[Asterisk-Users] VM low volume - testers needed

Sat Sep 24 12:33:16 MST 2005

Hmm. Thanks for the heads up, but I'm not sure that's it.

It's jumping to 208 rather than 209, so it looks more like an off-by-one
error.

I tried changing to priorityjumping=yes in /etc/asterisk/extensions.conf and
reinstalled the CVS-HEAD version, but it still jumps to 208 whereas it used
to jump to 209.

On 9/24/05, Julian Lyndon-Smith <asterisk at dotr.com> wrote:
>
> Under 1.2 the +101 jumping is not enabled by default. There is a
> variable returned showing the status of the application. You need to add
> a "j" flag or put priorityjumping=yes in extensions.conf
>
> Julian.
>
> Brian McEntire wrote:
> > Hmmm...
> >
> > I checked out CVS-HEAD, built and installed it this morning. Most
> testing
> > was going well, but then I found out the behavior of ChanIsAvail has
> changed
> > (is broken?)
> >
> >
> > In my Dial Plan, if a call comes in on the PSTN line, and is not
> answered by
> > the extension (or if the extension is busy), ChanIsAvail checks to see
> of
> > the outgoing VOIP line is available. If so, it forwards the call to the
> VOIP
> > voice mail. If not, it forwards the call to the Asterisk Voicemail.
> >
> > With 1.2-beta, ChanIsAvail works for me. With CVS-HEAD, it hangs up on
> the
> > caller.
> >
> > Here is the relevant portion of my extensions.conf:
> >
> > exten => s,7,Dial(${PHONE1},15)
> > exten => s,8,Goto(108)
> > exten => s,108,ChanIsAvail(${VOIP1})
> > exten => s,109,Dial(${VOIP1}/${VOIPNUM})
> > exten => s,209,VoiceMail(123|sbg(6))
> >
> >
> > In the globals section, VOIP1 is set equal to Zap/4
> >
> > With 1.2-beta, -vvv logs show this, which is successful:
> >
> > -- Executing ChanIsAvail("Zap/3-1", "Zap/4") in new stack
> > -- Executing VoiceMail("Zap/3-1", "123|sbg(6)") in new stack
> > -- Playing '/var/spool/asterisk/voicemail/default/123/busy' (language
> 'en')
> >
> >
> > With CVS-HEAD -vvv logs show this, which is unsuccessful:
> >
> > -- Executing ChanIsAvail("Zap/3-1", "Zap/4") in new stack
> > == Spawn extension (incoming-pstn, s, 208) exited non-zero on 'Zap/3-1'
> > -- Hungup 'Zap/3-1'
> >
> >
> > Is there another list or someone I should mention this to? Asterisk
> should
> > not hangup Zap/3-1 at this point.
> >
> >
> >
> > On 9/24/05, Rich Adamson <radamson at routers.com> wrote:
> >
> >>The patch is in cvs-head, which has been very stable for me. :)
> >>
> >>------------------------
> >>
> >>
> >>>Hi Richard,
> >>>I am experiencing the same problem. I'd like to test your patch. Thing,
> >>
> >>is, I don't know which
> >>CVS it's in :)
> >>
> >>>... I checked out 1.2-beta on Tuesday (9/21) and compiled it. When I
> >>
> >>type 'show application
> >>voicemail', it does not describe the
> >>
> >>>g(#) option, so I think my version must not have it.
> >>>
> >>>I am using a TDM22B card and voicemails seem very quiet if they are
> left
> >>
> >>from in incoming POTS
> >>connection. When I enter
> >>
> >>>voicemail by direct dialing a local extension and leave a message from
> >>
> >>the advanced options
> >>menu, the recorded message is much
> >>
> >>>louder.
> >>>
> >>>I should qualify, not only are my VMs coming in over POTS, I am
> actually
> >>
> >>calling out first
> >>through the TDM22B, to Sipura, to
> >>
> >>>VOIP provider, back in via PSTN, to TDM22B, to VM. I'm amazed it works
> >>
> >>at all :) ... I'm very
> >>impressed by Asterisk and
> >>
> >>>especially it's voicemail. I would like to resolve the low volume issue
> >>
> >>though.
> >>
> >>>If you can tell me which CVS to check out, I can try it. I'd like to
> >>
> >>stick to the 1.2-beta
> >>branch though because I don't want to
> >>
> >>>rework all my config files.
> >>>
> >>>On 9/21/05, Rich Adamson <radamson at routers.com> wrote:
> >>>
> >>>
> >>>>On Monday 19 September 2005 12:38, Rich Adamson wrote:
> >>>>
> >>>>>The g(6) adds a 6 db gain for zap calls that end up recording a
> >>
> >>Voicemail
> >>
> >>>>>message.
> >>>>
> >>>>...
> >>>>
