---------- In?cio da mensagem original -----------
De: asterisk-users-admin@lists.digium.com
Para: asterisk-users@lists.digium.com
Cc:
Data: Mon, 12 Jul 2004 11:48:05 -0500
Assunto: Asterisk-Users digest, Vol 1 #4502 - 11 msgs
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> Today's Topics:
>
> 1. Changed IP and subnet now no SIP Register 403
(Steve Totaro)> 2. Re: feature - VM gain adjust? (Chris Shaw)
> 3. dsp.c:1467 ast_dsp_process: Unable to process
inband DTMF on 2 frames (Stefan Rosik)> 4. Re: feature - VM gain adjust? (Rich Adamson)
> 5. Re: feature - VM gain adjust? (Steven
Critchfield)> 6. zaptel debugging tools (Glen Hinkle)
> 7. Re: permission problem (Wolfgang S. Rupprecht)
> 8. Re: New Asterisk bounty: SIP simultaneous
(Sunrise Ltd)> 9. Re: Gogoif with variables acting funny? (Shaun
Dawson)> 10. Re: PRI numbering plan (Martin List-Petersen)
> 11. Re: X101P FXO with RED alarm (Chris Stenton)
>
> --__--__--
>
> Message: 1
> From: "Steve Totaro" <asterisk@totarotechnologies.com>
> To: <asterisk-users@lists.digium.com>
> Date: Sat, 12 Jun 2004 11:40:07 -0400
> Subject: [Asterisk-Users] Changed IP and subnet now
no SIP Register 403> Reply-To: asterisk-users@lists.digium.com
>
> This is a multi-part message in MIME format.
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>
> I built a system and then changed the IP and subnet.
Now the phones > will not register, getting a 403.>
> Any ideas?
> ------=_NextPart_000_02C1_01C45072.02CC3E10
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Transitional//EN">> <HTML><HEAD>
> <META http-equiv=3DContent-Type content=3D"text/html;
> charset=3Diso-8859-1">
> <META content=3D"MSHTML 6.00.2800.1400"
name=3DGENERATOR>> <STYLE></STYLE>
> </HEAD>
> <BODY bgColor=3D#ffffff>
> <DIV><FONT face=3DArial size=3D2>I built a system and
then changed the > IP and=20> subnet. Now the phones will not register,
getting a > 403.</FONT></DIV>> <DIV><FONT face=3DArial
size=3D2></FONT> </DIV>
> <DIV><FONT face=3DArial size=3D2>Any ideas? >
</FONT></DIV></BODY></HTML>
>
> ------=_NextPart_000_02C1_01C45072.02CC3E10--
>
>
> --__--__--
>
> Message: 2
> From: "Chris Shaw" <chriss@watertech.com>
> To: <asterisk-users@lists.digium.com>
> Subject: Re: [Asterisk-Users] feature - VM gain
adjust?> Date: Mon, 12 Jul 2004 08:43:19 -0700
> Reply-To: asterisk-users@lists.digium.com
>
> Hmmm... I don't know if playing with the * code would
really be the best> here... Although if it was a plug-in app like
app_volume or something I> guess it couldn't hurt... It really sounds like you
have a line issue here.> You said that adjusting the gain on your card
introduced echo issues. It> sounds like you have an impedance mismatch/imbalance.
Like your telco is> trying to cut corners going from a 4-pair to 2-pair
or doing some creative> splitting... Do you possibly know where the source of
the echo might be> coming from? Maybe somewhere under your control? If
not it can be a pain> getting the telco to acknowledge/fix the problem.
>
> Most proprietary PBXs even would have this problem,
although they usually> don't introduce so much attenuation as your FXO card
seems to be doing... I> know I know * is way better than a PBX and it should
be more flexible. I'm> just saying that normally there's no way short of
getting the damn telco to> fix the problem or getting your own ISDN (T1 if
you're in the> Telco-Logically backward USA like me) with channel
bank... Even then they> don't always work...
>
> Just my $0.2 ...
>
>
>
> ----- Original Message -----
> From: "Rich Adamson" <radamson@routers.com>
> To: <asterisk-users@lists.digium.com>
> Sent: Monday, July 12, 2004 5:46 AM
> Subject: Re: [Asterisk-Users] feature - VM gain
adjust?>
>
> > > > Are you suggesting such a thing exists, or that
that would be a> > > > proposed future application?
> > >
> > > I propose to think if an AGC / dynamic compressor
could be used instead> of
> > > a config variable.
> > >
> > > Most sound editors have modules for this.
