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Sat 05 of Jul, 2008 [19:20 UTC]

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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.
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IPDialog Phones

SIP phone:
http://www.ipdialog.com/products.htm

According to IPDialog's tech support (as at 2005-04-05) the current model SipTone II phone supports POE using PowerDsine's capacitive detection method, not 802.3af. The SipTone II phones may be powered from a PowerDsine 60xx midspan power injector. The next version of the SipTone (v III) (due out in June 2005) will support 802.3af.

Read reviews:
Network Computing Review
VoIP User Review

And some configuration info for the SiptoneII:

in sip.conf:
sipguest
type=friend ; Friends place calls and receive calls
context=internal ; Context for incoming calls from this user
canreinvite=no ; works fine without, but need this for asterisk intervention (i.e., hitting # to xfer)
secret=xxx
host=dynamic
dtmfmode=rfc2833
username=sipguest ; Username to use in INVITE until peer registers
defaultip=192.168.0.153
disallow=all
allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
mailbox=43

This is probably standard, but the username= should match what's in the []'s.


On the siptone (via web interface), you want to set:
userid to whatever you have in []'s and username= in sip.conf (here it's sipguest)
voicemail server to your asterisk box (so the msg light works)
dial plan to whatever you need, mine's:
  • *|4x|7xx|9011x.T|9911|3xx|88|91xxxxxxxxxx|923456789xxxxxx

On servers, you want to set:
SIP PROXY to sip:192.168.0.150 (where 192.168.0.150 is your asterisk box)
and check both boxes (Forward all through proxy & Register through proxy)

Set your server password to your secret= in sip.conf

Under Advanced, I checked "disable * codes" since I want asterisk to get the *'s, not the phone.

I think there's a way to use asterisk as an autodial host, but I haven't gotten that working. We just dial into the extensions listed above and it works fine.

To make the distinctive ring work. You have to send <Bellcore-drN> (where N is 1,2,3,4 or 5 in the alert info). Mine looks like (in extensions.conf):

exten => 41,1,SetVar(Alert_Info=<Bellcore-dr4>)
exten => 41,2,Dial(${GUEST},10)


For Asterisk 1.4 or greater, you'll need to use the SipAddHeader() application, e.g.:
exten => 41,1,SipAddHeader(Alert-Info: <Bellcore-dr2>)
exten => 41,n,Dial(${GUEST},10)

Note that there's no need to escape the greater than/less than signs.

There is "no way" to change the actual ring tone, per support.


Created by JustRumours, Last modification by Nick Barnes on Thu 22 of May, 2008 [12:35 UTC]

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