IPDialog Phones

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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|9[23456789]xxxxxx


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.


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|9[23456789]xxxxxx


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 &amp; Register through proxy)

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

Under Advanced, I checked &quot;disable * codes&quot; 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 &lt;Bellcore-drN&gt; (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=&lt;Bellcore-dr4&gt;)
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: Thu 04 of Nov, 2010 (04:20 UTC) by admin
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