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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>> [EMAIL PROTECTED]
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>
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