[Asterisk-Users] dreaded Caller*ID failed checksum

[Jeff Gustafson]

Caller*ID used to work as some point, but I can't seem to get it going
these days.  The card is a x101p.  I've tried going up and down the
rxgain scale.  Can the txgain effect it at all?  When I plug in a phone
into the line with a splitter it can decode caller id with no problems.
Reading through the mailing list archives hasn't given me any
move clues.  Any ideas?

...Jeff

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[Asterisk-Users] Channelized T1, T100P problems

[Pat Boyle]

I've been having some problems getting a 
channelized T1 working with a T100P card. Perhaps someone can help:


I have an Eschelon T1 coming into a Vina Integrator 
box. This box splits out the T1 into an ethernet plug for bandwidth and a 
secondary T1 which I plug into the T100P card.I've connected the two with 
a T1 crossover cable. 

I get a green light on the t100p card and can make 
outbound calls. Inbound calls are not working as well. Via the 
console, I can see them come in and start the "s" extension, but then they hang 
up. Then the second channel accepts a call, then I get a "all 
circuits busy" message on the phone (from my cell as I call in to test). 
I'm testing from my cell phone and never hear a ring or any of the asterisk 
welcome message (even though it's showing in the console). It's as if the call 
appears to be answered by the card, but actually never is.

Changing the "immediate=yes" to "no" changes 
behavior somewhat. Then I get a message in the console that there is no 
extension "6" defined, but if I define it, the problem doesn't go 
away.

I am using a channelized T1, not PRI. So we 
don't have b and d channels. Just em signalling. Thanks for 
your help.
Pat Boyle

-
System:
Eschelon T1, channelized (8 channels and the rest 
for bandwidth)
Dell Poweredge
2.4 Celeron
Redhat Fedora
Asterisk -0.7.2
Zaptel drivers 0.9.0

Here are the config files and console 
output:
---
cat /etc/zaptel.confspan=1,1,0,esf,b8zsem=1-8loadzone 
= usdefaultzone=us
cat 
/etc/asterisk/zapata.conf[channels]context=incomingsignalling=em_wgroup=1immediate=yeschannel 
= 1-8


extensions.conf
;the menu[incoming]exten = s,1,noop; putting a wait here 
before would cause hangup issues if the person; hung up right after the 
phone rang. It seems that the card picks; up (simple switch) but 
doesn't answer until the answer command.; If you hang up before the answer 
command is executed, there are hangup issues.exten = s,2,Answerexten 
= s,3,DigitTimeout(5) ; timout for dtmf entries, 5 is 
defaultexten = s,4,ResponseTimeout(30); hangup after 30 sec of 
nothingexten = s,5,Background(hello-recording);welcome... then wait for 
response or timeout


; press 1 for directory; the argument should be the context specified 
in voicemail.confexten = 1,1,Directory(default)

; if invalid extention, then play invalid message and goto message 
againexten = i,1,Playback(pbx-invalid)exten = 
i,2,Goto(s,5)

; on timout hangupexten = t,1,Playback(goodbye)exten = 
t,2,Hangup

exten = h,1,Hangup


-- from the console ---

*CLI -- Starting simple switch on 
'Zap/1-1' -- Set Digit Timeout to 5 
-- Set Response Timeout to 30 -- Playing 'hello-recording' 
(language 'en')Urgent handler -- Starting simple 
switch on 'Zap/2-1' -- Set Digit Timeout to 
5 -- Set Response Timeout to 30 -- 
Playing 'hello-recording' (language 'en')Urgent 
handler -- Hungup 'Zap/1-1'Urgent handlerUrgent 
handler -- Hungup 'Zap/2-1'Urgent handlerUrgent 
handler

[Asterisk-Users] Re: indications.conf settings for spain

[Antonio Diego Almodóvar Cebrián]

Saludos.
Encantado de ver a alguien español (o no?), en el tiempo que llevo en la lista todavía no lo había visto, o no me había fijado. :)
Googlea lo siguiente:
"site:lists.digium.com indications conf spain"Sin las comillas claro :)
Do You Yahoo!?

Todo lo que quieres saber de Estados Unidos, América Latina y el resto del Mundo.
Visíta Yahoo! Noticias.

[Asterisk-Users] Fritz ISDN PCI v2 and CAPI

[jean-marie . goupil]

Hi,
First, here is my config: Kernel version 2.4.25 on a Fedora distro,
Asterisk and a Fritz! Isdn PCI Card (v2). I try to make the CAPI drivers
deal with Asterisk but I can't try to figure out to get of this issue.
As I see, Fritz modules are integrated with the kernel, so I directly
loaded the 'hisax_fcpcipnp' module from it. I install also Capi modules by
downloading archives of the web (make config - make install - insmod...).

When i check loaded modules:
# lsmod

Module  Size  Used byNot tainted
capi   20864   0
capifs  5424   1  [capi]
kernelcapi 34048   1  [capi]
capiutil   24864   0  [kernelcapi]
i810_audio 25244   1  (autoclean)
ac97_codec 17236   0  (autoclean) [i810_audio]
e100   55496   1
hisax_fcpcipnp  7968   0
hisax_isac  7800   0  [hisax_fcpcipnp]
hisax 131820   0  [hisax_fcpcipnp hisax_isac]
isdn  139744   0  [hisax]
slhc6740   0  [isdn]
mousedev5524   0  (unused)
keybdev 3140   0  (unused)
input   5728   0  [mousedev keybdev]
ext3   72896   2
jbd61996   2  [ext3]

I've edited /etc/capi.conf and write 'hisax_fcpcipnp - - - - - -' in it.
Anyway, 'capiinit' failed (and so for 'capiinfo'):
#  capiinit
ERROR: failed to load driver hisax_fcpcipnp
# capiinfo
capi not installed - No such device or address (6)

Please, could you explain me how to make CAPI drivers speak with the
hisax_fcpcipnp module ?

Cheers.


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Re: [Asterisk-Users] How to use ZapHFC ?

[Martin Schenkelberg]

Thank you, it is working.

But now it does not wait till i entered the whole number für outbound dialing, 
there may be an error in extensions.conf

Events from internal S0 are send to [default]

All configured extensions work except the outbound dial one (Numbers starting 
with 9 should be dialed over Capi)  After the second digit * starts Capi 
dialing. How can i fix this.

Thank You, Martin

PART OF EXTENSIONS.CONF

[macro-stdexten]
 ;
 ; Standard extension macro (with call forwarding):
 ; ${ARG1} - Extension(we could have used ${MACRO_EXTEN} here as well
 ; ${ARG2} - Device(s) to ring
 ;
 exten=s,1,DBget(temp=CFIM/${ARG1}) ; Get CFIM key, if not existing, goto 102
 exten=s,2,Dial(Local/[EMAIL PROTECTED]/n)   ; Unconditional forward
 exten=s,3,Dial(${ARG2},10,tr) ; 20sec timeout
 exten=s,4,DBget(temp=CFBS/${ARG1})  ; Get CFBS key, if not existing, goto 105
 exten=s,5,Dial(Local/[EMAIL PROTECTED]/n) ; Forward on busy or unavailable
  ; No CFIM key
 exten=s,102,Goto(s,3)
  ; No CFBS key - voicemail ?
 exten=s,105,NoOp


[default]

;  SIP PHONES #
exten = 792,1,Dial(SIP/792,${VBOXTIME},tr)
exten = 792,2,VoiceMail,u792
exten = 792,102,VoiceMail,b792

exten = 793,1,Dial(SIP/793,${VBOXTIME},tr)
exten = 793,2,VoiceMail,u793
exten = 793,102,VoiceMail,b793

;exten = 794,1,Dial(SIP/794,${VBOXTIME},tr)
;exten = 794,2,Dial(Zap/g2/1000,${VBOXTIME},tr)
exten = 794,1,Macro(stdexten,794,SIP/794)
exten = 794,2,VoiceMail,u794
exten = 794,102,VoiceMail,b794

;  ISDN PHONES 
exten = 1000,1,Macro(stdexten,1000,Zap/g2/1000)
exten = 2000,1,Macro(stdexten,2000,Zap/g2/2000)

;  FEATURES ###
 ; Unconditional Call Forward
 exten = _*21*X.,1,DBput(CFIM/${CALLERIDNUM}=${EXTEN:4})
 exten = _*21*X.,2,Hangup
 exten = #21#,1,DBdel(CFIM/${CALLERIDNUM})
 exten = #21#,2,Hangup

 ; Call Forward on Busy or Unavailable
 exten = _*61*X.,1,DBput(CFBS/${CALLERIDNUM}=${EXTEN:4})
 exten = _*61*X.,2,Hangup
 exten = #61#,1,DBdel(CFBS/${CALLERIDNUM})
 exten = #61#,2,Hangup

; Or a conference room (you'll need to edit meetme.conf to enable this room)
;
exten = ,1,Meetme,

;Mailbox

exten = ,1,Ringing
exten = ,2,Wait(2)
exten = ,3,VoicemailMain
exten = ,4,Wait(1)
exten = ,5,Hangup

;externalcalling
exten = _9.,1,StripMSD,1
exten = _.,2,Dial,CAPI/${CALLERIDNUM}:bBYEXTENSION

---

Am Mittwoch, 7. April 2004 00:26 schrieb Jan Baumann:
 Martin Schenkelberg wrote:
  My problem is that if i take off the handset the Asterisk starts dialing
  without presenting a dialtone. It dials an empty string wich could not
  work.
 
  If i type in the phone number first an then take off the handset the call
  works fine.
 
  How can i change this  i want asterisk to present a dialtone to the isdn
  phones first. After a Timeout asterisk should hang up.

 Hi Martin,

 overlap dial is what you are looking for.

 Try 'overlapdial=yes' in the channel definition in zapata.conf.


 Jan Baumann
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Re: [Asterisk-Users] quadBRI and UK ISDN2e

[Jon Fautley]

Stephen Karrington wrote:
Which brand of card did you get?
The Junghanns.net quadBRI PCI Card.

Just been back through BT order processing and told them to put Caller 
Display (as they call it) on the line, which they said they've done... 
getting fairly certain it's not a BT issue now :|

Any help greatly appreciated,

Thanks,

Jon
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Re: [Asterisk-Users] Callerid + Zaphfc

[Martin Schenkelberg]

Thank you problem solved.

I tried to use the (R) Button on my phone to place call on HOLD but Asterisk 
says something of PRI Error : Dont know how to post-handle message of Tye 
HOLD (36)

Is this feature not implemented in Bri-Stuff ?

