Do you have any general NAT settings in sip.conf, like externip and local network etc? How about RTP port range?
It would be easier to understand your setup if you post your sip.conf, as well as phone's <MAC>.cfg file (which may be too big for this newsgroup - email is fine).
Also, asterisk sip registration and error messages would give us some more info.
Andrei
Tim Jackson wrote:
I've already added nat=yes. Nothing fancy/special on the routing between
these.
-Tim
-----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrei (MPI) Sent: Thursday, December 02, 2004 10:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500
Hello Tim,
You are saying that: phone is on "10.24.102.0/24" and Asterisk resides on "10.24.100.0/24". Honestly, I see at least one hop forwarding here and possible network issues right away.
At one moment, not in a NAT environment, but having phone IP 172.16.100.xxx and Linux server ip 172.16.1.xxx - I had to add nat=yes in sip.conf in order to get the phone to work.
Sincerely, Andrei
Tim Jackson wrote:
dTheres no NAT going on here. Just 1 router in between, phones reside on 10.24.102.0/24 Asterisk resides on 10.24.100.0/24, so this isn't a NAT problem. Any other ideas?
-Tim
-----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrei (MPI) Sent: Wednesday, December 01, 2004 11:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500
Tim,
You may see description of new 1.3.4 firmware at polycom.com (check - http://www.polycom.com/common/pw_cmp_updateDocKeywords/0,1687,3641,00.p
f ) released in October."not
Though, it was proven over time that troubles with a SIP phone like
hearing" one side or the other is NAT related problems. You may want to
investigate firewall setup. I am not saying it is not phone related,but
the phone would be the last one to blame.
Also, may I express my feelfings about Cisco and Polycom - not allowing
direct firmware download for their phones - that sucks big time. I will
get the firmware this way or the other. They just force me to waste my time again and again contacting their dealers and searching the internet. That should just enrage customers, in my opinion. Are they so
ago...big, they do not even care?
Sincerely, Andrei
Tim Jackson wrote:
Any idea if 1.34 fixes the problems with the phones being up for long periods of time and weird call problems (I cannot hear remote caller, but they can hear me) ?
-Tim
-----Original Message-----
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul
Hales
Sent: Wednesday, December 01, 2004 6:00 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Polycom IP500
Any idea if 1.34 makes Daylight Savings work for us people in
Australia?
PaulH
-----Original Message-----
From: Andrei (MPI) [mailto:[EMAIL PROTECTED] Sent: Thursday, 2 December 2004 9:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Polycom IP500
Hi Chris,
First of all, you need to configure ftp or tftp and watch syslog
closely -
what the phone is looking for at boot time. You would need to put
config
files into (t)ftp directory, named according to MAC address of you
phone.
XML and Web is really weird - they do not even share same config data.
For
example, I had to change address of SNTP server (clock) and it still
was
showing as 'clock' on Web admin page for the phone. I will email you
the
config files I got from a good fellow from this list not so long
doneand those config files do really help!
Also, I suggest that you upgrade your SIP firmware if you have not
Interface.it yet. I got the Polycom 500 with firmware which was very old and incapable to work with asterisk. Mine is 1.3.1 now (http://www.freedomphones.net/polycom/files/).
If anyone has SIP firmware 1.3.4 - please make it downloadable
somewhere
on the internet?
Thank you, Andrei
Chris Cherry wrote:
Hey All,
First Time Writing.
I'm trying to set up my IP500 phones to register SIP with *. I input all the (I assume) correct data in to the fields on the Web
to
And I get no notification that the phone is even attempting to register, no failed messages etc. I have read that the Web interface
is
crap and the XML config files is the way to go. Does anyone have a basic config file that doesn't change any defaults? I couldn't seem
find one.
_______________________________________________Extra Info: Server is 192.168.0.3 Phone name/ext I want to be 301
Thanks, Chris Cherry
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