> >
> > So how would you detect the remote caller is 14.7
db away from *> > and adjust the 'outbound' voice message to be at
some higher> > audio level?
> >
> > I like the AGC approach, but I'm not sure its
realistic in terms of> > consistently being able to identify the
transmission loss from> > each and every vm call. Since we know what the loss
is for each> > pstn line (to the central office), it would appear
that static> > value would be a good starting point and the user
could adjust from> > there. Much easier (and more likely) to implement.
> >
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-
users> > To UNSUBSCRIBE or update options visit:
> >
http://lists.digium.com/mailman/listinfo/asterisk-users>
>
> --__--__--
>
> Message: 3
> Date: Mon, 12 Jul 2004 18:00:57 +0200
> From: Stefan Rosik <srosik@t-online.de>
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] dsp.c:1467 ast_dsp_process:
Unable to process inband DTMF on 2 frames> Reply-To: asterisk-users@lists.digium.com
>
> Hi can anyone help me on this error msg??
>
> dsp.c:1467 ast_dsp_process: Unable to process inband
DTMF on 2 frames>
> thnx
> St
>
>
> --__--__--
>
> Message: 4
> Date: Mon, 12 Jul 2004 10:51:04 -0600
> From: Rich Adamson <radamson@routers.com>
> Subject: Re: [Asterisk-Users] feature - VM gain
adjust?> To: asterisk-users@lists.digium.com
> Reply-To: asterisk-users@lists.digium.com
>
> > At 5:00 PM -0600 on 7/11/04, Rich Adamson wrote:
> > >I'm toying with adding a feature request to
provide some sort of> > >gain setting for voicemail when accessed
from "certain" interfaces.> > >Maybe something like voicemail=6.0 (db) within a
specific channel> > >section of zapata.conf corresponding to a pstn
line.> > >
> > >Situation:
> > >1. Someone calls into asterisk and leaves a
voicemail. The sound> > >is recorded at some volume well below 0 db, and is
directly related> > >to the distance asterisk is from the central
office (pstn cable> > >loss) plus whatever distance the user placing the
call is from> > >his/her central office.
> > >2. I receive a text message that a voicemail was
left.> > >3. I call into asterisk remotely (assume from a
cell phone) and> > >retreive the voicemail. My location creates
another xx db of loss> > >between myself and asterisk, and voicemail can
hardly be heard.> > >
> > >Actual Measured Values:
> > >1. Asterisk is 5.6 db from the central office.
Called from one> > >pstn line, through the central office, to asterisk
and sending a> > >1004 hz tone at 0db. Recorded the tone into
voicemail. (Tone should> > >have been recorded at about 11.2db, two times the
cable loss)> > >2. Called into asterisk again, this time to
retreive the voicemail> > >and measured the 1004 hz tone from voicemail. It
was -36db "actual".> > >This retreival added another 11.2db of loss due to
pstn interfaces> > >and plant loss.
> > >3. The calls were through a TDM FXO module with rx
and tx gains> > >set to 0. (Changing rx and tx gain to +3 db and
repeating the test> > >resulted in a measured -30.5db signal, but these
settings create> > >unwanted echo issues. Therefore adjusting channel
gain is not an> > >option.)
> > >
> > >The end result is that retreiving any voicemail
message left from> > >a distant location and retreived from a distant
location can hardly> > >be heard. By adding the proposed voicemail=6.0
statement to the> > >appropriate channel, any calls connected to
voicemail via that> > >channel would effectively increase transmission
levels by 6db (or> > >whatever the setting happened to be). In this
example case, the> > >setting would increase the vm volume by 12db (or
about 24db measured> > >in the above).
> > >
> > >Anyone have any thoughts on this?
> > >
> > >Rich
> >
> > Rich -
> > I'll say that this would be very useful.
Regardless of where the > > loss is being inserted, it still exists.
> >
> > I like the idea of associating the voicemail db
adjustment on a > > per-channel basis. I don't want to have to dink
around with yelling > > at the telco to fix something that "just works"
otherwise. Their > > answer will be "Well, turn up the volume on your
phone!" which is > > exactly what your proposed patch will do. A simple
trial-and-error > > process should be able to sort out the proper
adjustment on any > > typical system that doesn't have radical db changes
across time. I'm > > heartily in favor of this idea; I'll even throw a
donation towards > > it, if you have a PayPal account.
