[Asterisk-Users] New cvs compile; basic operational question, please.

Rich Adamson radamson at routers.com
Sun Sep 7 11:16:30 MST 2003


Just stumbled across the problem noted in my original post below. I added:
allow=ulaw
allow=ilbc

to sip.conf instead of the recommended 'allow=all' and now all phones work.

Can someone help me understand this?  (It would appear, based on my very much
lack of experience, that * was attempting to set up the conversation using g723,
when all of the phones have 'default=ulaw' definitions. Should I leave the
ulaw definition for future production use, or is this really something that
I did to read/learn more about for a very small office use?)

Rich

------------------------
> Can someone offer a hint on what I'm doing wrong with the basic * config?
> 
> Just implemented * for the first time using yesterday's cvs. The initial
> configs are based on John Todd's article at http://www.onlamp.com/lpt/a/3956,
> and using two 7960's for initial testing. When one 7960 calls the other, I
> get the following and the call is dropped:
> 
>     -- Executing Dial("SIP/3001-ec1c", "SIP/3000|20") in new stack
>     -- Called 3000
>     -- Got SIP response 488 "Not Acceptable Here" back from 206.222.193.92
>   == No one is available to answer at this time
>     -- Executing VoiceMail("SIP/3001-ec1c", "u3000") in new stack
>   == Parsing '/etc/asterisk/voicemail.conf': Found
>     -- Playing 'vm/3000/unavail'
> 
> My sip.conf looks like:
> [general]                                                                
>                                                                     
> port = 5060             ; Port to bind to (SIP is 5060)             
> bindaddr = 0.0.0.0      ; Address to bind to (all addresses on machine)
> allow=all               ; Allow all codecs                               
> context = bogon-calls   ; Send SIP callers that we don't know about here
>                                                                     
> [3000]                                                    
> type=friend             ; This device takes and makes calls              
> username=3000           ; Username on device                             
> secret=npi2003          ; Password for device                       
> host=dynamic            ; This host is not on the same IP addr every time
> context=from-sip        ; Inbound calls from this host go here
> mailbox=100             ; Activate the message waiting light if this     
>                         ;  voicemailbox has messages in it    
>                                                                          
> [3001]                                                                   
> type=friend             ; This device takes and makes calls
> username=3001           ; Username on device                  
> secret=npi2003          ; Password for device                       
> host=dynamic            ; This host is not on the same IP addr every time
> context=from-sip        ; Inbound calls from this host go here           
> mailbox=100             ; Activate the message waiting light if this     
> ---------------------------------------
> 
> and my extensions.conf looks like:
> 
> [general]
> static=yes              ; These two lines prevent the command-line interface
> writeprotect=yes        ; from overwriting the config file. Leave them here.
> 
> [bogon-calls]
> exten => _.,1,Congestion
> 
> [from-sip]
> exten => 3000,1,Dial(SIP/3000,20)
> exten => 3000,2,Voicemail(u3000)
> exten => 3000,102,Voicemail(b3000)
> exten => 3000,103,Hangup
> 
> exten => 3001,1,Dial(SIP/3001,20)
> exten => 3001,2,Voicemail(u3001)
> exten => 3001,102,Voicemail(b3001)
> exten => 3001,103,Hangup
> 
> exten => 3999,1,VoicemailMain(${CALLERIDNUM})
> 
> Apparently I'm doing something wrong, but since this is my first attempt
> at making * work, I don't actually have a clue what I'm doing (yet).
> 
> Asterisk did complile and install clean the first time (on new RH9 system),
> and both 7960's are registered. In some attempts to dial, I do receive
> vmail announcements, etc, so whatever I've done wrong I'm guessing it must
> be in the above config statements. 
> 
> If someone would kindly point out my error (and maybe a constructive comment
> about the error so I can learn), if would be greatly appreciated.
> 
> TIA,
> Rich
> 
> 
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> Asterisk-Users at lists.digium.com
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