SIP phone - Drop after DTMF cmd.

App_rpt-users
1

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=8&ved=0CEAQFjAF&url=http%3A%2F%2Fplcmtechnet.com%2Fdocuments%2Fvideo-collaboration%2Froom%2Fhdx-systems%2F3-1-3-2%2Frelease-notes%2Fknown-issues&ei=K1NqVeLbM43KogSanIKQCQ&usg=AFQjCNGKHbAzifP72f2S9pk9-Qun2ku94A&sig2=-RadxbQFhe-ohmN3ixCYfw

2

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

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