I got 1.3.4 already. No major changes, works smoothly so far.

Matthew Marlowe wrote:

Im working on getting 1.3.4.. Will post.


On Thu, 02 Dec 2004 13:15:29 -0500, Andrei (MPI)
<[EMAIL PROTECTED]> wrote:


I would sniff UDP packets with tcpdump and see what is going on:
separately in 10.24.102 and 10.24.100 segment.

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:





Theres 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




d




f
) released in October.

Though, it was proven over time that troubles with a SIP phone like




"not




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







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




ago...




and
those config files do really help!

Also, I suggest that you upgrade your SIP firmware if you have not




done




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




Interface.












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




to










find
one.








Extra Info:
Server is 192.168.0.3
Phone name/ext I want to be 301

Thanks,
Chris Cherry











_______________________________________________
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
CAUTION: This email message and accompanying data may contain
information
that is confidential. If you are not the intended recipient, you are
notified that any use, dissemination, distribution or copying of this
message or data is prohibited. If you have received this email message
in
error, please notify us immediately and erase all copies of this






message






and
attachments. Thank you.

CAUTION: This email message and accompanying data may contain
information that is confidential. If you are not the intended






recipient,






you are notified that any use, dissemination, distribution or copying






of






this message or data is prohibited. If you have received this email
message in error, please notify us immediately and erase all copies of
this message and attachments. Thank you.
_______________________________________________
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

_______________________________________________
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users










_______________________________________________
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users








_______________________________________________
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
 http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
 http://lists.digium.com/mailman/listinfo/asterisk-users






_______________________________________________
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
 http://lists.digium.com/mailman/listinfo/asterisk-users








_______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Reply via email to