date:20050211

Re: [Asterisk-Users] Asterisk won't answer incoming analog line

2005-02-11 Thread Lane

On Saturday 12 February 2005 00:07, Rich Adamson wrote:
> > I had to return my TDM11B because it put the PSTN line 'off hook' the
> > moment I plugged it in and wouldn't hang it up.
> >
> > The new card seems to work because I can actually make an outgoing call
> > from the FXO port to my cell phone, so I'm pretty happy about that.
> >
> > But Asterisk doesn't recognize incoming calls from the PSTN.  If I dial
> > my home phone from my cell phone asterisk does not even recognize the
> > ring.
> >
> > Below is (I think) the relevant config files.  My question is:  How do I
> > make asterisk recognize when the phone is ringing?
> >
> > Thanks in advance
> >
> > lane
> >
> > ;zaptel.conf
> > fxsks=4
> > fxoks=1
> > loadzone = us
> > defaultzone=us
> > ;
> >
> > ;zapata.conf
> >
> > [channels]
> > language=en
> > rxwink=300
> > usecallerid=yes
> > cidsignalling=bell
> > hidecallerid=no
> > callwaiting=yes
> > usecallingpres=yes
> > threewaycalling=yes
> > cancallforward=yes
> > callreturn=yes
> > echocancel=yes
> > echocancelwhenbridged=yes
> > rxgain=0.0
> > txgain=0.0
> > callgroup=1
> > pickupgroup=1
> > immediate=no
> > signalling => fxo_ks
> > channel => 1
> > signalling => fxs_ks
> > channel => 4
> >
> > ; extensions.conf
> > [globals]
> > CONSOLE=Zap/1
> > TRUNK=Zap/4
> > TRUNKMSD=1
> >
> > [trunklocal]
> > exten => _9NXX,1,PlayBack(vm-dialout)
> > exten => _9NXX,2,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> > exten => _9NXX,3,Congestion
> >
> > [incoming]
> > exten => s,1,Answer
> > exten => s,2,DigitTimeout,5
> > exten => s,3,ResponseTimeout,10
> > exten => s,4,Playback(tt-weasels)
> > ;
> > [local]
> > ignorepat => 9
> > include => trunklocal
> > ;
> > [default]
> > include=incoming
> > include=local
>
> In the zapata.conf definitions for the two channels, do something like
> this:
>
> immediate=no
> signalling => fxo_ks
> channel => 1
> context=incoming  ; < where to send incoming calls within
> extensions.conf signalling => fxs_ks
> channel => 4
>
> Add the above statement, and then your dialplan context 'incoming'
> will be executed when receiving a call.
>
> It might also help the learning process if you reorganize those
> two channels so that you can see which parameters are being applied
> to which channels. Something like this says exactly the same thing
> as above:
>
> [channels]
> language=en
> busydetect=yes
> busycount=6
> cidsignalling=bell
> cidstart=ring
> callerid=asreceived
>
> ; this section is for the fxs port (telephone instrument)
> context=fxs-line
> rxwink=300
> usecallerid=yes
> hidecallerid=no
> callwaiting=yes
> usecallingpres=yes
> threewaycalling=yes
> cancallforward=yes
> callreturn=yes
> echocancel=yes
> echocancelwhenbridged=yes
> rxgain=0.0
> txgain=0.0
> callgroup=1
> pickupgroup=1
> immediate=no
> signalling => fxo_ks
> channel => 1
>
> ; this section is for the fxo port (pstn line)
> context=incoming  ; < where to send incoming calls within
> extensions.conf rxwink=300
> usecallerid=yes
> hidecallerid=no
> callwaiting=yes
> usecallingpres=yes
> threewaycalling=yes
> cancallforward=yes
> callreturn=yes
> echocancel=yes
> echocancelwhenbridged=yes
> rxgain=0.0
> txgain=0.0
> callgroup=1
> pickupgroup=1
> immediate=no
> signalling => fxs_ks
> channel => 4
>
> Now if you look over the parameters within each section, you've got
> to ask yourself some questions.
> - Why would you have these parameters on a fxs port (telephone)?
> context=fxs-line
> rxwink=300
> usecallerid=yes
> cidsignalling=bell
> hidecallerid=no
> callwaiting=yes
> usecallingpres=yes
> threewaycalling=yes
> cancallforward=yes
> callreturn=yes
> The above parameters are associated with fxo ports (not fxs ports).
> Remove them.
>
> - Why do you have:
> rxwink=300
> in either of the two contexts above?  If you're interfacing to a US
> pstn line, I doubt you really need that parameter in either channel.
>
> - In the Channel => 4 section, you will probably want:
> echotraining=yes
>  or
> echotraining=800
> to improve the echo canceller operation. 800 works best for my system.
>
> - You'll probably want to add:
> busydetect=yes
> busycount=6
> in the top level [channels] section to help avoid voices falsely
> causing dropped pstn calls. (Some voices might be interpreted as
> dtmf or other tones causing * to drop the call.)
>
> - in the extensions.conf dialplan entry:
> [incoming]
> exten => s,1,Answer
> exten => s,2,DigitTimeout,5
> exten => s,3,ResponseTimeout,10
> exten => s,4,Playback(tt-weasels)
> you'll probably want to change that last statement to use 'Background'
> to play the audio instead of 'Playback'. See the 'show application
> background' for details. (Essentially, background allows you dialplan entry
> to listen for users to press a key, while the playback ignores user key
> presses.
>
> Rich
>
>
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Re: [Asterisk-Users] Is there a Caller ID issue in the latest CVSStable

