[Asterisk-Users] Polycom IP500

Thu Dec 2 11:15:29 MST 2004

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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