Thanks again

Am Mittwoch, 7. April 2004 16:54 schrieb Klaus-Peter Junghanns:
 Hi,

 use prilocaldialplan=local in zapata.conf.

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Re: [Asterisk-Users] Fritz ISDN PCI v2 and CAPI

[Jon Fautley]

[EMAIL PROTECTED] wrote:
Hi,
First, here is my config: Kernel version 2.4.25 on a Fedora distro,
Asterisk and a Fritz! Isdn PCI Card (v2). 
[SNIP]
I've edited /etc/capi.conf and write 'hisax_fcpcipnp - - - - - -' in it.
change this to:

fcpci - - - - - -

without the quotes, and all should be well.

HTH,

Jon
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Re: [Asterisk-Users] New Zealand indications.conf

[Vic Cross]

On Tue, 6 Apr 2004, Matt Riddell wrote:

 Here are the settings for New Zealand indications.  I have tested them and
 call progress works...voicemail messages used to contain 50 seconds of
 disconnect tones, now just 2.

snipped the detail

So, all you did was update indications.conf with what you posted, and 
everything worked?  Wow...

Wait a minute...  What kind of hardware are you using?

I am fighting with making the Zap stuff recognise and generate proper
tones for AU (on my X100P and TDM cards).  Just updating indications.conf
does not work for me -- the simple switch generates different tones that
are unrelated to what I've coded there (I've tested this with an extension
that runs a bunch of PlayTones() apps -- PlayTones is correct, but the
simple switch does its own thing).  As for analogue call progress, forget
it -- having read the code, I cannot see how it could work at all on any
service that does not present US tones.

(Digium et.al. -- please don't take this as criticism.  You guys have to
scratch the biggest and most annoying itches first!  I wish I had the time
and skill to contribute detection routines for other areas.  Skill is the
main problem for me, since progress tones in the US, based on MF tones as
they are, are much easier to code to recognise than AU tones which are all
different cadences - and in some cases, amplitudes - of the same single
frequency tone.)

Matt, good for you!

Cheers,
Vic Cross
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[Asterisk-Users] GUI?

[Altus Snyman]

Good day all
I'm looking for a GUI/Web interface for Asterisk.
What I need it for is to see who's line(SIP) is busy work?
Something like a switch board?
Please give me some info?
Thanks

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Re: [Asterisk-Users] GUI?

[Michiel Betel]

Altus Snyman wrote:

Good day all
I'm looking for a GUI/Web interface for Asterisk.
What I need it for is to see who's line(SIP) is busy work?
Something like a switch board?
Please give me some info?
Thanks
 

http://www.voip-info.org/wiki-Asterisk+GUI
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RE: [Asterisk-Users] res_motv: Request for Comment

[Andy Powell]

Just curious, but why does it strike you as such a bad idea?  Especially if
it was disabled by default.  I can understand you not wanting your system
security or your personal privacy compromised, but I think it would be
great to have it in place for:

A) Manual activation for those who want automated updates.
B) CLI execution for occasional comparison to the current set.

As a completely seperate application I'd probably not object (like zttool), I just 
don't think that this should be any part of *. Bearing in mind the number of outages 
that occur, the number of potential failures when connecting via the net. It's not 
just a privacy issue it's a functionality issue... if I manage 20 servers (same 
version)  I don't need all 20 of them to say what are the fixes for this .. I just 
want 1, and I want to do it manually... the other thing is that there isn't an easy 
way to check what version you are running, after updating from cvs the version doesn't 
always change (annoying at the best of times)...

...but the core of my 'problem' is software that calls home.


Perhaps it should be possible to flag the request with a token indicating
that you don't want to be part of the survey, and you don't want your
IP/host information stored.  A --anon option, if you will.

I would expect the --anon option to be the default, any communication to be encrypted, 
and the db secure...


 I can only wait until we see M$ like activation implemented... oh the
 joy...

I am going to guess that you're joking.  I just don't see that happening.
Mark and the team at Digium seem dedicated to open source and to the
Asterisk community.  His asking for comments on this idea is a pretty good
indicator of his concern for the community's opinions.

I was being facecious



 It would be much better just to have the information present on either
the
 Digium site or some other location. I see little point in wasting your
 valuable time doing something like this when there are so many
outstanding issues and feature requests that could offer more.

Perhaps Mark's time could be spent on other things, but I would still like
to see Digium offer this option -- perhaps one of the other developers
could head up the effort?

True, but just take a look at the bug tracker, feature requests are getting rejected 
because we don't have time, we're too busy fixing bugs


Just putting the current version information up on a web page is nice, but
it doesn't allow me to automatically query the system and discover known
issues and possible solutions.  I think that, for service providers that
could very well be a critical next step.  Several of my clients made the
decision to go with RedHat based on their update service (for which they
gladly paid).


I was actually thinking more along the lines of enter your version details on a web 
form and you get the updates and fixes info based on that, it's then only 2 minutes 
coding away for a simple perl (or other) app to automagically retrieve the info if you 
require. This makes it purely voluntary and you have to actively request the call home 
feature, saving us from programmer coding errors which could accidentally enable it by 
default during bug fixes etc.



Andy


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[Asterisk-Users] error compiling cdr_mysql support

[Alessio Focardi]

Here is the error I get compiling the asterisk-addons rpm

cc -fPIC -I../asterisk  -I/usr/include/mysql -c -o cdr_addon_mysql.o 
cdr_addon_mysql.c
cdr_addon_mysql.c:50: error: `PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP' undeclared here 
(not in a function)
make: *** [cdr_addon_mysql.o] Error 1

I'm pretty sure I got all that is needed:

# rpm -qa | grep mysql

mysql-3.23.58-4
mod_auth_mysql-20030510-3
php-mysql-4.3.3-6
mysql-server-3.23.58-4
mysql-devel-3.23.58-4
ser-mysql-0.8.12-0


Any idea ?

Tnx !




-- 
Best regards,
 Alessio  mailto:[EMAIL PROTECTED]


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[Asterisk-Users] Réf. : Re: [Asterisk-Users] Fritz ISDN PCI v2 and CAPI

[jean-marie . goupil]

If I do that, the modprobe doesn't find any module called fcpci and is
looking for any module called hisax_fcpcipnp as it's the one install for
the isdn fritz card.

# capiinit
modprobe: Can't locate module fcpci
ERROR: failed to load driver fcpci

Maybe, I should reinstall the fritz card drivers itselves? (however, I
don't think so...)


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Re: [Asterisk-Users] Fritz ISDN PCI v2 and CAPI

[Jakob Strebel]

Jean-Marie,


Hi,
First, here is my config: Kernel version 2.4.25 on a Fedora distro,
Asterisk and a Fritz! Isdn PCI Card (v2). I try to make the CAPI drivers
deal with Asterisk but I can't try to figure out to get of this issue.
As I see, Fritz modules are integrated with the kernel, so I directly
loaded the 'hisax_fcpcipnp' module from it. I install also Capi modules by
downloading archives of the web (make config - make install - insmod...).
It is my understanding not to use hisax. Use chan_capi instead.
http://www.voip-info.org/tiki-index.php?page=Asterisk+How+to+connect+with+CAPI
http://www.junghanns.net/asterisk/page1.html

I have this working.

jakob 

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[Asterisk-Users] DIAX and CallMe feature

[Dan]

Hi all,

Because I have changed the ISP, the CallMe feature in DIAX is not functional
for the moment.
I will release a new version with the required changes ASAP.
The e-mail address [EMAIL PROTECTED] will still be available for a max. of 2
months.
The new e-mail address for support requests regarding DIAX is the following:
[EMAIL PROTECTED]

Thank you for your understanding,
Dan



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Re: [Asterisk-Users] quadBRI and UK ISDN2e

[Linus Surguy]

 Just been back through BT order processing and told them to put Caller
 Display (as they call it) on the line, which they said they've done...
 getting fairly certain it's not a BT issue now :|

It might not be a BT issue, but BT *dont* call it 'Caller Display' on ISDN
lines, you want 'ISDN 2e Calling Line Identity Presentation'

See:

http://www.serviceview.bt.com/list/current/docs/Exch_Lines.boo/001319.htm

for pricing info.

Linus



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[Asterisk-Users] WAN Side Calling

[Paul Tyreman]

Hi, I am new to Asterisk, but I have to say 
it's an amazing peice of software. What I want to do is to be able to 
call other members of my family over our broadband conenctions (so that we don't 
have to pay huge phone bills anymore). We are both behind standard 
(NOT SIP aware) routers, so I would like 
to know if this is possible ? I want people external to my LAN to be 
able to use my server, as I want to specify the number they get. Is this 
possible and if so, what ports do I need to open at (1) The Server Side and (2) 
The Client Sides ? Thanks in advance. 
Paul.

Re: [Asterisk-Users] res_motv: Request for Comment

[Duane]

Andy Powell wrote:
...but the core of my 'problem' is software that calls home. 
I agree a separate tool is possibly the best option here for privacy 
reasons...

Simple solution seems to me, have a version.h file in each module and 
rather then calling home publish the version info that is downloaded 
rather then uploaded with some sort of changelog type system, and the 
app running locally will detail on request the changes between the 
different versions...

obviously gzip'ing the changelog will save download times...

I think it's a good idea if implemented correctly, everything off by 
default, and give good details on what it does specifically and how to 
enable it and that should keep everyone happy... Those that are 
especially paranoid, just don't compile the version checking tool, so if 
anyone complains later not only did they have to run the util, but they 
had to compile it as well...

--
Best regards,
 Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com - Telecommunications Freedom
http://happysnapper.com.au - Sell your photos over the net!
http://e164.org - Using Enum.164 to interconnect asterisk servers
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[Asterisk-Users] Réf. : Re: [Asterisk-Users] Fritz ISDN PCI v2 and CAPI

[jean-marie . goupil]

I tried that but it still doesn't work... I think I don't have the correct
approach. Have I to install any ISDN drivers (=modules ?) BEFORE dealing
with CAPI ?

If yes, why shouldn't I use the hisax drivers (which are kernel ones)
instead of fcpci drivers (which doesn't seems to work, by the way...)

And finally, how is it possible to link the two modules together?

As you can see, I am drowning in these difficulties and a help would be
very welcomed!

Cheers!


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RE: [Asterisk-Users] ANNOUNCE: Flash Operator Panel

[Joe Dennick]

I'm still having problems being able to get the Transfer function to
work.  I enter the correct password, but still can transfer or end calls
with the Flash Panel.  Any suggestions?

