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

John Todd jtodd at loligo.com
Sun Sep 7 11:48:41 MST 2003


Rich -
   Leave out the "allow" lines entirely, including the "allow=all" - 
this was a problem I discovered post-publishing (that I thought I 
corrected in the notes, but I see that it's not there.)  The "allow=" 
lines need a "disallow=" line to balance them.  If you leave both 
out, the system will choose the "right" codec, but if you only put 
one in, things get twisted up a bit.  I've updated the article 
(again?)

JT



>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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>
>---------------End of Original Message-----------------
>
>
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