Asterisk 13.33 And Polycom
I am using asterisk 13.33.0 and POlycom phone with the latest firmware.
The polycom phone is behind a firewall, the server is in the cloud. If the polycom has just booted – it receives a call, after some time
(couple minutes) it no longer receives a ring. I see no errors in the CLI –
looks just like the previous call as far as I can tell.
Then reboot the phone and as soon as its ready call it and it rings just liek before. then some time later no longer rings.
— Executing [something@smvoice-dialout:4] Dial(“SIP/1005-000000ab”,
“SIP/526,30000,tT”) in new stack
== Using SIP RTP CoS mark 5
— Called SIP/526
— SIP/526-000000ac is ringing
526 is the extension in question. (my definition follows):
[526]
type=friend defaultnameR6
defaultuserR6
secret=XXXXXXXXX
dtmfmode=RFC2833
host=dynamic description=Polycom context=sip qualify=yes rtptimeout`
rtpholdtimeout`
rtpkeepalive`
callerid=”Polycom ”
qualify=no canreinvite=yes timezone=1
nat=force_rport,comedia disallow=all allow=ulaw allow=alaw allow=gsm
Thoughts on what is happening here or what to try?
Jerry
One thought on - Asterisk 13.33 And Polycom
Sounds to me like you need to enable keep alives on the Polycom so it keeps the NAT pinhole open in the outbound direction. It will also help to enable the qualify setting on the PBX itself for the extension so it keeps sending SIP messages to the phone ensuring connectivity in the inbound direction.
qualify=yes