2005-02-11 Thread Robert L Mathews

Nicol?s Gudi?o <[EMAIL PROTECTED]> wrote:
Paul, 1.0.5 stable suffers from caller id issues as well, at least for
SIP channels. What fixed things for me was swapping in app_dial.c from
1.0.2 stable (didn't try others). You could also just diff app_dial.c
between versions to find the problem but I took the lazy way out the
first time around.

Drumkilla reverted the callerid changes on the latest stable (thanks
Russell!). You will be fine if you checkout stable from CVS now.
Hmmm; I think I'm still having problems with it, using a completely 
fresh checkout and compile:
Connected to Asterisk CVS-v1-0-02/11/05-17:34:08

I have two Zap FXS lines and two SIP phones, and:
- Zap channel to Zap channel, caller ID works (displays correctly on the 
analog phone display).
- SIP phone to Zap channel, caller ID works.
- SIP phone to ZIP phone, caller ID does NOT work (Grandstream phone 
displays "Err").
- Zap channel to SIP phone, caller ID does NOT work.
- Incoming Free World Dialup calls to Zap channel extension, caller ID 
works.
- Incoming Free World Dialup calls to SIP phone extension, caller ID 
does NOT work.

So it seems that asterisk stable, as of today, does not send correct 
caller ID on calls that end up on SIP phones, unless I'm doing something 
boneheaded (although I used almost-identical config files on 1.0.2 with 
no trouble).

A tcpdump shows that asterisk is sending this in the SIP INVITE header 
to the phone:

From: "asterisk" ;
(IP address obscured; it's correct in the original.) But somehow 
"asterisk" appears instead of the correct caller ID. Wasn't that the bug 
other people were seeing that the stable update was supposed to fix? 
Have I missed something obvious?

--
Robert L Mathews, Tiger Technologies   http://www.tigertech.net/
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Re: [Asterisk-Users] dtmfmode and IAX protocol

