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jerry.geis at gmail.com
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 Posted: Thu Aug 06, 2020 7:10 am Post subject: [asterisk-users] asterisk 13.33 and polycom |
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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
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 |
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andres at telesip.net
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| Posted: Thu Aug 06, 2020 8:48 am Post subject: [asterisk-users] asterisk 13.33 and polycom |
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On 8/6/20 8:09 AM, Jerry Geis wrote:
| Quote: | 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.
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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
| Quote: | -- 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
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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