I have a SIP phone I use with my AllStarLink node. It’s a Polycom VVX-410 and has 12 line buttons, of which one dials my node.
I can use the keypad to control connections but I’d like to program the remaining 11 buttons for common dtmf sequences. But when I program a button as a speed-dial, it parks the current call to the node and makes a new call instead of just sending the digit sequences on the current call.
Any suggestions? Thanks in advance
Interesting question. How would you get it to put DTMF on a current call rather than bringup a new line? Anything in the Polycom manual?
Their documentation is quite good. Worst case is the call is parked and I establish a new call to the node, pause, and make the new connections. This document looks interesting but it will be a while before I can test it.
Thinking out loud here….
Terminating my SIP call won’t break any connections, so what’s the harm in just having five dial, pause, connect strings, each to a different node on each of five buttons? And five disconnect strings. And use the last button for dial and disconnect all.
The dial would always be to my node and the connect would be a *99xxxxxxx to my favorite nodes. It might work. Thoughts?
While I have no direct answer for you,
I know that Polycom do things a bit proprietary but, I think you need to look at their commands for programming the stored dial for those buttons, I don’t think you can just store the plain dtmf in there.
I do not have any issue with doing the same on Cisco/Linksys 8 line phones or 4 line phones.
. Perhaps a google search will reveal something for you…
But for me there are 2 methods. You enter the required info on the quick button via html to the phone ip or to do this manually from the buttons on the phone. I found it easier to initially use the phone button method to get some of the required info stored in the string and go back and manually alter it via html interface.
I use 4 to just dial into each of my 4 servers and monitor a node.
One dials a wake-up call/alarm.
I use a actual pre-dial string prefix (1,2,3,4,5,6 in place of international code adjusted in dial plan) to select the outgoing line I use as I have 6 numbers, not using the quick buttons for that.
That’s about all I can give in in relationship to your issue.
I like your idea of using country codes in the dial plan for multiple servers, that’s neat. I’ll save that for when I increase my servers from one. And yes, initial test from the phone itself, further expansion from a browser, and then maybe export, edit, and import to expand and tidy it up.
My first try is to have each button dial the server extension, pause, and send dtmf *9912345 (12345 is the node number). I’ll use the Poly enhanced feature key instructions I posted. I probably won’t be able to get to this for a few weeks but I’ll post the result. I do have an Avaya 9608G I can test with too.
Something like this, but done without the UC manager.
https://docs.poly.com/bundle/ucs-ag-6-1-0/page/t-ucs-ag-example-speed-dial-soft-key-with-a-pause.html
Thank you for the help. I’m looking forward to getting that large color display populated.
After your last post, I think I see your problem…
The string you are programming in the line buttons is wrong.
You have a misconception as to how this all works.
When talking from the phone to the server as sip or iax,
You are first talking to ASTERISK
So, there has to be a point in the sting that calls up the dialplan (extensions.conf)
From there you can point it to your node by number.
In short and basic, you would dial the node number, be received by the dial plan and connect you to that node (your programming) and once connected to the node you can *99 tx
So it does make a difference which ‘context’ you place the sip phone in as it will look within that context in extensions.conf for the extension you dial by number for a match.
being node#
for instance:
exten => 29285,1,Answer
exten => 29285,n,Wait(3)
exten => 29285,n,Playback(rpt/node)
exten => 29285,n,Playback(/var/lib/asterisk/sounds/rpt/nodenames/29285)
exten => 29285,n,Rpt,29285|P
You can use a different context than we use for the app_rpt radio channel, and I do, but the matching extention must be inside of that context.
While I am not sure what the default contexts in use of the system are with the last few versions of the software, they use to be
radio
radio-secure
radio -control
If you already have a radio-control, that would be a good start for your sip extensions, that is, unless you are creating large telephony applications which I doubt.
So, set your sip extension context (sip.conf) to a desired stanza in extensions.conf and create the dialplan for it there.
Once you have it correct, you can make one of your speed buttons to dial the node and a new one to engage the tx (*99) and one more to unkey if you like (i use pound but just enter them on the dialpad)
Anyway, let me know if I have corrected your thinking or you can correct mine LOL
I believe the skinny is that you were trying to dial the node and engage the tx in one speed dial.
But the commend for connecting to the node is just the extension number, no *
and it has to be connected already to engage the TX.
You might be able to achive that in one string if you insert a a puse and give time for the connection to be established. But I don’t think this is worth a speed button.
It becomes second nature once you have done it for a while to just use the dialpad.
Use the speed dials to dial the node(s) on speakerphone.
Thank Mike. I agree that having one button to dial the asterisk server, and once the call is established, use other means to control the nodes is the preferred option. I was only thinking of combining the dial and control in the same sequence because the Polycom tried to dial when the button was pressed. I’ll pursue using the keys to output dtmf, and as you say, manual keypad presses are not too onerous.
That enhanced features method seems to be the go. I’ll try it.
Thanks,
K2IP