2005-02-11 Thread Rich Adamson

> > > There's enough information if he's using FWD's 8XX-gateway for his 
> > > toll-free calls to UPS, the bank, etc.
> > 
> > No there isn't. We don't know how the Sipura is configured, we don't
> > know for sure what the associated sip def's are, we don't know exactly
> > what path was used to dial each call (eg, x100p, tdm, fwd), etc. All
> > we know is something doesn't work via fwd, but calls via something
> > to the bank and fedex work. We do know that fwd-iax wants g711, and if
> > that path is being used for all calls, you'd have to guess that
> > at least one or two dtmf digits would get through to the destination
> > and cause invalid number messages. Since he didn't mention that, its
> > highly likely his path to the bank and fedex are different then the
> > fwd path. 
> > 
> 
> For example all calls go through Sipura-3000
> If I connect to UPS over PSTN line on Sipura box DTMF is working just
> fine I can select any option from the menu choices.  
> 
> Though I think I'm being connected to two different call centers even
> though I call the same tall-free number.
> If I dial over PSTN line it has different recording then if I dial over
> Line1 (on Sipura-3000) and IAX to FWD, recording is different.
> 
> But again the same thing is with FedEx I'm connected to two different
> call centers I think, but everything works with FedEx.
> 
> Calling DHL over IAX and FWD, DTMF doesn't work either but going though
> PSTN line on Sipura-3000 works fine.

Joseph,
The level of detail provided in the above is much better compared to
your previous posts. Now we have a better clue what you're trying to
do.

There are two possibilities for why dtmf is not working the way you'd
like.
1. the dtmf definitions between the spa-3000 and asterisk are not in
   sync. (look at the /usr/src/asterisk/configs/sip.conf.sample file
   for dtmfmode, and try different modes. You might have to stop/start
   asterisk and the spa-3k to force a renotiated mode.)
2. FWD's system might not be configured to correctly handle dtmf over
   iax. (If that is the problem, its likely the issue is their definitions
   between asterisk and whatever they are using for equipment to terminate
   that call. Nothing you can do about that other then see if you can get
   Ed Guy at FWD to help diagnose the problem.)

If you'd care to pass along the 800 numbers for each of those companies
that you are dialing, I can try making the same calls from my system
since I have the same devices available here. That would at least allow 
me to run some of the debug and packet tracing tools to help identify 
the issue.

Rich


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Re: [Asterisk-Users] Asterisk won't answer incoming analog line

2005-02-11 Thread Rich Adamson

> I had to return my TDM11B because it put the PSTN line 'off hook' the moment 
> I 
> plugged it in and wouldn't hang it up.
> 
> The new card seems to work because I can actually make an outgoing call from  
> the FXO port to my cell phone, so I'm pretty happy about that.
> 
> But Asterisk doesn't recognize incoming calls from the PSTN.  If I dial my 
> home phone from my cell phone asterisk does not even recognize the ring.
> 
> Below is (I think) the relevant config files.  My question is:  How do I make 
> asterisk recognize when the phone is ringing?
> 
> Thanks in advance
> 
> lane
> 
> ;zaptel.conf
> fxsks=4
> fxoks=1
> loadzone = us
> defaultzone=us
> ;
> 
> ;zapata.conf
> 
> [channels]
> language=en
> rxwink=300
> usecallerid=yes
> cidsignalling=bell
> hidecallerid=no
> callwaiting=yes
> usecallingpres=yes
> threewaycalling=yes
> cancallforward=yes
> callreturn=yes
> echocancel=yes
> echocancelwhenbridged=yes
> rxgain=0.0
> txgain=0.0
> callgroup=1
> pickupgroup=1
> immediate=no
> signalling => fxo_ks
> channel => 1
> signalling => fxs_ks
> channel => 4
> 
> ; extensions.conf
> [globals]
> CONSOLE=Zap/1
> TRUNK=Zap/4
> TRUNKMSD=1
> 
> [trunklocal]
> exten => _9NXX,1,PlayBack(vm-dialout)
> exten => _9NXX,2,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> exten => _9NXX,3,Congestion
> 
> [incoming]
> exten => s,1,Answer
> exten => s,2,DigitTimeout,5
> exten => s,3,ResponseTimeout,10
> exten => s,4,Playback(tt-weasels)
> ;
> [local]
> ignorepat => 9
> include => trunklocal
> ;
> [default]
> include=incoming
> include=local

In the zapata.conf definitions for the two channels, do something like this:

immediate=no
signalling => fxo_ks
channel => 1
context=incoming; < where to send incoming calls within 
extensions.conf
signalling => fxs_ks
channel => 4

Add the above statement, and then your dialplan context 'incoming'
will be executed when receiving a call.