Joe

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gary
Sent: Wednesday, April 07, 2004 10:11 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] ANNOUNCE: Flash Operator Panel


On Fri, 2 Apr 2004 09:32:34 -0500, Adams, Gavin wrote:

We run at 1600x1200, 96 buttons would be useful.

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users- 
 [EMAIL PROTECTED] On Behalf Of Nicolas Gudino
 Sent: Friday, April 02, 2004 9:26 AM
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] ANNOUNCE: Flash Operator Panel
 
 Hi Eric,
 
 - Original Message -
 From: Eric Wieling [EMAIL PROTECTED]
 Sent: Friday, April 02, 2004 11:17 AM
 Subject: Re: [Asterisk-Users] ANNOUNCE: Flash Operator Panel
 
 
  Being able to have more buttons as well as changing the button size

  would be useful.
 
 What screen resolutions do you use, how many buttons do you need?
 

Gee, dual screens, high res
Could be great system for hotel etc
.



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RE: [Asterisk-Users] ANNOUNCE: Flash Operator Panel

[Altus Snyman]

please let me know if anyone get this..please

On Thu, 2004-04-08 at 13:21, Joe Dennick wrote:
 I'm still having problems being able to get the Transfer function to
 work.  I enter the correct password, but still can transfer or end calls
 with the Flash Panel.  Any suggestions?
 
 Joe
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Gary
 Sent: Wednesday, April 07, 2004 10:11 PM
 To: [EMAIL PROTECTED]
 Subject: RE: [Asterisk-Users] ANNOUNCE: Flash Operator Panel
 
 
 On Fri, 2 Apr 2004 09:32:34 -0500, Adams, Gavin wrote:
 
 We run at 1600x1200, 96 buttons would be useful.
 
  -Original Message-
  From: [EMAIL PROTECTED] [mailto:asterisk-users- 
  [EMAIL PROTECTED] On Behalf Of Nicolas Gudino
  Sent: Friday, April 02, 2004 9:26 AM
  To: [EMAIL PROTECTED]
  Subject: Re: [Asterisk-Users] ANNOUNCE: Flash Operator Panel
  
  Hi Eric,
  
  - Original Message -
  From: Eric Wieling [EMAIL PROTECTED]
  Sent: Friday, April 02, 2004 11:17 AM
  Subject: Re: [Asterisk-Users] ANNOUNCE: Flash Operator Panel
  
  
   Being able to have more buttons as well as changing the button size
 
   would be useful.
  
  What screen resolutions do you use, how many buttons do you need?
  
 
 Gee, dual screens, high res
 Could be great system for hotel etc
 .
 
 
 
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[Asterisk-Users] Re: [Asterisk-Users] Réf. : Re: [ Asterisk-Users] Fritz ISDN PCI v2 and CAPI

[Jakob Strebel]

Jean-Marie,


I tried that but it still doesn't work... I think I don't have the correct
approach. Have I to install any ISDN drivers (=modules ?) BEFORE dealing
with CAPI ?
If yes, why shouldn't I use the hisax drivers (which are kernel ones)
instead of fcpci drivers (which doesn't seems to work, by the way...)
I remember that I read (found trough Google) that hisax and channel_capi 
can not be installed at the same time. Actually you have to remove hisax to 
run chan_capi

jakob

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RE: [Asterisk-Users] Fritz ISDN PCI v2 and CAPI

[Brian Cuthie]

Can this Frtiz card be used in the US?

-brian 

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Jakob Strebel
 Sent: Thursday, April 08, 2004 3:17 AM
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] Fritz ISDN PCI v2 and CAPI
 
 Jean-Marie,
 
 
 Hi,
 First, here is my config: Kernel version 2.4.25 on a Fedora distro, 
 Asterisk and a Fritz! Isdn PCI Card (v2). I try to make the CAPI 
 drivers deal with Asterisk but I can't try to figure out to 
 get of this issue.
 As I see, Fritz modules are integrated with the kernel, so I 
 directly 
 loaded the 'hisax_fcpcipnp' module from it. I install also 
 Capi modules 
 by downloading archives of the web (make config - make 
 install - insmod...).
 
 It is my understanding not to use hisax. Use chan_capi instead.
 http://www.voip-info.org/tiki-index.php?page=Asterisk+How+to+c
onnect+with+CAPI
 
 http://www.junghanns.net/asterisk/page1.html
 
 I have this working.
 
 jakob 
 
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Re: [Asterisk-Users] Out of trunk data space on call number 16386, dropping

[Warren H. Prince]

I work with Tony, so I'm responding for him. Yes, it appears only
during a conference call. So, if we disable conferencing, we do not
receive the error.

Justin Carlson wrote:

  if you disable conferencing does the problem go away?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]]On Behalf Of Tony Buser
Sent: Wednesday, April 07, 2004 2:30 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Out of trunk data space on call number
16386, dropping


I'm having the same kind of issues.  We get the out of trunk data space
error consistently during conference calls between asterisk servers.
And occasionally on regular iax calls.  Also while we're on a conference
call it seems to cause other calls going out through iax to fail and
also give this error.  (weather its to another asterisk server or
through say oneunified)

If you figure this out, please let us know here.  I'm pretty much at a
loss as to what could be causing it.

Justin Carlson wrote:

  
  
	Hi all,

We keep getting these and all the calls between these two asterisk boxes

  
  get
  
  
dropped.  what is going on here, I have been trying to solve this problem

  
  on
  
  
my own but maybe I don't have the trunk setup right.  also I have posed

  
  the
  
  
output of my full log of the machine with the zap interface, the other is
using ztdummy.

  
  

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Re: [Asterisk-Users] quadBRI and UK ISDN2e

[stan]

On Wed, Apr 07, 2004 at 12:37:43PM +0100, Jon Fautley wrote:
 Morning Asterikians,
 
 I've just got my nice shiny quadBRI card, and it seems to be working 
 very well - except for one little issue - CallerID.
 
 The card is currently connected to an ISDN2e line in P2P mode, and an S0 
 adapter on our existing alcatel PBX.

Is this an omnipcx?

 The S0 connection recieves callerID 
 and displays it correctly - the 2e line doesn't, and BT have said that 
 CLID was enabled on the line two days ago. Does anyone have any pointers 
 on this?

I assume the callerid is also being displayed on the alcatel handsets?
or is this just callerid generated on internal calls?

If it is then that should show the bt line is setup correctly.  I have
callerid working from a bt 2e line in ptp mode using zaphfc.  So
assuming the bri-stuff versions match the only difference would be the
quadbri card.  Not sure where that leads because I think/thought all the
callerid stuff would be handled by libpri from a q.931 SETUP message on
the d-channel and not be driver/card specific.

I could only get my s0 box to operate in ptmp mode, so there would be
change in the signalling line in zapata.conf, but then if you weren't
changing that nothing tends to work rather than just callerid.  I also
note that you have to match the msn assigned to the s0 when dialling out
through the omnipcx whereas bt doesn't seem to be as fussy, but again
nothing todo with callerid.

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RE: [Asterisk-Users] Out of trunk data space on call number 16386, dropping

[Justin Carlson]

how 
did you guys go about diableing it. Is it the threwaycalling directive in 
zapata.conf ?

  -Original Message-From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED]On Behalf Of Warren H. 
  PrinceSent: Thursday, April 08, 2004 8:01 AMTo: 
  [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] Out of 
  trunk data space on call number 16386, droppingI work 
  with Tony, so I'm responding for him. Yes, it appears only during a 
  conference call. So, if we disable conferencing, we do not receive the 
  error.Justin Carlson wrote: 
  if you disable conferencing does the problem go away?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]]On Behalf Of Tony Buser
Sent: Wednesday, April 07, 2004 2:30 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Out of trunk data space on call number
16386, dropping


I'm having the same kind of issues.  We get the out of trunk data space
error consistently during conference calls between asterisk servers.
And occasionally on regular iax calls.  Also while we're on a conference
call it seems to cause other calls going out through iax to fail and
also give this error.  (weather its to another asterisk server or
through say oneunified)

If you figure this out, please let us know here.  I'm pretty much at a
loss as to what could be causing it.

Justin Carlson wrote:

  
	Hi all,

We keep getting these and all the calls between these two asterisk boxes
get
  
dropped.  what is going on here, I have been trying to solve this problem
on
  
my own but maybe I don't have the trunk setup right.  also I have posed
the
  
output of my full log of the machine with the zap interface, the other is
using ztdummy.


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[Asterisk-Users] Caller ID on TDM400P Quad FXS

[Daniel ANDRE]

Hello,

I have a quad FXS TDM400P and it works fine with my asterisk 
configuration. I wonder to know if there is any configuration option so 
that Caler ID information should be properly sent by the TDM400 to the 
phone connected to it.

Best Regards,

Daniel

--
Daniel ANDRE (mailto:[EMAIL PROTECTED])
IRIS Technologies - http://www.iris-tech.com
Serveur kwartz - http://www.kwartz.com


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Re: [Asterisk-Users] Fritz ISDN PCI v2 and CAPI

[Michael Welter]

Can CAPI and the ASUSCOM ISDNLink card be used in the US?  What goes on 
the /etc/capi.conf file instead of fcpci?

Brian Cuthie wrote:
Can this Frtiz card be used in the US?

-brian 

--
Michael Welter
Introspect Telephony Corp.
Denver, Colorado
+1 303 674 2575
[EMAIL PROTECTED]
www.introspect.com
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[Asterisk-Users] Web interface for Asterisk

[Jain, Sonal]

I installed the flash operator panel and I also installed the flash-shockwave in my 
mozilla browser. I followed the read me instructions in the Flash operator and made 
the changes to the op_server.pl but when I run the browser I get transferring data and 
just sits there. I don't see anything being transferred. If any body has used this 
software please tell me what am I doing wrong.
 I copied the two files from the html directory to /var/www/html/panel directory which 
is the web root.
I also changed the manager.conf file and created a user ID and secret which I 
specified in the op_server.pl.

Thanks,
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[Asterisk-Users] PC based Switchboard application

[Keith D'Atrio]

Hello All
 I am looking for a PC based 
switchboard application. Cisco CallManager has a web attendant console that 
allows you to use the PC to transfer calls and the like and I was wondering if 
there was a similar program compatible with *.
Thank you in advance
Keith D'Atrio

[Asterisk-Users] Restart Asterisk

[Jain, Sonal]

Is it true that every time we make a change in the configuration file we need to 
restart the asterisk server. This will not be practical in the production environment. 
Thanks,
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Re: [Asterisk-Users] Restart Asterisk

[Thomas Gallaway]

Jain, Sonal wrote:

Is it true that every time we make a change in the configuration file we need to restart the asterisk server. This will not be practical in the production environment. 
Thanks,
 

Entering reload in the console should do if you edit the extensions.conf 
and some other files. There are some files if you edit them you need to 
shut down and restart asterisk.
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Re: [Asterisk-Users] Restart Asterisk

[Steve Foy]

You can reload the config files with the 'reload' command in the CLI.

