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

[Rich Adamson]

John,

Excellent, I removed them and works fine.
I started playing with MOH, but haven't seen an example of how to 
specify this on a per system or per extension (but I haven't googled
for it yet either). I do have one line uncommented in the moh config
file, but I assume I need to do something in sip.conf or extensions.conf
to make it work.

Suggestions?

Rich

------------------------
> 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
> >>  http://lists.digium.com/mailman/listinfo/asterisk-users
> >
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> >
> >
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