> >
> > Another cool feature would be app_volume, which
would turn up/turn > > down tx/rx levels dynamically, but that's left for
a different day, > > and after we have an enhanced app_dial that lets
single-digit dtmf > > sequences jump to dialplan routines and then can
reconnect bridged > > calls. See my various rantings about this in
months (years!) past. > > When I get some spare time (ha ha ha) I should
really learn how to > > code this stuff...
> >
> > JT
>
> The above "feature" request has been entered as bug
#2023.>
> It also appears that VM has an issue (by itself) with
recording/playing> volume. Transmitting a 1004hz tone at 0db through a
ata186 (set for> -1db fxs loss), and then retreiving the same VM
results in that tone> measured at ~ -10db. Doing the same from a pstn
location (via TDM FXO)> suggests the same -10db loss (in addition to the pstn
loss). Zapata.conf> rxgain and txgain set to 0. Using CVS-HEAD-07/12/04,
but same result> with CVS-HEAD-07/1/04. Entered as bug #2022.
>
> Add comments to either if you'd like.
>
>
>
>
> --__--__--
>
> Message: 5
> Subject: Re: [Asterisk-Users] feature - VM gain
adjust?> From: Steven Critchfield <critch@basesys.com>
> To: asterisk-users@lists.digium.com
> Date: Mon, 12 Jul 2004 11:08:33 -0500
> Reply-To: asterisk-users@lists.digium.com
>
> On Mon, 2004-07-12 at 09:31, Seth Remington wrote:
> > What about a "post processor" that performs
Compression/Normalization on> > the recorded voice mail file?
> >
> > On the down side I can see this being a big CPU hog
if you are handling> > a huge amount of calls and trying to normalize a 5
minute long voicemail> > at the same time.
> >
> > On the upside you don't have to concern yourself
determining line loss> > or similar things. You also wouldn't have to worry
about what I call the> > "Seinfeld Syndrome": quit talker / loud talker
issues. You would just> > have two new variables in voicemail.conf -
normalization=yes or no and> > another to set the db value.
>
> While I have tried to stay out of the comments here
for a while, I would> suggest not going post processing. While it might get
the problem fixed> for now, it isn't a good long term solution.
>
> I have experienced similar trouble with recordings
from AGI. We have> some recordings that where dead on sound wise, and
others that ended up> being so soft as to be useless.
>
> Would it be something people would like to be able to
add filters to a> line? Consider normalization as a filter. Monitor
could then be moved to> a filter as well. Echo cancel could be a filter. Set
it up so multiple> filters could be added and chained together. This
could help those with> echo chain a couple of filters together and see if
that helps.>
> --
> Steven Critchfield <critch@basesys.com>
>
>
> --__--__--
>
> Message: 6
> From: Glen Hinkle <asterisk@empireenterprises.com>
> To: asterisk-users@lists.digium.com
> Date: Mon, 12 Jul 2004 12:09:37 -0400
> Subject: [Asterisk-Users] zaptel debugging tools
> Reply-To: asterisk-users@lists.digium.com
>
> Are there any debugging tools for the digium zaptel
cards that would> report the activity on the line, such as DTMF and/or
connection> protocol?
>
> I'm looking to debug the connection with a T100P, & I
don't have $2000> for a T1 test set.
>
> Thanks,
> Glen
>
>
>
> --__--__--
>
> Message: 7
> To: asterisk-users@lists.digium.com
> Date: 12 Jul 2004 09:14:36 -0700
> From: "Wolfgang S. Rupprecht" <list+asterisk-
users@lists.wsrcc.com>> Organization: W S Rupprecht Computer Consulting,
Fremont CA> Subject: Re: [Asterisk-Users] permission problem
> Reply-To: asterisk-users@lists.digium.com
>
>
> cyprien.simons@tu-berlin.de (Cyprien Simons) writes:
> > Is the only way to use asterisk _not_ as root to
change the> > permission of all the directories where asterisk
need to create a> > file? ("/var/run/", "/var/log/asterisk/messages")
> >
> > any help will be appreciated,
>
> Grab my patches below. It does both chroot and
setuid to user> "asterisk". (You might need to back out one or two
of the obvious> Openbsd fixes.)
>
> I've been running chroot and as user "asterisk" for a
few weeks now on> this sip-only server. There are still few loose ends
(like "music on> hold" not running correctly, but part of that appears
to be an> asterisk scheduler problem under OpenBSD that happens
even with no> chroot etc.)