On Thu, Apr 08, 2004 at 09:48:57AM -0400, Jain, Sonal wrote:
 Is it true that every time we make a change in the configuration file we need to 
 restart the asterisk server. This will not be practical in the production 
 environment. 
 Thanks,
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-- 
Steve Foy|  http://www.unite.net
UNITE Solutions  |  Tel: 028 9077 7338 
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Re: [Asterisk-Users] Restart Asterisk

[WipeOut]

Jain, Sonal wrote:

Is it true that every time we make a change in the configuration file we need to restart the asterisk server. This will not be practical in the production environment. 
Thanks,
 

No, you don't have to restart, you have to reload..

From the CLI just type reload and hit enter..
or for a command line run asterisk -rx reload..
Later..

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Re: [Asterisk-Users] Web interface for Asterisk

[Steve Foy]

Hi again :)

Can you give me a URL for the software you mentioned?

Cheers,
Steve

On Thu, Apr 08, 2004 at 09:45:47AM -0400, Jain, Sonal wrote:
 I installed the flash operator panel and I also installed the flash-shockwave in my 
 mozilla browser. I followed the read me instructions in the Flash operator and made 
 the changes to the op_server.pl but when I run the browser I get transferring data 
 and just sits there. I don't see anything being transferred. If any body has used 
 this software please tell me what am I doing wrong.
  I copied the two files from the html directory to /var/www/html/panel directory 
 which is the web root.
 I also changed the manager.conf file and created a user ID and secret which I 
 specified in the op_server.pl.
 
 Thanks,
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-- 
Steve Foy|  http://www.unite.net
UNITE Solutions  |  Tel: 028 9077 7338 
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Re: [Asterisk-Users] Web interface for Asterisk

[Altus Snyman]

ok this is what I did
I moved all to my /var/www/html/control. did the changes is my files and
used the copy of manager.conf. I started asterisk and did
/var/www/html/control/op_server.pl and pointed my browser to
192.168.0.1/control/html ... had the  same problem. Then I went and set
debug to 1 in op_server(did not help cant read the lang?). So I made the
dir /var/www/html/wtf and moved the 2 files in the html dir to
here,restarted asterisk and /var/www/html/control/op_server.p and
pointed my browser to 192.168.0.1/wtf and wtf it worked,now Im not
talking about the transfer and hangup??
here is my conf
##
# CONFIGURATION
#
# parameters to connect to Asterisk Manager
my $manager_host   = 192.168.0.1;
my $manager_user   = altus;
my $manager_secret = altus;
   
  
#
# parameters for the op_server
my $web_hostname  = 192.168.0.1;   # must be the same address you use
to contact the web server
my $listen_port   = 4445;
my $security_code = 'd39i393kd';   # secret code for performing
hangups and transfers
   
  
#
# location of variables.txt needed by the flash applet
# (must be the same directory as the web page and swf file)
my $flash_dir = /var/www/html/wtf/;
   
  
#
# Debug level to stdot
my $debug = 1;




On Thu, 2004-04-08 at 15:45, Jain, Sonal wrote:
 I installed the flash operator panel and I also installed the flash-shockwave in my 
 mozilla browser. I followed the read me instructions in the Flash operator and made 
 the changes to the op_server.pl but when I run the browser I get transferring data 
 and just sits there. I don't see anything being transferred. If any body has used 
 this software please tell me what am I doing wrong.
  I copied the two files from the html directory to /var/www/html/panel directory 
 which is the web root.
 I also changed the manager.conf file and created a user ID and secret which I 
 specified in the .
 
 Thanks,
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RE: [Asterisk-Users] dreaded Caller*ID failed checksum

[Jeremy Hall]

Jeff,

I see the same thing on my FXO card, but it is an Intel modem, not a
true Digium X100P.  I suspected it was my card, but if you are seeing it
on a true card, maybe there is hope for mine yet.  I haven't had time to
troubleshoot yet as I have been having too much fun playing with other
features.

Let us know if you find the solution, and I will do the same if I get
mine working.  I am hoping to be able to do some work on it this weekend
to try and see what is going on.  In my case I have several other phones
plugged into the line as I don't have any FXS ports yet, so eliminating
them was going to be one of my first steps.  The jack that my * server
is attached to is CAT5 run directly from the telco access box.

Aside from being a software decoding error or a telco sending error, my
first suspects are line noise on the cabling from other devices or
devices near the phone cabling.  Electrical noise introduced into the
signal inside the asterisk system is another failure point I want to try
to eliminate.

As a last resort, I was thinking of throwing that modem into my Windows
PC and loading the drivers and software for it and see if CallerID works
in that mode.  I don't know if Windows would be able to load modem
drivers for the Digium card or not, but that is another idea for you to
try.  These cards are basically glorified sound cards that attach to a
telephone line, so if the Windows software can correctly read the
signal, that would maybe point it in the software or driver area.  If
that turns out to be the case, I may be forced to go ahead and get an
actual Digium card sooner than I anticipated in order to prove the
theory.

Regards,

Jeremy

-Original Message-
From: Jeff Gustafson [mailto:[EMAIL PROTECTED] 
Sent: Thursday, April 08, 2004 12:06 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] dreaded Caller*ID failed checksum

Caller*ID used to work as some point, but I can't seem to get it
going
these days.  The card is a x101p.  I've tried going up and down the
rxgain scale.  Can the txgain effect it at all?  When I plug in a phone
into the line with a splitter it can decode caller id with no problems.
Reading through the mailing list archives hasn't given me any
move clues.  Any ideas?

...Jeff

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Re: [Asterisk-Users] Restart Asterisk

[Areski]

Nop, just make a reload to load your new configuration and that will
not alter the current connections.


On Thu, 2004-04-08 at 15:48, Jain, Sonal wrote:
 Is it true that every time we make a change in the configuration file we need to 
 restart the asterisk server. This will not be practical in the production 
 environment. 
 Thanks,
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[Asterisk-Users] transfer sip

[Altus Snyman]

Good day all.
I need a windows client that can transfer calls from 1 user 2 another
with a nice GUI for non PC iterated people
Thanks

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[Asterisk-Users] Fwd: Sasquatch, the Loch Ness Monster, UFOs and...

[John Todd]

Every half year or so, I probably will repost this list, adding and 
subtracting as the community makes advances (or ignores what isn't 
required.)


Date: Thu, 9 Oct 2003 04:51:23 -0400
To: asterisk-users-lists.digium.com
From: John Todd [EMAIL PROTECTED]
Subject: Sasquatch, the Loch Ness Monster, UFOs and...
Mythical Asterisk Creatures, oft-discussed, rarely seen:

1) An advanced graphical user interface
We're getting there.  There are starting to appear a crop of PHP or 
in at least one case, Flash-driven front ends for users.  These 
haven't been compiled as part of asterisk-addons, but perhaps 
sometime in the next month or two the code from the existing various 
projects can be pushed into the addons directory.

2) An IAX2 hardware device
Any Day Now(tm).  Wasim has fallen off the face of the Earth, but 
I've seen with my own two eyes a working copy of the Iaxy from 
Digium, so this holds promise.  My request for a 1u 24-port IAX-based 
box that takes Digium daughterboards (FXO or FXS) generated some 
interest when a show of hands was asked for at the VON show... Bob 
Knight seemed to have an interest and some time on his hands.  ;-)

3) A Radius CDR report module
This sort-of exists now, but again is not a completely robust 
solution.  I've not implemented it yet (due to other pressing issues 
of life and profit) but it should hopefully work with some of the 
traditional billing systems that existing VoIP carriers are using.

4) A live-method, robust SQL-based dialplan
Not sure on this one - anyone care to comment?

5) LDAP/SQL/Radius authentication for SIP phones
I hear rumors of this existing, but again, I haven't had the time to 
investigate.  The SQL-friends database hacks might be the answer for 
an SQL system.

6) Robust R2 signalling support
Steve Underwood says that he's made advances... has anyone else done 
any work on R2?

7) Multilingual language recordings of all existing * .gsm files
Nothing that I know of towards this end, or at least, nothing that is 
available on the CVS server.  Anyone?

8) Free exchange of PSTN gateways in a centralized routing arbiter model
HO ho ho ho ho... that's a funny one.  Actually, I have someone 
working on TRIP now, but I suspect that budget will get cut as soon 
as another project starts to explode.

9) Speech recognition support
Nothing towards this yet - sphinx keeps getting mentioned, though I 
don't know anyone who has had it running in anything other than a 
crippled test, or at least I don't remember anyone saying anything 
about it.

Here are this halfyear's additions:

10) Encryption

I'd love to see TLS/SRTP built into the SIP stack, to support the 
Zultys and Sipura devices which now handle crypto natively.  More 
clients will support this functionality; time to start building 
Asterisk to work with them.  Additionally, IAX2 would be much cooler 
if it had a full-channel encryption method, which I know is at least 
being thought about (the aes header files have appeared in the CVS 
distro.)

11) Presence.

Support for presence integration into devices would be great, and is 
this year's hot-button technology.  Just simply supporting line 
appearances would help out quite a bit for business users on newer 
devices which support that feature, but the same technology 
(subscribe/notify) could be used for more advanced presence features. 
My ideas about integration into existing chat services might have 
some merit, or maybe not.

12) BSD Support

We've got Asterisk compiling, now to get Zaptel/libpri working with 
Digium cards...  rumors have someone Almost Done(tm)

13) High-density Zap cards

Inexpensive DS3 Zap-driven cards would be a boon for large providers. 
The cards exist, there are Linux drivers, all that is required is 
some GPL'ed glue code and hair-pulling to weave it into 
Zaptel/libpri.  With the data mode on Asterisk, it might also be 
possible to provide the equivalent of a Cisco CT3+ card that does 
voice as well.

That's all I can think of at the moment.  Comments are welcome.

JT
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Re: [Asterisk-Users] Restart Asterisk

[Ryan Thrash]

You should be able to do a reload, not having to restart (and bringing 
the system down).

