[Asterisk-Users] Polycom IP500
Andrei (MPI)
asterisk at markovprocesses.com
Fri Dec 3 15:39:59 MST 2004
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)
><asterisk at markovprocesses.com> 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: asterisk-users-bounces at lists.digium.com
>>>[mailto:asterisk-users-bounces at lists.digium.com] 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: asterisk-users-bounces at lists.digium.com
>>>>[mailto:asterisk-users-bounces at lists.digium.com] 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: asterisk-users-bounces at lists.digium.com
>>>>>[mailto:asterisk-users-bounces at lists.digium.com] 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:asterisk at markovprocesses.com]
>>>>>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
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
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>>>>>
>>>>>
>>>>message
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>>and
>>>>>attachments. Thank you.
>>>>>
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>>>>recipient,
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