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 <[EMAIL PROTECTED]> 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 <[EMAIL PROTECTED]> 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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