On Apr 8, 2004, at 8:48 AM, Jain, Sonal wrote:

Is it true that every time we make a change in the configuration file 
we need to restart the asterisk server. This will not be practical in 
the production environment.
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RE: [Asterisk-Users] dreaded Caller*ID failed checksum

[Oliver Wilcock]

Some input for those with bigger brains
like, perhaps, Jeremy Hall, to ponder in relation to the Caller*ID failed
checksum message.

1. I have a Digium X101P (and
TDM420P)
2. I installed in January and
had no problems with Caller ID
3. I played with Asterisk including
a few CSV updates in February (stable 1.0, I think).
4. Then I had problems with Asterisk
never reporting Caller ID to phones on the TMD420P and returning the failed
checksum message sometimes.
5. I updated to Apri 4 CSV of
zaptel and asterisk and now Caller ID works.
6. I have ADSL, which introduces
noise on the line and I have an emergency phone which bypasses Asterisk
(in case of power failure). That phone reports Caller ID without
problems.
7. Now the X101P fails to detect
hangups reliably, though I think it is more reliable than it was in March
when I was using the stable code.

Re: [Asterisk-Users] Web interface for Asterisk

[Altus Snyman]

http://sip.house.com.ar/operator/

On Thu, 2004-04-08 at 16:01, Steve Foy wrote:
 Hi again :)
 
 Can you give me a URL for the software you mentioned?
 
 Cheers,
 Steve
 
 On Thu, Apr 08, 2004 at 09:45:47AM -0400, Jain, Sonal wrote:
  I installed the flash operator panel and I also installed the flash-shockwave in 
  my mozilla browser. I followed the read me instructions in the Flash operator and 
  made the changes to the op_server.pl but when I run the browser I get transferring 
  data and just sits there. I don't see anything being transferred. If any body has 
  used this software please tell me what am I doing wrong.
   I copied the two files from the html directory to /var/www/html/panel directory 
  which is the web root.
  I also changed the manager.conf file and created a user ID and secret which I 
  specified in the op_server.pl.
  
  Thanks,
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[Asterisk-Users] Zapata required?

[Steven Kokinos]

Hello-

As part of the 
asterisk build/installation instructions it mentions that the zaptel drivers 
should be built and configured first. My question is whether they are required 
at all, in the case of a system with no hardware cards at all (as is the 
situation in my case).

With them loaded I 
continually get the following message on my console (server not 
asterisk):

Zapata Telephony 
Interface Registered on major 196No ISA tormenta card found at 
dZapata Telephony Interface Unloaded

which seems logical 
given that I don't have any zap hardware. how would i go about unloading this 
module and/or does it need to be compiled at all in this 
case?

Regards,

-Steve

RE: [Asterisk-Users] res_motv: Request for Comment

[Robert Jackson]

I completely agree.  This way you can get the same functionality on
demand instead of automatically.

-Original Message-
From: Duane [mailto:[EMAIL PROTECTED] 
Sent: Thursday, April 08, 2004 5:23 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] res_motv: Request for Comment


Andy Powell wrote:
 ...but the core of my 'problem' is software that calls home.

I agree a separate tool is possibly the best option here for privacy 
reasons...

Simple solution seems to me, have a version.h file in each module and 
rather then calling home publish the version info that is downloaded 
rather then uploaded with some sort of changelog type system, and the 
app running locally will detail on request the changes between the 
different versions...

obviously gzip'ing the changelog will save download times...

I think it's a good idea if implemented correctly, everything off by 
default, and give good details on what it does specifically and how to 
enable it and that should keep everyone happy... Those that are 
especially paranoid, just don't compile the version checking tool, so if

anyone complains later not only did they have to run the util, but they 
had to compile it as well...

-- 
Best regards,
  Duane

http://www.cacert.org - Free Security Certificates http://www.nodedb.com
- Think globally, network locally http://www.sydneywireless.com -
Telecommunications Freedom http://happysnapper.com.au - Sell your photos
over the net! http://e164.org - Using Enum.164 to interconnect asterisk
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Re: [Asterisk-Users] Fwd: Sasquatch, the Loch Ness Monster, UFOs and...

[Andy Powell]

On 08/04/2004 at 10:00 John Todd wrote:

Any Day Now(tm).  Wasim has fallen off the face of the Earth, but
I've seen with my own two eyes a working copy of the Iaxy from
Digium, so this holds promise.  My request for a 1u 24-port IAX-based
box that takes Digium daughterboards (FXO or FXS) generated some
interest when a show of hands was asked for at the VON show... Bob
Knight seemed to have an interest and some time on his hands.  ;-)

Nope, Wasim is alive and kicking and I have the demo iax phone here, it's currently 
doing a tour of Europe and then is off to the USA... (complete with mouse cheese ;) ). 
I also have an IAXy here and can confirm it works very well (except for not being able 
to turn of ADSI eerrk).. I've mentioned a couple of times that I'd like to see an 
equivalent of an IAXy channel bank device (much like you describe) unfortuantely this 
idea was pooh-poohed by people who simply had no interest in it. I'd be prepared to 
take a look, but telecoms electronics is not my field...

Andy


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Re: [Asterisk-Users] Out of trunk data space on call number 16386, dropping

[Tony Buser]

Actually what he means is we can only get the error to show up while in 
a conference call no zap involved.

Rarely if ever have we seen out of trunk on regular trunked iax calls 
between servers, but we can get it to happen every time (3 or 4 times 
every minute or two) when 2 users connected to server 2 hosted 
conference using sip phones and 1 user from server 1 calls in to the 
conference on server 2 via an iax trunk using a sip phone or a zap line.

Justin Carlson wrote:

how did you guys go about diableing it.  Is it the threwaycalling 
directive in zapata.conf ?


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Re: [Asterisk-Users] Zapata required?

[Michiel Betel]

Steve,

No you don't need the zap drivers if you are not using zap-based 
hardware. You might want to use the ztdummy driver though as a timing 
source for conferencing.
Since the asterisk zaptel  support is in a loadable module you can 
instruct Asterisk *not* to use it by specifiing
   noload =  chan_zap.so
in  /etc/asterisk/modules.conf

Steven Kokinos wrote:

Hello-
 
As part of the asterisk build/installation instructions it mentions 
that the zaptel drivers should be built and configured first. My 
question is whether they are required at all, in the case of a system 
with no hardware cards at all (as is the situation in my case).
 
With them loaded I continually get the following message on my console 
(server not asterisk):
 
Zapata Telephony Interface Registered on major 196
No ISA tormenta card found at d
Zapata Telephony Interface Unloaded
 
which seems logical given that I don't have any zap hardware. how 
would i go about unloading this module and/or does it need to be 
compiled at all in this case?
 
Regards,
 
-Steve


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RE: [Asterisk-Users] dreaded Caller*ID failed checksum

[Bisker, Scott (7805)]

Did 
you install the micro filters that came with with your ADSL modem. Usually 
you get 3-4 of these. They are used to protect your analog lines from the 
additional signal noise from the ADSL signal.

-sb

Radio Shack item number 279-103 for 
about $15 each

  -Original Message-From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED]On Behalf Of Oliver 
  WilcockSent: Thursday, April 08, 2004 10:30 AMTo: 
  [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] 
  dreaded Caller*ID failed checksumSome input for those with bigger brains like, perhaps, 
  Jeremy Hall, to ponder in relation to the Caller*ID failed checksum 
  message. 1. I have a Digium 
  X101P (and TDM420P) 2. I 
  installed in January and had no problems with Caller ID 3. I played with Asterisk including a few CSV 
  updates in February (stable 1.0, I think). 4. Then I had problems with Asterisk never reporting Caller ID to 
  phones on the TMD420P and returning the "failed checksum" message 
  sometimes. 5. I updated to Apri 
  4 CSV of zaptel and asterisk and now Caller ID works. 6. I have ADSL, which introduces noise on the 
  line and I have an emergency phone which bypasses Asterisk (in case of power 
  failure). That phone reports Caller ID without problems. 
  7. Now the X101P fails to detect 
  hangups reliably, though I think it is more reliable than it was in March when 
  I was using the stable code. 

[Asterisk-Users] NTL PRI Config required please

[Jon Shamash]

Hi,

I've today received an E100P.

Can anyone help me out with the config of this card please. We have an
NTL PRI circuit installed with 10 channels active.


Also what do the LEDs on the back of the card mean ?


Many thanks

Jon

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Re: [Asterisk-Users] quadBRI and UK ISDN2e

[Jon Fautley]

stan wrote:
On Wed, Apr 07, 2004 at 12:37:43PM +0100, Jon Fautley wrote:

Morning Asterikians,

I've just got my nice shiny quadBRI card, and it seems to be working 
very well - except for one little issue - CallerID.

The card is currently connected to an ISDN2e line in P2P mode, and an S0 
adapter on our existing alcatel PBX.


Is this an omnipcx?
Jup, omnipcx 4400.


The S0 connection recieves callerID 
and displays it correctly - the 2e line doesn't, and BT have said that 
CLID was enabled on the line two days ago. Does anyone have any pointers 
on this?


I assume the callerid is also being displayed on the alcatel handsets?
or is this just callerid generated on internal calls?
The * box displays callerid in internal/external calls (the omnipcx is 
owned by a seperate company, we just didn't want to pay to call them 
(they're in the same building))

If it is then that should show the bt line is setup correctly.  I have
callerid working from a bt 2e line in ptp mode using zaphfc.  So
assuming the bri-stuff versions match the only difference would be the
quadbri card.  Not sure where that leads because I think/thought all the
callerid stuff would be handled by libpri from a q.931 SETUP message on
the d-channel and not be driver/card specific.
I could only get my s0 box to operate in ptmp mode, so there would be
change in the signalling line in zapata.conf, but then if you weren't
changing that nothing tends to work rather than just callerid.  I also
note that you have to match the msn assigned to the s0 when dialling out
through the omnipcx whereas bt doesn't seem to be as fussy, but again
nothing todo with callerid.
I had a slight configuration problem with the S0 adapter that resulted 
in nothing working, but that's all fixed now, you're right though, it's 
an all/nothing senario - either it all worked or none of it worked.

I've just been onto BT again, and it seems they did have CLIP on the 
line for a few minutes, but then they removed it again and put COLP on, 
which they then told me was the same as CLIP... monkeys :(

Wait... I take that back.. calling BT a bunch of monkeys is insulting to 
monkeys.

