Polycom VVX desk set disconnects at exactly 35 seconds

So I’m not really sure where to start debugging this, which is to say I’ve checked “all the things”. I’ve looked the debug logs for asterisk, the SIP conversation and the debug level logs in my Polycom phone. No hints at all.

I have a device [CALLSIGN] in sip.conf that looks like this:

[CALLSIGN]
deny=0.0.0.0/0.0.0.0
secret=REDACTED
dtmfmode=rfc2833
canreinvite=no
context=iax-client
host=dynamic
type=friend
port=5060
qualify=yes
dial=SIP/CALLSIGN
permit=0.0.0.0/0.0.0.0
trustrpid=yes
sendrpid=no
pedantic=no
qualifyfreq=20
transport=udp
encryption=no
allow=ulaw
callerid=CALLSIGN <CALLSIGN>
mailbox=CALLSIGN@device
faxdetect=no

I’m using the canned [iax-client] dialplan context (which works from Zoiper), and can dial into the repeater but my SIP device (a Polycom VVX 411 with working registrations on multiple other systems) gets disconnected at the 35s mark every time.

Happy to dig out logs, etc. Would any of you like to take a stab at this with me? Thanks in advance,

Andy

Here is the way I give my sip phone access to my node on the same server…

sip.conf

[210]
type=friend
host=dynamic
username=210
secret=xxxxxxxxxxxx
dtmfmode=rfc2833 
mailbox=210 ; Mailbox for message waiting indicator
context=sip-phones
callerid="Tim Line 1" <210>

extensions.conf

[sip-phones]
; Allow SIP calls to local nodes
exten => 25330,1,rpt(${EXTEN}|P)
exten => 2522,1,rpt(2522|P)

Let me know if this helps.

Thanks for your help Tim. I can’t put my finger on the specific difference that made the difference, but this config (which is only slightly different) is working just fine for me.

Would you be willing to point me in the right direction for the appropriate commands for PHONE mode? It looks like phone_functions is set to [functions] in rpt.conf. I see 99 is defined as cop,6 but it doesn’t seem like I’m getting PTT from it. Do I need to prefix it with anything? I may test your SIP config with the iax-client context, as that seemed to be working too.

Thanks!

Glad the config is working for you. Asterisk is exacting. The smallest config change can make a big difference.

You do need to do *99 to get PTT. All Asterisk functions start with an asterisk :slight_smile: except for the # by its self. Use # to unkey from your sip phone.

Your iax-client context is not likely to work with sip phones. IAX and SIP are two completely different protocols.