> >>>>
> >>>>>* 'g(#)' the specified amount of gain will be requested during
> >>
> >>message
> >>
> >>>>>recording (units are whole-number decibels (dB))
> >>>>
> >>>>How in the hell does that make any sense? are your normal incoming
> >>
> >>calls
> >>
> >>>>quiet too or just voicemail?
> >>>
> >>>Yes, see bug 2022 and 2023 for details, as well as
> >>>http://www.routers.com/asteriskprob/asterisk-config.htm
> >>>for a very detailed analysis of the problem.
> >>>
> >>>I believe one of the more serious issues amounts to: if asterisk is
> >>>located a fair distance from the central office (-7db in my case),
> >>
> >>setting
> >>
> >>>the rxgain and/or txgain to any level that would be considered
> >>
> >>reasonable
> >>
> >>>for that loss (eg, rxgain=5, txgain=5), hugh amounts of echo result
> that
> >>>cannot be addressed through zapata.conf echo entris, and changing
> >>>compile options to agressive, etc, does not help. Its my believe
> >>>(from working with several TDM users), the further one is from the CO,
> >>>the bigger the problem. (Or, short pstn cable lengths less then about
> >>>4 or 5db can almost always be addressed via parameters.)
> >>>
> >>>The above workaround is very usable (assuming it works) when someone
> >>>calls in via the pstn and leaves a voicemail (which is already at
> >>>least 7db down plus their own pstn loss), and then I call in via the
> >>>pstn to retrive the voicemail (now 14db down PLUS the original callers
> >>>pstn loss), the audio is so faint its difficult to impossible to
> >>>listen to.
> >>>
> >>>
> >>>>>In my case, the asterisk box is located about 7db from the central
> >>>>>office. As noted in bug 2023 (and 2022), calls from an outside pstn
> >>>>>line coming into asterisk incure a 7db pstn loss (which can't be
> >>
> >>adjusted
> >>
> >>>>>for with rxgain and txgain as changing those values to something
> >>>>>reasonable generates echo). Retrieving that VM message from an
> >>
> >>outside
> >>
> >>>>>location creates another 7db loss (now -14db down in total), making
> >>
> >>it
> >>
> >>>>>very difficult (if not impossible) to hear the message. (And, yes
> >>
> >>I've
> >>
> >>>>>gone through all the recommendations with wav vs gsm files, etc.)
> >>>>
> >>>>I am not sure I understand why the txgain/rxgain isn't fixing it
> >>
> >>without
> >>
> >>>>adding unacceptable echo... this all seems very odd... I mean for a
> >>
> >>test
> >>
> >>>>you should be able to dial an echo() application and have extremely
> >>
> >>quiet
> >>
> >>>>echoed audio... is this the case?
> >>>
> >>>As an ex-telco transmission engineer, believe me I've done my homework
> >>>and some very solid testing with expensive well-calibrated test
> >>
> >>equipment.
> >>
> >>>As I've mentioned to Kevin, its almost like the TigerJet pci controller
> >>>on the TDM card is reversing bits six and seven (or something very odd
> >>>like that). Digium apparently now has a pci engineering type looking
> >>>at the issues, which I'm told is using a pci logic analyzer, etc.
> >>>
> >>>
> >>>>>The work around "only" kicks in if the call comes from a zap channel
> >>>>>and ends up in voicemail, adding a 6db gain to that recorded
> >>
> >>message.
> >>
> >>>>>No other channel types are impacted by this new parameter.
> >>>>
> >>>>This is a HELL of a band-aid.
> >>>
> >>>If you actually follow the logic that was originally stated in 2023,
> >>>this gain setting "is" highly useful for those systems that are
> >>>further away from the CO (as mentioned above). For those closer to
> >>>the CO, it has zero value.
> >>>
> >>>Rich
> >>>
> >>>_______________________________________________
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> <http://Easynews.com>--
> >>>
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> >>
> >>---------------End of Original Message-----------------
> >>
> >>
> >>
> >
> >
> >
> > ------------------------------------------------------------------------
> >
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