They then proceeded to tell me that I didn't need to purchase a 
presentation number service (that allows us to display our 
non-geographic numberto people with CLIP rather than our geographic 
one)... I could just tell my switch to send whatever callerID I wanted 
and it'd get displayed... BT ISDN2e - the phreakers delight :)

Thanks for all your help...

slightly bitter and twisted about BT Jon
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[Asterisk-Users] can't hear vm audio

[Mark Phillips]

So I've been fighting to get the X100P working. A battle which I've kinda
won but not without a cost.

Before I won the Zaptel battle I was able to hear all of the messages that
asterisk plays. For example, when I'm accessing VoiceMail I would have
been requested to input my password. This did work but now it doesn't.
Asterisk does show that it is playing the file but no audio is heard.

I have audio on regular SIP based calls as well as IAX based ones. I'me
not getting and audio when I make a ZAP call.

Ideas?

Mark



G7LTT/KC2ENI
Mark Phillips



G7LTT/KC2ENI
Mark Phillips
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[Asterisk-Users] RE: Asterisk-Users digest, Vol 1 #3368 - 12 msgs

[Jain, Sonal]

This message is in response to Flash operator problem. My op_server.pl seems to be 
same. I also created the variable.txt to the /var/www/html/panel folder and when I run 
htt://192.168.0.0/panel it just says at the bottom transferring data. I don't see 
anything on the screen.
I also checked my manager.conf file. I was able to telnet into the manager interface 
and it's running fine.
So I am not sure why I don't see anything on the screen. What about he op_server.cfg 
file. Do I need to change that. Can the default one still work at least bring up the 
screen to tell me it is working fine.
Thanks
 -Original Message-
From:   [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]  On Behalf Of [EMAIL PROTECTED]
Sent:   Thursday, April 08, 2004 11:13 AM
To: [EMAIL PROTECTED]
Subject:Asterisk-Users digest, Vol 1 #3368 - 12 msgs

Send Asterisk-Users mailing list submissions to
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To subscribe or unsubscribe via the World Wide Web, visit
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or, via email, send a message with subject or body 'help' to
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When replying, please edit your Subject line so it is more specific
than Re: Contents of Asterisk-Users digest...


Today's Topics:

   1. RE: Out of trunk data space on call number 16386, dropping (Justin Carlson)
   2. Caller ID on TDM400P Quad FXS (Daniel ANDRE)
   3. Re: Fritz ISDN PCI v2 and CAPI (Michael Welter)
   4. Web interface for Asterisk (Jain, Sonal)
   5. PC based Switchboard application (Keith D'Atrio)
   6. Restart Asterisk (Jain, Sonal)
   7. Re: Restart Asterisk (Thomas Gallaway)
   8. Re: Restart Asterisk (Steve Foy)
   9. Re: Restart Asterisk (WipeOut)
  10. Re: Web interface for Asterisk (Steve Foy)
  11. Re: Web interface for Asterisk (Altus Snyman)
  12. RE: dreaded Caller*ID failed checksum (Jeremy Hall)

--__--__--

Message: 1
From: Justin Carlson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Out of trunk data space on call number 16386, dropping
Date: Thu, 8 Apr 2004 08:16:09 -0500
Reply-To: [EMAIL PROTECTED]

This is a multi-part message in MIME format.

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how did you guys go about diableing it.  Is it the threwaycalling directive
in zapata.conf ?
  -Original Message-
  From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Warren H. Prince
  Sent: Thursday, April 08, 2004 8:01 AM
  To: [EMAIL PROTECTED]
  Subject: Re: [Asterisk-Users] Out of trunk data space on call number
16386, dropping


  I work with Tony, so I'm responding for him.  Yes, it appears only during
a conference call.  So, if we disable conferencing, we do not receive the
error.

  Justin Carlson wrote:
if you disable conferencing does the problem go away?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Tony Buser
Sent: Wednesday, April 07, 2004 2:30 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Out of trunk data space on call number
16386, dropping


I'm having the same kind of issues.  We get the out of trunk data space
error consistently during conference calls between asterisk servers.
And occasionally on regular iax calls.  Also while we're on a conference
call it seems to cause other calls going out through iax to fail and
also give this error.  (weather its to another asterisk server or
through say oneunified)

If you figure this out, please let us know here.  I'm pretty much at a
loss as to what could be causing it.

Justin Carlson wrote:


Hi all,

We keep getting these and all the calls between these two asterisk boxes

get

dropped.  what is going on here, I have been trying to solve this problem

on

my own but maybe I don't have the trunk setup right.  also I have posed

the

output of my full log of the machine with the zap interface, the other is
using ztdummy.



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HTMLHEAD
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BODY
DIVSPAN class=3D334301513-08042004FONT face=3DArial color=3D#ff =
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did you guys 

[Asterisk-Users] TigerJet ISDN card

[Brian Cuthie]

Title: TigerJet ISDN card







Is there any Linux/* support for the TigerJet ISDN card?


-brian

Re: [Asterisk-Users] Re: IAXTel toll-free gateway

[Chris Sullivan]

On Apr 7, 2004, at 11:43 AM, James H. Cloos Jr. wrote:

I'd suggest using enum lookups on freenum.org instead.

Cf:

http://www.mail-archive.com/[EMAIL PROTECTED]/ 
msg23732.html
This brings up an interesting question.

I tried the recipe in that message, and I keep getting this error  
message:

Apr  8 08:29:13 WARNING[360468]: pbx.c:1198 pbx_extension_helper: No  
application 'GotoIf($[$[${ENUM:0:3} = SIP] | $[${ENUM:0:3} =' for  
extension (all-access, 18005551212, 2)

I'm using the 3/27 CVS checkout.

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[Asterisk-Users] Asterisk 3com nbx 100 support

[Jeremy Koski]

Does anybody know if the 3com NBX 100 phones will work with
Asterisk? The phones require a boot image to be sent either through
layer2 or layer3 before they will function properly after being powered
on each time.

Thanks.





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Re: [Asterisk-Users] SIP -- PSTN gateways

[Chris Sullivan]

On Apr 7, 2004, at 10:53 PM, Tom wrote:

1. Do any of the standard companies (Packet8, Broadvox, Vonage, etc.) 
allow
you to use your own SIP device (phone or something like *) instead of 
the
interface hardware they usually provide?
  Many of the providers use SIP, so even if they don't explicitly tell 
you
the SIP settings to us, it is possible to hookup something else.  I've
heard that Packet8 gets annoyed when people use self-supplied devices, 
but
it would be possible to hack the code to emulate the device identifier.
I've seen an Vonage example posted.
I've had great luck with VoicePulse's Connect service.  They just 
opened up a bunch of new rate centers, including Los Angeles/Orange 
County.  $7.99/mo per phone number, 2.9 cents a minute for long 
distance outbound, no minimums or contract.


3. Finally, do any of the providers deliver more than one call via 
SIP?  In
otherwords, if I'm already on a call and another comes in will they 
attempt
to deliver it?
  Good question.  I wonder how many of the providers even have the 
ability
to restrict you to one call!  I hope the unlimited providers do, 
because
otherwise people using Asterisk as a SIP UA are going to have a field 
day.
VoicePulse Connect does.  I had four people in a MeetMe conference last 
night, all dialing in via a VPConnect-supplied number.

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[Asterisk-Users] i'm looking for reference guide for Skinny SCCP

[Tomas Hook]

Hi all,
I'm writing my graduation theses : analysis VO-IP protocols , and I cannot 
find any documents about Cisko Skinny Client Control Protocol. I have Cisco 
CallManager and some IP-phone and I'm sniffing traffic between that, but I 
don't understand, how this protocol works. Clearly i'm looking for 
description of SCCP commands and explanation some basic SCCP scenarios or 
what including SCCP packets ...
I have a lot of alredy information about SIP,H323 and others, but about SCCP 
nothing:-(

Thanks

_
FREE pop-up blocking with the new MSN Toolbar – get it now! 
http://toolbar.msn.com/go/onm00200415ave/direct/01/

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Re: [Asterisk-Users] Zapata required?

[Michiel Betel]

Steven Kokinos wrote:

Ho do I go about loading the ztdummy driver after unloading zap?

 

$ su -
# modprobe ztdummy

Thanks,

-Steve 

 

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RE: [Asterisk-Users] dreaded Caller*ID failed checksum

[Jeremy Hall]

I had DSL at my old house, and am stuck using Cable at the moment as DSL
is not available.  But when at my old house, I had no noise problems
with the DSL filter (yes, filter, I installed it at the line entry point
and ran a separate unfiltered cable directly to the DSL modem.)  Either
Qwest lines are inherently noisy to the point that I couldn't hear a
difference, or my filter worked as it should.  I had no extra noise on
the line, caller ID worked fine.  I didn't have an Asterisk server then,
so I can't comment on that part of it.

If you are still hearing noise from the DSL, I suspect you have a bad
filter or batch of filters.  I have heard some instances where the
signal is strong enough that 2 or 3 filters in line were required to
filter the noise, and that could be your case as well.

Just my $.02 worth.

Jeremy

-Original Message-
From: Oliver Wilcock [mailto:[EMAIL PROTECTED] 
Sent: Thursday, April 08, 2004 9:17 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] dreaded Caller*ID failed checksum

Yes, I installed the filters that came with the modem.  I'm sure you've 
noticed that they don't block all the noise.


 Subject: RE: [Asterisk-Users] dreaded Caller*ID failed checksum
 Date: Thu, 8 Apr 2004 10:55:43 -0400
 From: Bisker, Scott (7805) [EMAIL PROTECTED]
 
 Did you install the micro filters that came with with your ADSL modem.
 Usually you get 3-4 of these.  They are used to protect your analog
 lines from the additional signal noise from the ADSL signal.
 
 Radio Shack item number 279-103 for about $15 each
 

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Re: [Asterisk-Users] The maximum capacity of MeetMe

[Steve Kann]

Christopher Arnold wrote:

  
On Wed, 7 Apr 2004, Steve Kann wrote:

  
  
If people are looking for a higher-capacity conferencing application,
take a look at app_conference, in the iaxclient (on sourceforge) CVS.
I haven't really benchmarked meetme, but I _think_ that app_conference
might be able to beat it.  Certainly in it's designed application it
will (iax clients which use VAD on the client side).


  
  Which version of asterisk is it meant to compile against?

It seems lite the calls to ast_set_read_format and ast_set_write_format
are missing the needlock parameter.
  

It was up to date with asterisk, but I haven't update the CVS on
sourceforge in a bit (nobody had even been looking at it).

I've updated it now, and it should compile against the current 1.0
branch.

Hmm, looking at the code now, we still don't have an extra parameter
there. maybe that was post-1.0?