It might also help the learning process if you reorganize those
two channels so that you can see which parameters are being applied
to which channels. Something like this says exactly the same thing
as above:

[channels]
language=en
busydetect=yes
busycount=6
cidsignalling=bell
cidstart=ring
callerid=asreceived

; this section is for the fxs port (telephone instrument)
context=fxs-line
rxwink=300
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
threewaycalling=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
callgroup=1
pickupgroup=1
immediate=no
signalling => fxo_ks
channel => 1

; this section is for the fxo port (pstn line)
context=incoming; < where to send incoming calls within 
extensions.conf
rxwink=300
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
threewaycalling=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
callgroup=1
pickupgroup=1
immediate=no
signalling => fxs_ks
channel => 4

Now if you look over the parameters within each section, you've got
to ask yourself some questions.
- Why would you have these parameters on a fxs port (telephone)?
context=fxs-line
rxwink=300
usecallerid=yes
cidsignalling=bell
hidecallerid=no
callwaiting=yes
usecallingpres=yes
threewaycalling=yes
cancallforward=yes
callreturn=yes
The above parameters are associated with fxo ports (not fxs ports).
Remove them.

- Why do you have:
rxwink=300
in either of the two contexts above?  If you're interfacing to a US
pstn line, I doubt you really need that parameter in either channel.

- In the Channel => 4 section, you will probably want:
echotraining=yes
 or
echotraining=800
to improve the echo canceller operation. 800 works best for my system.

- You'll probably want to add:
busydetect=yes
busycount=6
in the top level [channels] section to help avoid voices falsely
causing dropped pstn calls. (Some voices might be interpreted as
dtmf or other tones causing * to drop the call.)

- in the extensions.conf dialplan entry:
[incoming]
exten => s,1,Answer
exten => s,2,DigitTimeout,5
exten => s,3,ResponseTimeout,10
exten => s,4,Playback(tt-weasels)
you'll probably want to change that last statement to use 'Background'
to play the audio instead of 'Playback'. See the 'show application background'
for details. (Essentially, background allows you dialplan entry to listen
for users to press a key, while the playback ignores user key presses.

Rich


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[Asterisk-Users] Q: Can Zap channels be arbitrarily numbered?

2005-02-11 Thread asterisk

Folks,

My production * box is equipped with a T100P card and a TDM400P
card, which gives me channels numbered 1 through 28. Is it possible to
assign arbitrary channel numbers so that, for example, the phone at
extension 2100 is connected to channel 2100? Such a scheme would make
selecting a channel for barge-in much easier . . .

Cheers,

Rob

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[Asterisk-Users] Asterisk won't answer incoming analog line

2005-02-11 Thread Lane

I had to return my TDM11B because it put the PSTN line 'off hook' the moment I 
plugged it in and wouldn't hang it up.

The new card seems to work because I can actually make an outgoing call from  
the FXO port to my cell phone, so I'm pretty happy about that.

But Asterisk doesn't recognize incoming calls from the PSTN.  If I dial my 
home phone from my cell phone asterisk does not even recognize the ring.

Below is (I think) the relevant config files.  My question is:  How do I make 
asterisk recognize when the phone is ringing?

Thanks in advance

lane

;zaptel.conf
fxsks=4
fxoks=1
loadzone = us
defaultzone=us
;

;zapata.conf

[channels]
language=en
rxwink=300
usecallerid=yes
cidsignalling=bell
hidecallerid=no
callwaiting=yes
usecallingpres=yes
threewaycalling=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
callgroup=1
pickupgroup=1
immediate=no
signalling => fxo_ks
channel => 1
signalling => fxs_ks
channel => 4

; extensions.conf
[globals]
CONSOLE=Zap/1
TRUNK=Zap/4
TRUNKMSD=1

[trunklocal]
exten => _9NXX,1,PlayBack(vm-dialout)
exten => _9NXX,2,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _9NXX,3,Congestion

