On 8/6/20 8:09 AM, Jerry Geis wrote:> 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.
>
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
> -- Executing [something at 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
> defaultname=526
> defaultuser=526
> secret=XXXXXXXXX
> dtmfmode=RFC2833
> host=dynamic
> description=Polycom
> context=sip
> qualify=yes
> rtptimeout=60
> rtpholdtimeout=60
> rtpkeepalive=60
> 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
>
--
Andres
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