> > 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. > > JTOk, 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? Ray Burkholder 519 570 0689 x2002
On Mon, 25 Aug 2003 18:45:22 -0400 "Ray Burkholder" <ray@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@jazz-inc.net Web: www.jazz-inc.net Phone: +61-414-365-466 Jabber: jazz@netmindz.net
I'll start by mentioning that the newer Cisco SIP dumps let you hit "#" instead of "Dial" when you're done dialing, which I find to be much more intuitive than the "Dial" softbutton.> 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.Even if Asterisk does the overlap stuff defined in RFC 3578, I seriously doubt you'll see the Cisco phones (or any hardware phones, for that matter) doing it. The overlap stuff is really designed for gateways from the PSTN, not end terminals. That said, there's nothing that would *prevent* implementing it in end devices. I note that it would cause an awful lot of signaling traffic if you did so, though. As a side note, I'll point out that the Pingtel phones let you provision client-side digitmaps. Based on asterisk-like pattern matching, you get to say how long a digit string should be matched, and the phone will automatically dial when it matches (no need to hit send!). You can even make different patterns go different places, like: 972xxxxxxx : sip:{digits}@localprovider.net 214xxxxxxx : sip:{digits}@localprovider.net 489xxxxxxx : sip:{digits}@localprovider.net 1xxxxxxxxxx : sip:{digits}@longdistanceprovder.net (To clarify: Dallas has three local area codes and 10 digit local dialing) /a
> > I'll start by mentioning that the newer Cisco SIP dumps > let you hit "#" instead of "Dial" when you're done dialing, > which I find to be much more intuitive than the "Dial" > softbutton. >Ah, cool. Further info about this further down.> > Good question.. Does * support overlap dialing with SIP?In the url referenced further down, the dialplan.xml file allows you to do pseudo overlap dialing, as in the phone will fake the downstream dial tones for you.> > That said, there's nothing that would *prevent* implementing > it in end devices. I note that it would cause an awful lot > of signaling traffic if you did so, though.Actually, this is what SCCP is all about. Every little thing you do on a SCCP phone is forwarded to CallManager for processing. Every button, every numeric key stroke. So if it is a good thing for Cisco and their installations with thousands of phones, it can't be too detrimental if we needed to do it.> > As a side note, I'll point out that the Pingtel phones let > you provision client-side digitmaps. Based on asterisk-like > pattern matching, you get to say how long a digit string > should be matched, and the phone will automatically dial > when it matches (no need to hit send!). You can even make > different patterns go different places, ...: >Ok, after further reading, the Cisco SIP can do the same thing with a dialplan.xml file in your Cisco phone tftpd directory. I took their docs and added a one line refinement for our four digit extensions: <DIALTEMPLATE> <TEMPLATE MATCH="0" Timeout="1" "User=phone"/> <!-- Local operator --> <TEMPLATE MATCH="9,011*" Timeout="6" "User=phone"/> <!-- International calls --> <TEMPLATE MATCH="9,0" Timeout="2" User="Phone"/> <!-- PSTN Operator--> <TEMPLATE MATCH="9,11" Timeout="0" User="Phone" Rewrite="9911"/> <!-- Emergency --> <TEMPLATE MATCH="w!" Timeout="1" User="PHONE" Rewrite="9911"/> <!-- 911 when entered in Alpha mode --> <TEMPLATE MATCH="9,.11" Timeout="0" User="Phone"/> <!-- Service numbers --> <TEMPLATE MATCH="9,101..............." Timeout="0" User="Phone"/> <!-- Long Distance Service --> <TEMPLATE MATCH="9,10.............." Timeout="0" User="Phone"/> <!-- Long Distance Service --> <TEMPLATE MATCH="9,10*" Timeout="6" User="Phone"/> <!-- Long Distance Service --> <TEMPLATE MATCH="9,1.........." Timeout="0" User="Phone"/> <!-- Long Distance --> <TEMPLATE MATCH="9,......." Timeout="0" User="Phone"/> <!-- Local numbers --> <TEMPLATE MATCH="2..." Timeout="0" User="Phone"/> <!-- Local Extensions --> <TEMPLATE MATCH="*" Timeout="8"/> <!-- Anything else --> </DIALTEMPLATE> Further information, including how to simulate overlapped dialing can be found at: http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/i pp7960/addprot/sip/admin/ver5_1/sipins44.htm#1047995 Ray.
> As to prior comments about SCCP documentation: if you'd like to help > contribute to the SCCP channel project, it seems far from 'aborted' > at the moment. Check out http://sourceforge.net/projects/sccp/ and > download the channel. Compile, test, send bugs, submit code. >The web site indicates that "This Project Has Not Released Any Files". Am I not seeing something?
> > As to prior comments about SCCP documentation: if you'd > like to help > > contribute to the SCCP channel project, it seems far from 'aborted' > > at the moment. Check out > http://sourceforge.net/projects/sccp/ and > > download the > channel. Compile, test, send bugs, submit code. > > > > The web site indicates that "This Project Has Not Released Any Files". > Am I not seeing something?Yep, I was missing something. Didn't look at the CVS tree. So code and SCCP phones are being used in production? Are there any caveats? I'll give this a try. Sooooo, what this means then, based upon all the other messages I sent today, if * does work with SCCP 7960's and 7940's, the intercom stuff should be a good addition for those phones, as they handle that streaming channel 1???!!! Ray.
Checkout the "dialplan.xml" file... Jared Smith On Mon, 2003-08-25 at 16:45, Ray Burkholder wrote:> > > > 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? > > Ray Burkholder > 519 570 0689 x2002 > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users