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  • Samuel, Thu 03 of Jul, 2008 [13:41 UTC]: ok thank you
  • Mats Karlsson, Thu 03 of Jul, 2008 [13:37 UTC]: Nice Samuel, will look forward to rad it.
  • bwl_fernstudent, Thu 03 of Jul, 2008 [09:08 UTC]: Your blog shows some usefull code
  • Samuel, Thu 03 of Jul, 2008 [08:04 UTC]: I'll translate it, for sure
  • Mats Karlsson, Wed 02 of Jul, 2008 [20:46 UTC]: LOL, in french! Translate it to English and I will read it.
  • Samuel, Wed 02 of Jul, 2008 [08:07 UTC]: Hello, i wrote a blog about Asterisk, speaking about installation,programming and more http://sambranche.blogspot.com/
  • Nick Barnes, Tue 01 of Jul, 2008 [17:46 UTC]: Steve - Asterisk doesn't 'fit into linux' - it's an application which runs on top of Linux.
  • Steve, Mon 30 of Jun, 2008 [18:07 UTC]: anyone know where I can find a block diagram of how asterisk fits into linux. my f'ing bosses want me to draw something up.. ugh.
  • akbar, Fri 27 of Jun, 2008 [10:37 UTC]: marley_boyz@yahoo.com how to configure call forward, call back, call pick up using TDM and asterisk 1.2.13... please help me.. thx...
  • Matthew Williams, Tue 24 of Jun, 2008 [22:37 UTC]: We are looking for Tier II VoIP Support Technicians in St Louis. Send resumes to mwilliams AT voxitas DOT com.
Server Stats
  • Execution time: 0.39s
  • Memory usage: 2.58MB
  • Database queries: 33
  • GZIP: Disabled
  • Server load: 0.61

Asterisk setup success 4

Hardware Configurations


Machine name pbx:

  • Dell PowerEdge 300SC
    • Dual 1Ghz slot-1 PIIIs
    • 1GB memory
    • 2 160gig IDE HD (SATA) connected to 3ware RAID card, doing RAID-1 (mirroring)
    • Digium T100P

Other hardware

  • Sipura 2000 POTS gateway (connected to our Polycom conference room phone and cordless 900Mhz analog phone)
  • D-link D104S MGCP POTS gateway (not used but configured)
  • WiFi (802.11) SIP Phone
  • Rocksteady NSA solution (http://www.rocksteady.com) for VoIP prioritization/queuing

Description of deployment.

  • PRI from SBC (Southwestern Bell) into our facility (migrated entire company from Lucent/Avaya Partner PBX which we outgrew)
  • PRI has 16B channels and 1D channel (room for growth to 23B)
  • 100 DID numbers from SBC
  • 35 Cisco 79XX phones (mixture of 7960s and 7940s) with latest 6.1 SIP firmware (connected to two Cisco Catalyst 3524PWR for in-line power)
  • 3 Cisco 7940 phones with latest 6.1 SIP firmware at remote locations over nailed up IPSEC VPN tunnels
  • 2 Remote Sipura 2000 POTS gateways connected over nailed up IPSEC VPN to users
  • 5 Grandstream Budgetone 100 phones
  • 5 Cisco 7905G phones, remote via VPN tunnels for Support personnel
  • Phones are on a separate Layer 3 VLAN internally for voice traffic, and tagged with a TOS bit of 5 for prioritization
  • NuFone for connectivity and dialing and additional 866 number (dial 8 to get NuFone, 9 to get SBC)
  • System configuration /etc/asterisk and /var/spool/asterisk (voicemail, etc) gets backed up to tape library daily.

Special features in Extensions ruleset with Asterisk

  • All phones have DID extensions
  • All Cisco phones have additional Intercom (autoanswer extension)
  • ACD used for incoming calls to our main number, queue is only called during "working hours" otherwise defaults to IVR
  • Several DIDs and extensions ring multiple extensions including home office connections over VPN (some ring up to 4 phones at one time)
  • ACD queue used for Support personnel, "static" agents always ring certain SIP extensions, but other agents can login/logout remotely to be added to the virtual/extended support personnel.
  • IVR menu for main number
  • Call Forward on no answer
  • Call Forward unconditionally
  • Speed Dials
  • Remote Call Forwarding
  • Ability to record calls
  • Conference Bridge (room)
  • MP3 Music on Hold
  • Voicemail (of course) w/e-mail and paging notification
  • Directory
  • 877 number for toll-free access
  • CallerID lookup in external Database for Name (if not supplied)

Overall experience


We've had wonderful feedback on everything from system installation and operation. Migration was smooth and SBC worked well with us to port our existing numbers to DIDs. I had extensive experience with Asterisk for almost 1 year prior to doing this for the company. System is monitored via Nagios.

Initially wanted to use RxFax for incoming faxes, but incompatibility with many fax machines forced me to use Hylafax (www.hylafax.org) and keep an SBC analog line for the purpose.

Lenny Tropiano
voip-wiki@voiping.com
http://www.voiping.com

Created by InetNomad, Last modification by Paul Gillman on Mon 25 of Jun, 2007 [23:20 UTC]

Comments

Comments Filter

SpeedVoIP Debuts VoIP Anti-blocking in Dubai

by jenniferhan on Wednesday 17 of October, 2007 [02:51:26 UTC]
As VoIP business users in Dubai are being blocked. Many users are turning to VPN solutions to allow the ability to use VoIP and get around the current blocking issue. This however is an expensive and unnecessary solution with SpeedVoIP Technology. To resolve this situation, SpeedVoIP has released it's new solution for Voip Blocking called VGCP (VoiceGuard Control Protocol).

In today’s market, VoIP for business has become more and more popular and necessary than ever before.

Dubai has become a big market, many big companies need to open branch offices in the UAE allowing more profit and larger market access. Technology Issues become apparent during this process that can cripple communications for that company. The primary communications issues are with VOIP blocking policies implemented in Dubai.

Now, here is the good news, A Canada based company SpeedVoIP with their integral R&D team have work out a new way to solve this VoIP blocking issue. This new system VGCP (VoiceGuard@ Control Protocol) has now laid the path to streamline low cost telephony solutions removing country limitations.

VGCP is a proprietary layer 2 link protocol working at between IP stack and NIC driver for VoIP anti-blocking. The core patent-pending VGCP is industry's most state-of-art voice service provider class security protocol whose scalability and flexibility results in not to compromise voice quality and overhead. VGCP controls and monitors full voice signalling and media flow intelligently, meanwhile disguises sip and RTP packets into normal allowed data packets such as DNS and TFTP, and makes two-way encryption and decryption driven by user-customized policy. VGCP is fully transparent to upper SIP proxy or UA which means VoiceGuard@ can work with any 3rd party soft phone/ATA/Gateway/IP Phone/IADs and SIP Proxy or Server not like some competitors which take effect on their own device and soft switch.

Korea Telecom has implemented this solution successfully for more than one year. And it has been operational within a group of Dubai companies. The trials and implementations proves that, The VGCP solution is the best solution to solve the VoIP Blocking issue and provides stable communications platforms providing an indispensable part of the business network.

Andy Wong ~ ~
MSN: andywong-01@hotmail.com
Email: xd.wong@speed-voip.com
www.speed-voip.com

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