle readers about this topic, so I decide to write in Bahasa Indonesia for the Part 4. But don't worry google translate is there. Happy reading :-)

Pertama-tama terima kasih atas antusiasme rekan-rekan yang sudah nge-japri dan memberi komentar. Saya mohon maaf karena bagian ke-4 ini agak telat, namanya kuli harus tour of duty dan ngejar setoran :-)
Pada part 3 saya telah memberikan contoh extensions.conf, saya perlu menyertakan beberapa contoh file konfigurasi lain yang dibutuhkan agar penjelasan extension.conf bisa dimengerti. File-file tersebut adalah:

/etc/asterisk/chan_dahdi.conf
/etc/asterisk/sip.conf
/etc/asterisk/iax.conf
/etc/asterisk/meetme.conf
/etc/asterisk/voicemail.conf
Contoh chan_dahdi.conf:

;                                                                                                              
; dahdi_channels.conf configuration of digium card                                                             

Asterisk 1.6.1 on openSUSE 11.1 (Part 3)

To enable asterisk to communicate with PSTN lines we should have either a VOIP-PSTN gateway or FXO card. I will not explain about VOIP-PSTN gateway, there are some service providers out there who provides this service for their customers. In my work I use Digium TDM 410P with 4 FXO port per card. There are some alternatives in the market like Sangoma, Rhino, etc, the important is we should make sure that it works with Asterisk either with dahdi driver or zaptel/zapata driver. Also if possible select the card that already has hardware echo-canceler. Echo is a problem in voip communication, and if you have card with no echo-canceler than your server CPU will busy do the job.

Just remember that Digium cards are no longer use zapata driver, and some changes has been made to the configuration file name and location, /etc/zaptel.conf become /etc/dahdi/system.conf and /etc/asterisk/zapata.conf become /etc/asterisk/chan_dahdi.conf

In the client site you can use any SIP client hardwares or soft…