-SteveK

[Asterisk-Users] Adding two FXO cards - not working

[Hermann Wecke]

I'm trying to add 2 FXO cards but when the second is added, asterisk stop
to respond (zap channel):

app_dial.c:545 dial_exec: Unable to create channel of type 'Zap'
modprobe zaptel and modprobe wcfxo didn't return any error.

ztcfg -vv is reporting only 1 card:
Zaptel Configuration
==
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
1 channels configured.

Where I can find a how to about adding 2 cards to an * box?
(yes - I did change the zaptel.conf and zapata.conf, but I don't know if I
did it right). I also tried to find it using google but nothing came back.

*CLI zap show channels
Chan Extension  Context Language   MusicOnHold
   1bell   random
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Re: [Asterisk-Users] Asterisk 3com nbx 100 support

[Eric Wieling]

On Thu, 2004-04-08 at 10:30, Jeremy Koski wrote:
 Does anybody know if the 3com NBX 100 phones will work with
 Asterisk? The phones require a boot image to be sent either through
 layer2 or layer3 before they will function properly after being powered
 on each time.

You didn't see the information when you searched the mailing list
archives?

--Eric
-- 
Useful Asterisk Docs (BOOKMARK THEM!):
http://www.digium.com/index.php?menu=documentation (look at the
Unofficial Links) and http://www.voip-info.org/wiki-Asterisk and
http://www.fnords.org/~eric/asterisk/ (my site) and
http://asteriskdocs.org/

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Re: [Asterisk-Users] i'm looking for reference guide for Skinny SCCP

[Eric Wieling]

On Thu, 2004-04-08 at 10:52, Tomas Hook wrote:
 Hi all,
 I'm writing my graduation theses : analysis VO-IP protocols , and I cannot 
 find any documents about Cisko Skinny Client Control Protocol. I have Cisco 
 CallManager and some IP-phone and I'm sniffing traffic between that, but I 
 don't understand, how this protocol works. Clearly i'm looking for 
 description of SCCP commands and explanation some basic SCCP scenarios or 
 what including SCCP packets ...
 I have a lot of alredy information about SIP,H323 and others, but about SCCP 
 nothing:-(

Cisco does not document the SCCP protocol.  There are no public docs
available for this protocol.  The Asterisk SCCP/Skinny support was done
by using a packet sniffer.

-- 
  Eric Wieling * BTEL Consulting * 504-899-1387 x2111
In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss.

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Re: [Asterisk-Users] quadBRI and UK ISDN2e

[Linus Surguy]

 I've just been onto BT again, and it seems they did have CLIP on the
 line for a few minutes, but then they removed it again and put COLP on,
 which they then told me was the same as CLIP... monkeys :(

 Wait... I take that back.. calling BT a bunch of monkeys is insulting to
 monkeys.

 They then proceeded to tell me that I didn't need to purchase a
 presentation number service (that allows us to display our
 non-geographic numberto people with CLIP rather than our geographic
 one)... I could just tell my switch to send whatever callerID I wanted
 and it'd get displayed... BT ISDN2e - the phreakers delight :)

You clearly have been speaking to a branch of BT that doesnt have the
slightest clue. I can confirm that the above is definately NOT true. You can
set the caller ID you send out, but only within the range of DDI/MSN numbers
that has been assigned to your ISDN 2e line.

Linus

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Re: [Asterisk-Users] Zaptel/PRI problem

[Chris A. Icide]

Just a note to the list, this problem still seems to exist as late as 
2004-04-07.

Symptoms as we have seen it are this.  System runs along just fine, no 
errors.  At some random interval, not based upon time, and not apparently 
based upon call load, sound drops out on all calls made over the Zap (T1 
PRI interface in our case T100P card).  Any new calls made or received from 
this time forward have no sound.  The problem is temporarily fixed by a 
shutdown and restart of the asterisk code.

This will be a major issue for anyone who is installing or upgrading a 
system using a T1 card (our experience is with the single port digium card, 
but previous postings concerning this issue indicate the problem exists 
with the 4 port as well).

A temporary work around has been to fall back to the CVS version 2004-03-05 
10:28 for zaptel and asterisk.  We've just upgraded (downgraded) from the 
current versions to the march 5th versions and will be watching to see if 
the problem goes away.

Here are some of the error messages we are seeing:

Apr  5 07:57:01 WARNING[-1200284752]: PRI: Read on 31 failed: Unknown error 500
Apr  5 07:57:01 NOTICE[-1200284752]: PRI got event: 8 on span 1
Apr  6 07:10:28 WARNING[-1200215120]: PRI: Read on 29 failed: Unknown error 500
Apr  6 07:10:28 NOTICE[-1200215120]: PRI got event: 5 on span 1
Apr  6 14:22:23 WARNING[-1200260176]: PRI: Read on 29 failed: Unknown error 500
Apr  6 14:22:23 NOTICE[-1200260176]: PRI got event: 6 on span 1
Apr  6 14:53:17 WARNING[-1200165968]: PRI: Read on 30 failed: Unknown error 500
Apr  6 14:53:17 NOTICE[-1200165968]: PRI got event: 5 on span 1
Apr  7 15:00:26 WARNING[-1210471504]: PRI: Read on 64 failed: Unknown error 500
Apr  7 15:00:26 NOTICE[-1210471504]: PRI got event: 5 on span 1
Apr  7 20:33:52 WARNING[-1210508368]: PRI: Read on 55 failed: Unknown error 500
Apr  7 20:33:52 NOTICE[-1210508368]: PRI got event: 8 on span 1
Apr  8 09:42:15 WARNING[-1210725456]: PRI: Read on 32 failed: Unknown error 500
Apr  8 09:42:15 NOTICE[-1210725456]: PRI got event: 5 on span 1
-Chris

On 03:14 AM 3/30/2004, Tais M. Hansen wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Hi.

I'm getting the following error at random intervals on my TE410P with 
Asterisk
CVS-03/30/04-11:49:01-CEST.

I have two spans active, one connected to my Telco, the other to a Siemens
PABX. Both spans display this behavior at random intervals.

All calls are dropped when this happens. Spans are not necessarily in use 
when
this happens.


Mar 30 12:01:41 WARNING[81926]: chan_zap.c:6017 zt_pri_error: PRI: Read 
on 72
failed: Unknown error 500
Mar 30 12:01:41 NOTICE[81926]: chan_zap.c:6733 pri_dchannel: PRI got 
event: 4

snip

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Re: [Asterisk-Users] error compiling cdr_mysql support

[Tilghman Lesher]

On Thursday 08 April 2004 03:07, Alessio Focardi wrote:
 Here is the error I get compiling the asterisk-addons rpm

 cc -fPIC -I../asterisk  -I/usr/include/mysql -c -o
 cdr_addon_mysql.o cdr_addon_mysql.c cdr_addon_mysql.c:50: error:
 `PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP' undeclared here (not in a
 function) make: *** [cdr_addon_mysql.o] Error 1

 I'm pretty sure I got all that is needed:

 # rpm -qa | grep mysql

The problem is not lack of mysql support, the problem is that you
don't have asterisk in the same directory as the asterisk-addons
directory (which is where cdr_addon_mysql.c lives).  Get that in place
and it will compile just fine.

-Tilghman

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Re: [Asterisk-Users] i'm looking for reference guide for Skinny SCCP

[Jeremy McNamara]

Tomas Hook wrote:

Hi all,
I'm writing my graduation theses : analysis VO-IP protocols , and I 
cannot find any documents about Cisco Skinny Client Control Protocol. 
I have Cisco CallManager and some IP-phone and I'm sniffing traffic 
between that, but I don't understand, how this protocol works. Clearly 
i'm looking for description of SCCP commands and explanation some 
basic SCCP scenarios or what including SCCP packets ...
I have a lot of alredy information about SIP,H323 and others, but 
about SCCP nothing:-(



SCCP is not a public protocol.   Lets consider ourselves lucky that 
Cisco is not actively protecting what little documentation is out there.

Your on your own for this...We in the US of A have to worry about those 
nasty four letters:   DMCA

Jeremy McNamara





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Re: [Asterisk-Users] TigerJet ISDN card

[Mark Phillips]

Is it CAPI compliant? if so yes



 Is there any Linux/* support for the TigerJet ISDN card?

 -brian



G7LTT/KC2ENI
Mark Phillips
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Re: [Asterisk-Users] Asterisk 3com nbx 100 support

[Jeremy Koski]

No. The only way I found to search the archives was month by month.

There's an option to download all of the mailing list archives, but its
129MB...

On Thu, 8 Apr 2004, Eric Wieling wrote:

 On Thu, 2004-04-08 at 10:30, Jeremy Koski wrote:
  Does anybody know if the 3com NBX 100 phones will work with
  Asterisk? The phones require a boot image to be sent either through
  layer2 or layer3 before they will function properly after being powered
  on each time.

 You didn't see the information when you searched the mailing list
 archives?

 --Eric
 --
 Useful Asterisk Docs (BOOKMARK THEM!):
 http://www.digium.com/index.php?menu=documentation (look at the
 Unofficial Links) and http://www.voip-info.org/wiki-Asterisk and
 http://www.fnords.org/~eric/asterisk/ (my site) and
 http://asteriskdocs.org/

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Re: [Asterisk-Users] quadBRI and UK ISDN2e

[Steve Kennedy]

On Thu, Apr 08, 2004 at 05:33:18PM +0100, Linus Surguy wrote:

  I've just been onto BT again, and it seems they did have CLIP on the
  line for a few minutes, but then they removed it again and put COLP on,
  which they then told me was the same as CLIP... monkeys :(
  Wait... I take that back.. calling BT a bunch of monkeys is insulting to
  monkeys.
  They then proceeded to tell me that I didn't need to purchase a
  presentation number service (that allows us to display our
  non-geographic numberto people with CLIP rather than our geographic
  one)... I could just tell my switch to send whatever callerID I wanted
  and it'd get displayed... BT ISDN2e - the phreakers delight :)
 You clearly have been speaking to a branch of BT that doesnt have the
 slightest clue. I can confirm that the above is definately NOT true. You can
 set the caller ID you send out, but only within the range of DDI/MSN numbers
 that has been assigned to your ISDN 2e line.

Exactly, you can do lots more if you have SS7 connect, but you then need
a telco switch ... anything that is presented at the premises level will
have severe constraints on what can and cant be set.


