[Asterisk-Users] Polycom IP500

Andrei (MPI) asterisk at markovprocesses.com
Thu Jan 6 10:07:54 MST 2005


Tim,

For what it's worth, from my working sip.conf for Polycoms:

[2010]
type=friend
username=usr2010
callerid="MyName" <2010>
secret=nobodyknowswhatitis
host=dynamic
dtmfmode=inband
context=admin
defaultip=192.168.1.10
progressinband=no

Notes:

dtmfmode=inband and progressinband=no - that seems to be recommended 
from * sample sip.conf file for Polycoms.

defaultip= setting helped with network issues, not only with Polycoms, 
with Cisco 7940 as well.

Also in main sip.conf:
[general]
...
disallow=all              ; Allow all codecs
allow=ulaw,alaw

maxexpirey=7200
defaultexpirey=3600
canreinvite=no

Also, if you are not behind NAT, why nat=yes? And if NAT is in use, what 
is your network infrastructure?

Also, what is Polycom's SIP firmware version? (mine is 1.3.4 from 
October 2004).

And of course: what is Asterisk and zaptel version? What is your 
zapata.conf (just curious)?

Andrei

Tim Jackson wrote:

>Earlier tonight I moved our * box from an old Compaq w/ 3 X100P clone
>cards to a 1U IBM server with a TDM04B card. I finally got the card
>working in the server, but I'm having issues with these Polycom IP500s
>now. Using the exact same config from the old server I'm getting weird
>errors. Dial a number on the phone and it gives you dialtone but no user
>interaction (if that makes sense) then after about 35-40 seconds it
>displays "Line used remotely" and hangs up. Inbound calls ring, but you
>can't answer them, registration seems to be ok, but I'm at a loss.
>
>sip.conf:
>[101]
>type=friend
>callerid="Tim Jackson - Home" <101>
>secret=itsasekret
>username=101
>host=dynamic
>dtmfmode=rfc2833
>nat=yes
>canreinvite=no
>context=default
>allow=all
>  
>




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