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  • Aykut, Wed 01 of Aug, 2007 [07:53 UTC]: Hi all, does anybody know about Thomson ST2030 SIP phone. I have upgraded it to latest version (1.56) but "Hold" and "Conf" features are not working after the upgrade ?? Do you know any solution or do you have Ver. 1.52 ?? Where can I find it?
  • Edward J Brown, Tue 31 of Jul, 2007 [23:33 UTC]: Has anybody experienced Choppy voice quality when using a Linksys SPA942 in an Asterisk Conference bridge? It works fine with my polycom and Cisco, but sucks with my Linksys.
  • www.astawerks.com, Fri 27 of Jul, 2007 [18:00 UTC]: does anyone use asterisk on top of clark connect? does it work good?
  • simon, Fri 27 of Jul, 2007 [14:16 UTC]: Hi All, Has anyone here managed to get the Cisco79x1 to successfully fail over to the backup proxy. I have 2 asterisk servers , handsets all register and function, except that backup proxy function doesn't work. Any working example would be very apprecia
  • Matthew Richmond, Thu 26 of Jul, 2007 [03:40 UTC]: using the page() application to page across our building...often the meetme conferences don't disconnect after the caller hangs up. Anyone else having this problem. (using Polycom phones)
  • Matthew Richmond, Wed 25 of Jul, 2007 [02:58 UTC]: thanks Nicholas Blasgen! I haven't worked with AGI before, but there's always a first! Thanks again!
  • Nicholas Blasgen, Tue 24 of Jul, 2007 [19:18 UTC]: Matthew Richmond, AGI will handle all that for you.
  • sam, Mon 23 of Jul, 2007 [16:39 UTC]: need help - certain voicemail extension will stop working and recording voicemail on asterisk - anyone know why and how to fix it? Thanks
  • john haji, Mon 23 of Jul, 2007 [14:55 UTC]: free calls to pakistan
  • bong, Sat 21 of Jul, 2007 [19:09 UTC]: hi good day to all can anyone help me how to configured the nortel sip to the signaling server and how to activate in mobile w/ sip compatible without mcs
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Voicetronix

VoiceTronix


VoiceTronix is a telephony equipment and solutions provider headquartered in Adelaide, Australia.

Their main products are a line of PCI based analog telephony interfaces:
  • OpenLine 4 - four port analog (FXO only)
  • OpenSwitch 6 - six port analog (3 banks of 2 ports configurable as FXO or FXS)
  • OpenSwitch 12 - twelve port analog (3 banks of 4 ports configurable as FXO or FXS)
  • OpenPCI 4 and 8 - four and eight port analog FXO and FXS.New Release Asterisk Compatible!

They have also recently released their OpenPri series of PCI based single and dual span E1/T1/J1 Primary Rate Telephony interfaces.


Voicetronix have their own open source PBX software, called OpenPBX. However, the company is also actively involved in the Asterisk project, providing open source drivers for their telephony interfaces which work with Asterisk. Voicetronix cards are configured in vpb.conf.

For more information on products, prices and where to buy please visit the VoiceTronix web site.



Changing Dialtone
If you want to change the default dialtone being heard (Australian) to the USA one, you'll need to comment the TONES_AU settings and uncomment the TONES_US settings. In the channels folder, search the chan_vpb.c file "/usr/src/asterisk/channels/chan_vpb.c"

?grep ?i ?A 10 tones_au chan_vpb.c?

/* #define TONES_AU */
  1. define TONES_USA

/* #ifdef TONES_AU
static VPB_TONE Dialtone
static VPB_TONE Busytone
static VPB_TONE Ringbacktone
  1. endif */

  1. ifdef TONES_USA
static VPB_TONE Dialtone
static VPB_TONE Busytone
static VPB_TONE Ringbacktone


I had to comment the tones_au, because the country i specified in the indications.conf doesn't give me the US dialtone.

If asterisk is already installed you'll have to recompile.

G. Powell

Changing Ringback

The ringback tone period for USA is 6s (2s on, 4s off), thus one also needs to change the following define in chan_vpb.c to 6000

 #define TIMER_PERIOD_RINGBACK 6000





Timing source
I've had problems using ztdummy for timing along with the voicetronix so, if you want to use the OpenSwitch card for timing which you'll probably want to do, edit the vpb Makefile, and uncomment ?AST_TIMING?.

My Makefile is located in ?/usr/src/vpb-driver-2.4.0/src/Makefile?, if you type:
?grep ?i ast_timing Makefile? you?ll see that it has been commented. So to use the timing feature of the card you need to uncomment it and then recompile Voicetronix drivers & Asterisk (in that order)

1. modprobe zaptel
2. modprobe vpbhp
3. lsmod

After the lsmod you'll see that the voicetronix card will use the zaptel drivers for timing. Now accessing Voicemail & IAX calls should be clear.

G. Powell

Voicetronix Resellers



Created by benjk, Last modification by Eric Chamberlain on Wed 23 of May, 2007 [23:09 UTC]

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