* Not working after upgrade

"Frank Griffith" <glassdude45yahoo.com> wrote:
To: <asterisk-bsdlists.digium.com>
Sent: Monday, October 09, 2006 1:04 AM
Subject: Re: [Asterisk-bsd] * Not working after upgrade


> Totally whacked....my system is totally whacked. I
upgraded * from 1.2.9.1 
> to 1.2.12.1 via the ports collection and the audio
stopped working. 
> Incoming calls do not hear anything. So I reinstalled
1.2.9.1 but the 
> problem persisted. I then turned back on my older *
server which was not 
> upgraded and everything works fine again. But my new
system will not even 
> let me put 1.2.9.1 back on again.
>
>  The port is not doing what it should. Can anyone offer
some advice here?

and also wrote:
> > ...
> I'm only running * as a hobby, it's not a production
server. Still I 
> really count on it for my testing and increasing my
experience with VOIP. 
> So if you call harmless shutting my * server down for
incoming calls, then 
> I guess we don't see the same problem here. All my
incoming calls receive 
> no audio. Can't here anything even though the CLI shows
the GSM's are 
> playing.
>
>  In the meantime, I fired up an older server which was
running * 1.2.9.1 
> and at least I'm able to function again. I'm no
programmer so I'm at the 
> mercy of the bugfixers. Let me know how I can help
gang.

I'm still a little confused as to _exactly_ what you are
seeing.

Can you confirm my understanding that, with the default
dialplan (or a local 
equivalent), the demo (extension 1000) does not audibly play
the 
"Congratulations you have successfully installed and
executed..." message 
(or rather that the GSM file is being accessed, but the
audio is not 
reaching the client).

Also, above you said "So I reinstalled 1.2.9.1 but the
problem persisted. I 
then turned back on my older * server which was not upgraded
and everything 
works fine again. But my new system will not even let me put
1.2.9.1 back on 
again". I can't work out whether this means that you
did, or did not, manage 
to "downgrade" to 1.2.9.1 on the original system.

What device do your "incoming calls" come from -
is it a Digium FXO card, 
or a SIP or IAX server? What codec (do you think) is being
used?

As a data point - with 1.2.12.1 installed on FreeBSD 5.4 I
can successfully 
run the demo dialplan when inbound from a SIP device, and
can successfully 
route an incoming IAX call to an outbound SIP call with 2
way audio in both 
cases.

I note that all the problem reports for far have come from
people running on 
FreeBSD 6.X.
 - Is there anyone out there seeing this problem on 5.X?
 - Is there anyone running on 6.X and _not_ seeing this
problem?

--
Thomas Sandford 


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