[Asterisk-Users] SIP vs SCCP vs XML

[Jamie Carl]

On Mon, 25 Aug 2003 18:45:22 -0400
  "Ray Burkholder" <ray at oneunified.net> wrote:
>*This message was transferred with a trial version of 
>CommuniGate(tm) Pro*
>
>> 
>> No, this is not the case currently with any of the Cisco 
>>SIP software 
>> loads that I am aware of.  If you find this to be 
>>incorrect, please 
>> let the list know.  Cisco has not deployed much of the 
>>featureset in 
>> their SCCP phones (such as paging/intercom) into the SIP 
>>phones due 
>> to lack of standards/interest/political capital.
>> 
>> JT
>
>
>Ok, after further research in the 7960 administrators 
>guide for SIP 5.1
>(current is 5.3 and probably not changed much), they do 
>state that
>support is not provided for CiscoIPPhoneExecute in the 
>current SIP load,
>which is needed to make streaming channel 1 work. 
> Bummer.
>
>So, in looking around at HotDispatch.com, I see a number 
>of companies
>charging outrageous dollars for their own SCCP versions 
>of a softphone.
>
>Also, a while back, for $1000, a person could join 
>Cisco's developer
>program and gain access to SCCP docs.  Perhaps an 
>Asterisk group member
>has the funds available to attempt joining?  Then we 
>could finish up on
>some of the aborted attempts at SCCP integration, if the 
>license
>agreement allows this sort of development.
>
>Perhaps, through a little creativity, it might be 
>possible to use a SCCP
>796x phone and not worry about SCCP.  With XML, screens 
>could be
>programmed to send responses back to *.  Then * could 
>drive streaming
>channel 1 directly and simulate the phone call.  So, on a 
>SCCP phone,
>you don't use SCCP, nor SIP.  You use XML.  Would that 
>work?  Hopefully
>soft button presses don't interfere with the streaming 
>media.
>
>Oh, and if it does work, then you can use multicasting to 
>intercom a
>number of phones simultaneously.
>
>The thing I miss on SIP phones that was available on the 
>Callmanager
>version of 796x, is the ability to go off hook, dial some 
>numbers, and
>callmanager automatically dials the call.  The SIP 
>version requires you
>to go off hook, dial the digits, then press dial.  Any 
>way around this
>for 4, 7, 10 or 11 digit dialling?
>


Good question..   Does * support overlap dialing with SIP?

I have a feeling it does, I do vaguely remember getting an 
Address Incomplete response when not dialing enough 
digits.  I guess all you have to do is set your cisco 
phone for overlap dialing.  Hopefully there is an option 
for it in is config. 

Regards,

Jamie Carl
Jazz Inc.
Email:  me at jazz-inc.net
Web:    www.jazz-inc.net
Phone:  +61-414-365-466
Jabber: jazz at netmindz.net