[Asterisk-Users] New cvs compile; basic operational question,
please.
John Todd
jtodd at loligo.com
Sun Sep 7 14:56:58 MST 2003
Rich -
To get MOH working, I'd suggest looking at the archives for this
list a bit - lots of clues there. Ensure that you actually have
mpg123, and not mpg321. Restart asterisk. It should "just work" with
7960's.
JT
>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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>> >
>> >---------------End of Original Message-----------------
>> >
>> >
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>---------------End of Original Message-----------------
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