[incoming]
exten => s,1,Answer
exten => s,2,DigitTimeout,5
exten => s,3,ResponseTimeout,10
exten => s,4,Playback(tt-weasels)
;
[local]
ignorepat => 9
include => trunklocal
;
[default]
include=incoming
include=local
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Re: [Asterisk-Users] Re: Codec Issue on IAX trunk? (Solved)

2005-02-11 Thread Rich Adamson

> > Personal opinion (and everyone has one) is the problem is a little
> > deeper then just a codec. Try 'iax2 debug', place a call, and look
> > through the early part of that trace. Are IP's right, etc?
> >
> > I believe the native 0 is simply suggesting there is no match when
> > negotiating the codec. The "4" is ulaw (show codecs).
> >
> > If you can, post the appropriate iax contexts from both sides (change
> > the passwords) and maybe some of the debug. (Or better yet for now,
> > email that to me directly and save the list a little bandwidth.)
> 
> 
> I believe you are right that there was a deeper problem.  
> Unfortunately, I don't know what it was, but at least things are 
> working right now.
> 
> I got really suspicious when the iax2 debug showed nothing!  I 
> screwed around with the config a little more and found that the error 
> message I got was really only coming from one side, and not the other.  
> I restarted asterisk, no dice.  I stopped asterisk, recompiled from 
> exactly the same source, started up again, and all is well.  Don't know 
> what's behind that.  Hmm.
> 
> Anyhoo, thanks for the user/peer setup and your time!

Those type changes to iax.conf require a full stop of asterisk followed
by a cold asterisk startup. A restart from the CLI won't cut it.

Were you previously doing the CLI restart?

Rich


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[Asterisk-Users] i want to load chan_h323.so

2005-02-11 Thread ???





I use Fedora core 2, and openssl-0.9.7, expat-1.95.7 is installed by rpm packages.

I downloaded pwlib-1.5.2 and openh323-1.12.2 at my home directory(/root/root_src), asterisk 1.0.4 at directory /usr/src/ and have installed successfully.

Asterisk is executed normally, but module chan_h323.so cannot be loaded.

The message is :

# asterisk ?vvvgc
.
.some message
.
Asterisk Ready.
*CLI> load chan_h323.so
/root/root_src/openh323/lib/libh323_linux_x86_r.so.1.12.2: undefined symbol: _Z13vpb_dial_synciPc
Unable to load module chan_h323.so
*CLI>
 

Please give me your solutions. Thank you for your reading.

My install log is :

# tar xvfz pwlib-1.5.2.tar.gz
# tar xvfz openh323-1.12.2.tar.gz
# cd /root/root_src/pwlib
# ./configure
# make
# cd /root/root_src/openh323
# ./configure
# make opt
# cd /usr/src 
# export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot 
# cvs login
# cvs co -r v1-0 asterisk 
# echo $PWLIBDIR
/root/root_src/pwlib
# echo $OPENH323DIR
/root/root_src/openh323
# echo $LD_LIBRARY_PATH
/root/root_src/pwlib/lib:/root/root_src/openh323/lib
# cd /usr/src/asterisk/channels/h323
# make
# cd /usr/src/asterisk
# make install


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Re: [Asterisk-Users] Asterisk@home .05 release questions on setup.

2005-02-11 Thread [EMAIL PROTECTED]

1. There is a help file you can run from the Linux
command line help-aah. This will tell you how to
change the passwords. On a clean install it tells you
this in the motd.

2. Not sure about this second one. I made some big
changes in asterisk for this release. It now runs as
asterisk not as root and it uses amportal to start not
the startup files in /etc/init.d I think only a clean
install will fix this.

3. A lot of changes in FOP too the config files are in
a different place could cause this problem.

Sorry about all the changes. As we get closer to a 1.0
release of [EMAIL PROTECTED] a lot of this will stabilize.

