"Frank Griffith" <[EMAIL PROTECTED]> wrote:
To: <[email protected]>
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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