[Asterisk-Users] Re: Polycom phones do not talk to each other

[Kanuri, Seshu (Company IT)]

>Also, I'm sure you've probably checked on this one, 
>but are the phones registered with asterisk?  
>You can make outbound calls on them without them 
>actually being registered.  I'm assuming you can 
>still get in and see the CLI.  What does "sip show peers" 
>look like?  What does "sip show peer xxx" show?  
>What does the CLI show when you dial a phone?

The Phones are registered with Asterisk. 
They ring each other, through Asterisk, but 
DIALSTATUS appears as NOANSWER  even when you lift 
the handset

The CLI Log is here when Extension 50 Called 51.

VAR:  agi_request: init.sh    
VAR:  agi_channel: SIP/50-6c72    
VAR:  agi_language: en    
VAR:  agi_type: SIP    
VAR:  agi_uniqueid: asterisk-28947-1110484788.110    
VAR:  agi_callerid: 50 <50>    
VAR:  agi_dnid: 51    
VAR:  agi_rdnis: unknown    
VAR:  agi_context: default    
VAR:  agi_extension: 51    
VAR:  agi_priority: 1    
VAR:  agi_enhanced: 0.0    
VAR:  agi_accountcode:    

   Detected protocol 'sip' ...  200 result=1  
   Detected caller '50' ...  200 result=1  
   Set limit - 24  200 result=1  
   Limit not exceeded (3 < 24) for localextensions  200 result=1  
   Set limit - 5  200 result=1  
   Limit not exceeded (3 < 5) for 50_out  200 result=1  
   Detecting destination for '51' ...  200 result=1  
   Found Destination localextensions (range 51 - 51)  200 result=1  
   Setting destination 'localextensions' ...  200 result=1  
   This is local extension, skipping Time Based Dialing/miniLCR ...  200
result=1  
   Set limit - 24  200 result=1  
   Limit not exceeded (4 < 24) for localextensions  200 result=1  
   Detecting Vertical Services ...  200 result=1  
   Set limit - 1  200 result=1  
   Limit not exceeded (1 < 1) for 51_in  200 result=1  
   Checking for channel SIP/51 ...  200 result=1  

APP:  exec ChanIsAvail SIP/51  200 result=0  
   Dialing '51' ...  200 result=1  
APP:  exec Dial SIP/51|32|tr  200 result=0  
APP:  get variable DIALSTATUS  200 result=1 (NOANSWER)  
   Dialing Voicemail 51 ...  200 result=1  
APP:  exec Voicemail u51  200 result=-1  
APP:  answer  200 result=0  
   Playing macro 'vm-goodbye' ...  200 result=1  
APP:  stream file vm-goodbye  200 result=-1 endpos=6880  
APP:  hangup    
 

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Noah
Miller
Sent: Thursday, March 10, 2005 4:52 PM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Re: Polycom phones do not talk to each other

>> We have bought PBXware GUI from Bicom systems and configured 
>> extensions with Polycom Phones as UAs.
>>
>> The Polycom Phones can dial out and make calls but I cannot make 
>> extension to extension calling.
>>
>> Googling did not help much.
>>
>> As you may be aware PBXware is a closed source software GUI from 
>> Bicom Systems for configuring extensions. It is a good tool to 
>> configure and manage users and phones but it does not allow to do any

>> of the customization tasks that are possible by directly editing the 
>> .conf files, which may be required in for Polycom.
>>
>> However if this is an issue of configuration on the Phone itself, we 
>> want to be able to make changes and fix this problem.

> Never used pbxware, but the context the sip phones dial out using 
> specified in sip.conf needs to include the dialplan context of the 
> phones in extensions.conf. 
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