>
> -wolfgang
> --
> Wolfgang S. Rupprecht
http://www.wsrcc.com/wolfgang/> openbsd amd64
http://www.wsrcc.com/wolfgang/ftp/asterisk-
openbsd35.patch>
> --__--__--
>
> Message: 8
> Date: Tue, 13 Jul 2004 01:20:11 +0900 (JST)
> From: Sunrise Ltd <stsltdtyo@yahoo.co.jp>
> Subject: Re: [Asterisk-Users] New Asterisk bounty:
SIP simultaneous> To: asterisk-users@lists.digium.com
> Reply-To: asterisk-users@lists.digium.com
>
> in response to Olle's excellent post, ...
>
> in particular ...
>
> >Asterisk is *not* a SIP proxy. It's a SIP registrar
and> location server.
> >It's a very clever SIP UA. It wants to be in the
middle> of the call
> >and wants to be in control of each device. This
> device-slave view >doesn't match the SIP architecture.
>
> and ...
>
> >I've spent a considerable amount of time
investigating> support for
> >multiple registrations on one Asterisk sip [peer]
account> and after
> >learning about Asterisk's architecture come to the
> conclusion that
> >it is not an easy or even a desirable feature to
> implement.
>
> and ...
>
> >It may be possible, but will probably lead to a lot
of> changes to
> >Asterisk, both core and applications, that no other
> channel will
> >benefit from. A quick hack to support it may lead to
a> lot of
> >confusion on how to handle other apps. And it's a lot
> more work
> >than the bounty will cover. I suggest that you use a
> forking SIP
> >proxy in conjunction with Asterisk to get this
> functionality.
>
> Precisely! A fairly simple and elegant solution.
>
> For those rare occasions where one would really need
> multiple concurrent SIP registrations I'd say one
should> consider running Asterisk in combination with a SIP
proxy.> Since SER is a free download, this wouldn't seem to be
> such a big deal IF IT WASNT for the fact that one will
> then need to run two boxes.
>
> It would make a lot of sense to provide support for an
> easy-to-configure set up where Asterisk can live
together> with another SIP speaking piece of software on the
same> box.
>
> Something along the lines of ...
>
> (ip1:5060)---[*]---[portswapper]---(ip1:5061)---[SER]-
--(ip2:5060)>
> Something like this should allow you to run Asterisk
on> one address (ie LAN side) and SER on another (ie WAN
> side), so you get the best of both Asterisk and a SIP
> proxy all in one box.
>
> This would also make it possible to run a SIP
softphone> alongside Asterisk on a notebook, so it would solve
two> birds with one stone.
>
> I'd like to emphasise however, that most of the
problems> described in this thread are NOT good reasons for
multiple> concurrent SIP registrations. Those problems have
other> solutions. Let's take a look at them.
>
> 1) Call centre scenario
>
> Problem: multiple agents should receive calls on the
same> phone number
>
> Solution: assign a number to a call queue and let the
call> queue distribute incoming calls to the agents on
different> SIP phones, each of which should have unique logins
for> reasons of accounting and quality assurance.
>
> multiple concurrent registrations on the same SIP
account> in call centres is a BAD IDEA.
>
> 2) Overworked admin scenario
>
> Problem: asterisk admin doesn't want to deal with
support> calls for adding additional SIP phones
>
> Solution: a simple self provisioning system, either
web> based or even IVR based.
>
> 3) Dual line desk phone scenario
>
> Problem: dual line desk phone requires multiple
> registrations, one per line
>
> Solution: let the phone register on two different SIP
> accounts, which is how any conventional PBX handles
dual> line phones: one extension per line.
>
> 4) Call group scenario
>
> Problem: multiple phones to ring on the same extension
>
> Solution: use the call group feature or use the dial
> command with multiple SIP peers
>
>
> For the avoidance of doubt, I am not saying there is
no> situation for which multiple concurrent SIP
registrations> may be the right solution, but the problems described
so> far are *not*.
>
> But if anybody has a problem that truly warrants
parallel> forking, then I propose you look into sponsoring
somebody> to work on the little port swapping trick to run SER
> concurrently on your Asterisk box.
>
> rgds
> benjk
>
>
> __________________________________________________
> Do You Yahoo!?
> http://bb.yahoo.co.jp/
>
>
> --__--__--
>
> Message: 9
> Date: Mon, 12 Jul 2004 09:26:02 -0700 (PDT)
> From: Shaun Dawson <bigchiefscd@yahoo.com>
> Subject: Re: [Asterisk-Users] Gogoif with variables
acting funny?> To: asterisk-users@lists.digium.com
> Reply-To: asterisk-users@lists.digium.com
>
>
> <snip>
>
> > -- Executing SetVar("Zap/99-1",
"counter=[0+1]")
> > in new stack
> > -- Executing GotoIf("Zap/99-1",
> > "[[0+1]<3]?s|7:h|1") in new stack
> > -- Goto (inbound-analog,h,1)
>
> <snip>
>
> >
> > It looks to me as if the Gotoif thinks that [0+1] is
> > greater than or
> > equal to 3 and therefore jumps to hangup.
