Mark, I have one suggestion to narrow the the hunt. Set-up a new sip extension and perhaps use a soft-sip-phone from you computer or linphone from your smartphone and see what you get. There are so many possibilities of problems with a unique sip-phone that you need to see if it is the phone set-up or with settings (or lack of them)in app_rpt. I will say you might try removing (rem out) the following setting for a test- ;progressinband=no Or a try standard good generic set-up [5010] ;iPad deny=0.0.0.0/0.0.0.0 username=5010 secret= dtmfmode=rfc2833 canreinvite=no context=radio-control host=dynamic trustrpid=yes sendrpid=no type=friend nat=no ;???????????????? port=5060 ;???????? qualify=yes qualifyfreq=60 transport=udp ;??????????????? encryption=no callgroup=rpbx2 pickupgroup=rphones2 dial=SIP/5010 mailbox=5010@device permit=0.0.0.0/0.0.0.0 callerid=iPad <5010> Asterisk is somewhat good with fuzzy logic but sometimes with unique phones you need to be specific. Let me know what you find out, ...mike/kb8jnm

I am trying to use a sip/hard phone - Polycom VVX500 If I try to use DTMF commands, they do work - but 30 seconds later, the call disconnects. - if I do not enter dtmf commands, seems to work fine - local connection using the 192.168 address - connected via gigabit ethernet switch - I also have the phone setup for callcentric - works fine there when I enter dtmf...

asterisk -r

Connected to Asterisk currently running on KC7DMF-1 (pid = 2689)

Verbosity is at least 3

-- Executing [29627@radio-control:1] Answer("SIP/polycom-0863ce98", "") in new stack

-- Executing [29627@radio-control:2] Playback("SIP/polycom-0863ce98", "rpt/node") in new stack

-- <SIP/polycom-0863ce98> Playing 'rpt/node' (language 'en')

-- Executing [29627@radio-control:3] Rpt("SIP/polycom-0863ce98", "29627|P") in new stack

== Spawn extension (radio-control, 29627, 3) exited KEEPALIVE on 'SIP/polycom-0863ce98'

-- <Zap/pseudo-1488761527> Playing 'rpt/goodafternoon' (language 'en')

-- <Zap/pseudo-1488761527> Playing 'rpt/thetimeis' (language 'en')

-- <Zap/pseudo-1488761527> Playing 'digits/5' (language 'en')

-- <Zap/pseudo-1488761527> Playing 'digits/20' (language 'en')

-- <Zap/pseudo-1488761527> Playing 'digits/8' (language 'en')

-- <Zap/pseudo-1488761527> Playing 'digits/p-m' (language 'en')

-- Hungup 'Zap/pseudo-1488761527'

-- Hungup 'Zap/pseudo-1505109141'

- sip.conf -

[polycom]

type=friend

context=radio-control

secret=secret-word-duh

host=dynamic

dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info

username=userpoly ; Username to use in INVITE until peer registers

disallow=all

allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!

progressinband=no ; Polycom phones don't work properly with "never"

- extensions.conf -

[radio-control]

exten=29627,1,Answer

exten=29627,n,Playback,rpt/node

exten=29627,n,Rpt,29627|Pv

found this: maybe related - I'm using a Polycom vvx 500

https://www.google.com/url?sa=t&rct=j&q=&esrc=s&source=web&cd=6&cad=rja&uact... _______________________________________________ App_rpt-users mailing list App_rpt-users@ohnosec.org http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users

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