--- Ariel Batista <[EMAIL PROTECTED]> wrote:

> Hello,
> 
> Great job on the [EMAIL PROTECTED] project. Looks great
> this new version is really nicer looking.  But I
> have a few questions.
> 
> 1) For the new web access http://localIP/maint how
> and where do I change the password.
> 2) Since I don't use the Amp section for setup the
> .conf files I use my own. How do I get the asterisk
> server running status up.  I have it running and
> works but shows up as not running on the web page.
> 3) I upgraded my system from the older .04 by
> downloading the new tar and running your script.
> Then I copied my .conf files back and rebooted. I
> had already changed my password and logins names
> before this.  Asterisk is up and running without any
> issue's. But the Flash Operator panel comes up
> flashing and I can't seem to get it to work.
> 
> I feel you have done a great job and I would like to
> thank you for your setup to us.  I will be sending
> you a donation soon. I am at a small self employed
> computer consultant that has limited funds at
> present.  This is one of the best setups for
> Asterisk that I have seen. I feel your name does not
> do it right due to it can be used for SOHO's and
> other setups.  It's great keep up the good work. You
> actually make AMP work.
> 
> P.S. one more question do you have an area in the
> freenode for chat? If you don't I would love to help
> out in it.  Something like Asterisk-athome would be
> nice.
> 
> Ariel
> 
> 
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Re: [Asterisk-Users] Asterisk@Home 0.5 Released today

2005-02-11 Thread [EMAIL PROTECTED]

not really. you could try to use the tarball install
over what you had but it will wipe out your config
files. In the futurte we will try to be better about
an upgrade path.

--- Ariel Batista <[EMAIL PROTECTED]> wrote:

> [EMAIL PROTECTED] wrote:
> > We are releasing a new version of our one-button
> > Asterisk install, [EMAIL PROTECTED], today. This
> release
> > includes a redesigned web interface and
> auto-detection
> > of Digium fxo and fxs cards. We have also fixed a
> lot
> > of bugs and added numerous customer requested
> > enhancements. [EMAIL PROTECTED] is now more secure
> with
> > passwords on the web pages and better Linux
> security.
> 
> Is there an upgrade for current users of .04?
> 
> > http://asteriskathome.sourceforge.net/
> > 
> > 
> > 
> > 
> > __
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Re: [Asterisk-Users] Asterisk@home .05 release questions on setup.

2005-02-11 Thread Ariel Batista

Thank you that worked for the passwords..
Thanks

- Original Message -
From:
dean
collins
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Friday, February 11, 2005 10:06
PM
Subject: RE: [Asterisk-Users] [EMAIL PROTECTED] .05 release questions on
setup.

If you log in via
cosole and type in the help-aah command it brings up the new security
requirements for changing passwords
I cant help you with
the rest as I installed from scratch using the
iso.

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ariel BatistaSent: Friday, February 11, 2005 7:55
PMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] [EMAIL PROTECTED]
.05 release questions on setup.

Hello,

Great job on the [EMAIL PROTECTED] project. Looks great this new
version is really nicer looking. But I have a few
questions.

1) For the new web access http://localIP/maint how and where do I
change the password.

2) Since I don't use the Amp
section for setup the conf files I use my own. How do I get the asterisk
server running status up. I have it running and works but shows up as
not running on the web page.

3) I upgraded my system from the
older 04 by downloading the new tar and running your script. Then I
copied my conf files back and rebooted. I had already changed my password and
logins names before this. Asterisk is up and running without any
issue's. But the Flash Operator panel comes up flashing and I can't seem to
get it to work.

I feel you have done a great job
and I would like to thank you for your setup to us. I will be sending
you a donation soon. I am at a small self employed computer consultant that
has limited funds at present. This is one of the best setups for
Asterisk that I have seen. I feel your name does not do it right due to it can
be used for SOHO's and other setups.
It's great keep up the good work. You actually make AMP
work.

P.S. one more question do you have
an area in the freenode for chat? If you don't I would love to help out in
it. Something like Asterisk-athome would be
nice.

Ariel

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