> >
> > Am I missing something here?
> >
>
> I apologize in advance for the stupid question, but
> is it at all possible that counter is being evaluated
> in a string context either in the additionor the
> GoToIf command? (One quick way to check that is to
> see what happens if you put a second addition in right
> after the first, and see if you get '2', or
> '[[0+1]+1]').
>
>
> Shaun
>
>
>
> __________________________________
> Do you Yahoo!?
> New and Improved Yahoo! Mail - Send 10MB messages!
> http://promotions.yahoo.com/new_mail
>
> --__--__--
>
> Message: 10
> Subject: Re: [Asterisk-Users] PRI numbering plan
> From: Martin List-Petersen <martin+asterisk@list-
petersen.net>> To: asterisk-users@lists.digium.com
> Cc: al.maw@mxtelecom.com
> Date: Mon, 12 Jul 2004 17:32:59 +0100
> Reply-To: asterisk-users@lists.digium.com
>
> On Mon, 2004-07-12 at 15:30, Alastair Maw wrote:
> > On 12/07/04 11:11, Michael Sandee wrote:
> > > pridialplan=unknown
> > > prilocaldialplan=national
> >
> > Not only is this that undocumented, but the
string "prilocaldialplan" > > doesn't even show up in the latest CVS HEAD source
code, so that's not > > going to work...
>
> prilocaldialplan is not something that is part of
asterisk, but> introduced in the patches from kapejod's bristuff
0.0.2> (http://www.junghanns.net), which add's BRI zaptel
telephony to> asterisk. That is the reason why some people have it
and some not.>
> Kind regards,
> Martin List-Petersen
>
>
>
> --__--__--
>
> Message: 11
> From: "Chris Stenton" <jacs@gnome.co.uk>
> To: <asterisk-users@lists.digium.com>
> Subject: Re: [Asterisk-Users] X101P FXO with RED alarm
> Date: Mon, 12 Jul 2004 17:34:12 +0100
> Reply-To: asterisk-users@lists.digium.com
>
> Richard,
>
> 1. don't run 0.5 zaptel driver with asterisk-head it
will panic the kernel.> 2. I am pretty sure that the current BSD zaptel
driver only supports the fxs> modules and the x100p card.
>
> Chris
>
> ----- Original Message -----
> From: "Richard Airlie" <richard@darq.net>
> To: <asterisk-users@lists.digium.com>
> Sent: Sunday, July 11, 2004 11:02 PM
> Subject: Re: [Asterisk-Users] X101P FXO with RED alarm
>
>
> > On Sat, Jul 10, 2004 at 05:55:21PM +0100, Kevin
Walsh wrote:> > > Richard Airlie [richard@darq.net] wrote:
> >
> > > First things first. Scrap the "ports" and build
from the latest> > > CVS source. 0.9 is far to old and buggy, and
suspect the same of> > > the Zaptel driver you have, although I don't use
*BSD myself.> >
> > I cvsup'd to the latest source yesterday and tried
to build zaptel,> > but it failed right away. (trying to include
linux/*.h)> > I didn't try building asterisk as it seems like the
problem is with> > zaptel -- i.e. I should be able to load the zaptel
driver and not> > see a red alarm, irrespective of my asterisk
version, right?.> >
> > > Secondly, the red alarm does tend to mean that
the line is not> > > connected, but I got what you're describing when
I moved Asterisk to> > > a new machine. Try the X100P card in a different
PCI slot. That> > > cleared it for me, for whatever reason.
> >
> > Thanks for that, I gave it a try but unfortunately
it's made no> > difference.
> >
> > I am suspecting the problem is either with the
zaptel driver in> > ports (which is the only version I can get to
build) or i've got> > a hardware issue.
> >
> > For what it's worth I can plug a phone into the
back of the FXO> > and get dial tone, so I guess that proves that the
cabling is OK?> >
> > best,
> > Richard.
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-
users> > To UNSUBSCRIBE or update options visit:
> >
http://lists.digium.com/mailman/listinfo/asterisk-users> >
>
>
>
> --__--__--
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-
users>
>
> End of Asterisk-Users Digest
>
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