Steve

-- 
NetTek Ltd Phone/Fax +44-(0)20 7483 2455
SMS steve-epage (at) gbnet.net [body] gpg 1024D/468952DB 2001-09-19
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Re: [Asterisk-Users] can't hear vm audio

[Mark Phillips]

OK, this is too damn freaky!

Now it work but I didn't do anything.

Actually, that's not quite true. I found that if I do modprobe zaptel then
modprobe wcfxo then ztcfg the vm audio works. If I do service zaptel start
it doesn't.

What gives?


 So I've been fighting to get the X100P working. A battle which I've kinda
 won but not without a cost.

 Before I won the Zaptel battle I was able to hear all of the messages that
 asterisk plays. For example, when I'm accessing VoiceMail I would have
 been requested to input my password. This did work but now it doesn't.
 Asterisk does show that it is playing the file but no audio is heard.

 I have audio on regular SIP based calls as well as IAX based ones. I'me
 not getting and audio when I make a ZAP call.

 Ideas?

 Mark



 G7LTT/KC2ENI
 Mark Phillips



 G7LTT/KC2ENI
 Mark Phillips
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G7LTT/KC2ENI
Mark Phillips
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Re: [Asterisk-Users] Zapata required?

[Mark Phillips]

You did unhash the ztdummy in the Makefile before compiling it right?

 Steven Kokinos wrote:

Ho do I go about loading the ztdummy driver after unloading zap?



 $ su -
 # modprobe ztdummy


Thanks,

-Steve




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G7LTT/KC2ENI
Mark Phillips
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Re: [Asterisk-Users] TigerJet ISDN card

[Michael Welter]

In Linux config, CAPI2.0 support is under the Active ISDN Cards 
category.  Does this mean CAPI will not work with static cards?

Thanks,

Mark Phillips wrote:

Is it CAPI compliant? if so yes



Is there any Linux/* support for the TigerJet ISDN card?

-brian



G7LTT/KC2ENI
Mark Phillips
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--
Michael Welter
Introspect Telephony Corp.
Denver, Colorado
+1 303 674 2575
[EMAIL PROTECTED]
www.introspect.com
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[Asterisk-Users] Using Skinny with a 7905G phone

[Chris Barnett]

Hi All,

I'm trying to get a Cisco 7905g IP Phone to work with our Asterisk
server but I'm having problems getting the phone to answer a call or
make a call. I'm using the stable branch of the asterisk CVS on a RH9
box.

I have got the phone to register with * and it retrieves it's extension
number, date  time etc., but when I pick the handset up it's just dead
silence (but I get tones when pressing the numbers on the keypad). There
are no debug messages or otherwise shown on the * console (even with
very verbose turned on and skinny debugging turned on).

If I ring the phone from a SIP PC softphone (X-Ten lite) the phone rings
but continues to ring even when the handset is picked up - until the 20
seconds timeout occurs and the call is transferred to the extensions
voicemail box. When the call is made I do see debug messages telling me
the call is being made and sent to the skinny extension.

I have experimented with allowing and disallowing various formats to no
avail. I'm sure I've missed something out but after many days and hours
searching every resource I can find I haven't been able to progress any
further - any ideas what I've done wrong or missed out - or is what I am
trying to achieve simply not possible at the moment.

Below is my skinny.conf;

[general]
port = 2000
Bindaddr = 0.0.0.0
dateFormat = D-M-Y
keepAlive = 120
; allow = all
; disallow =

[Ext104]
Device=SEP000E386DCF06
Version=P021114C; version number from the phone
Nat=0
Callerid=Cisco IP Phone 104
Mailbox=1004
Callwaiting=1
Transfer=1
Threewaycalling=1
Context=default
Line = 104
Host=192.168.1.55

And this is the section from extensions.conf;

snip

Exten = 104,1,Dial(Skinny/[EMAIL PROTECTED],20,tr)
Exten = 104,2,VoiceMail,u104
Exten = 104,102,VoiceMail,b104

snip


Many thanks,

Chris Barnett
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Re: [Asterisk-Users] Asterisk 3com nbx 100 support

[Eric Wieling]

Try www.google.com site:lists.digium.com NBX without the quotes, of
course.  There's also a google search box on the same page you use to
sign up for the mailing lists, but it never worked right for me.

On Thu, 2004-04-08 at 11:59, Jeremy Koski wrote:
 No. The only way I found to search the archives was month by month.
 
 There's an option to download all of the mailing list archives, but its
 129MB...
 
 On Thu, 8 Apr 2004, Eric Wieling wrote:
 
  On Thu, 2004-04-08 at 10:30, Jeremy Koski wrote:
   Does anybody know if the 3com NBX 100 phones will work with
   Asterisk? The phones require a boot image to be sent either through
   layer2 or layer3 before they will function properly after being powered
   on each time.
 
  You didn't see the information when you searched the mailing list
  archives?

-- 
  Eric Wieling * BTEL Consulting * 504-899-1387 x2111
In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss.

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Re: [Asterisk-Users] Siemens EWSD 13

[CW_ASN]

In fact, with EWSD V13 you can't remove CRC4 in PRI mode.

Regards,

Gus

- Original Message -
From: Storer, Darren [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, April 07, 2004 8:32 PM
Subject: RE: [Asterisk-Users] Siemens EWSD 13


 Hi,

 I had exactly the same symptoms today with a co-located * connected to a
 Public Switch here in the UK. The problem was solved by insisting that the
 Telco turned on CRC4 at their end and then, after an 'init 6', layer two
 settled down on both systems.

 I was taught that if you are connecting to a full specification Q.931
 circuit, CRC4 should be enabled by default; in the event that one end does
 not support CRC4 the other end should auto-negotiate back and the circuit
 should still align without problems. Having said all of this I have yet to
 see auto-negotiation of CRC4 on any equipment (Public Network or CPE) and
 suspect that I was not told the truth in the first place...

 Selection of CRC4 seems to be random from Telco to Telco even on an
install
 by install basis within the same Carrier. It's the first thing to check
when
 new kit appears to be unstable..

 HTH

 Darren
 --
 Comgate
 TelcoInternetBroadcast

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Behalf Of
 [EMAIL PROTECTED]
 Sent: 07 April 2004 14:59
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] Siemens EWSD 13


 Hi all,

 Has anyone got any experience with hooking Asterisk up with a
 Siemens EWSD 13 switch over a E1/PRI ?
 We're located in Belgium (Europe) and one of our telecom partners
 uses this switch.

 We connected one of our TE410P ports with their switch, but the status
 light on the TE410P card keeps blinking red.
 On their side they are getting a DSA (distance service alarm) error, so
 this normally means the devices 'see' eachother.. but there are still
 problems with the signalling.

 Our config below is the same as we are using for MCI, one of our other
 telecom partners.

 We tried changing the LBO and timing, but no luck.
 As you see the signalling is carried over channel 16 (default).

 TX and RX have also been regularly switched, so no luck..

 Their switch is providing the timing.

 The telecom operator has double checked the asterisk config several
 times, and it's conform to their setup.

 The only thing they couldn't find in the Asterisk config is a
 'multiframing' option. But I presume this is automatically detected or
 set by default ?
 They also tried normal/single(?) framing, but no difference.

 The card has also been tested with our MCI E1, and works flawlessly, so
 no hardware issue.

 Anyone got any further ideas ?

 Any info or help greatly appreciated!

 Our config,

 *** zaptel.conf ***
 span=1,1,6,ccs,hdb3,crc4,yellow
 bchan=1-15
 bchan=17-31
 dchan=16

 *** zapata.conf ***
 [channels]
 switchtype=euroisdn
 signalling=pri_cpe
 pridialplan=unknown

 group=1
 channel = 1-15,17-31

 other zapata standard config



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[Asterisk-Users] Auto Attendant??

[James Moran]

I'm having trouble finding documentation for the auto attendant does
anyone have an idea where there might be some???

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Re: [Asterisk-Users] TigerJet ISDN card

[Michael Welter]

In Linux config, CAPI2.0 support is under the Active ISDN Cards 
category.  Does this mean CAPI will not work with _passive_ cards?

Michael Welter wrote:

In Linux config, CAPI2.0 support is under the Active ISDN Cards 
category.  Does this mean CAPI will not work with static cards?

Thanks,

Mark Phillips wrote:

Is it CAPI compliant? if so yes



Is there any Linux/* support for the TigerJet ISDN card?

-brian



G7LTT/KC2ENI
Mark Phillips
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--
Michael Welter
Introspect Telephony Corp.
Denver, Colorado
+1 303 674 2575
[EMAIL PROTECTED]
www.introspect.com
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[Asterisk-Users] FreeBSD port of asterisk

[David W. Chapman Jr.]

In our FreeBSD port of Asterisk, we have a lot of local patches and I 
was wondering if it were possible to get some of them merged into the 
Asterisk source base.

Thanks

-- 
David W. Chapman Jr.
[EMAIL PROTECTED]   Raintree Network Services, Inc. www.inethouston.net
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[Asterisk-Users] TDM Stater kit all working - WOOHOO - wondering about Asterisk FAX Support

[Kyle Hagan]

 Finally got my Asterisk (Test) system up and working with a TDM Starter
kit. WOOHOO!!!
Goona be buying the T1 cards soon to fully implement the system and FINALLY
get rid of our old Fujitsu Starlog POS.

Was just wondering about the Fax Support. Features say its not complete. Any
idea when this could be available?


Kyle



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Re: [Asterisk-Users] RE: RxFax/spandsp: not disconnecting

[Derek]

I installed rxfax and spandsp. Everything looks good except when 
receiving a fax, only the first page is captured, even though the 
sending machine seems to be transmitting all pages. Am I doing 
something wrong?

-D

Derek Irwin
Information Technology
IncredibleFresh Inc.
Naples, FL
I've been booting my Windows box daily for the last year. The 
computer's OK, but my shoe's starting to wear out.

On Mar 31, 2004, at 11:01 AM, Steve Underwood wrote:

Reynaldo Simbulan wrote:

Hi Steve,

I am having this problem in which RxFax is still holding the file 
after
receiving a complete fax. Somehow the zap channel is still active but 
on the
fax client it was sent successfully.
If you call the line it is still busy.


spandsp-0.0.1k.tar.gz and updated app_rxfax.c and app_txfax.c files 
are available for download. They address this disconnect issue, and 
have a few other minor tweaks.

Regards,
Steve
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[Asterisk-Users] Local Calling Area database?

[Scott Laird]

Is there an easy way to get information about local calling areas out 
of telcos?  I'm trying to get a list of area codes and prefixes in my 
local calling area out of Verizon, and it looks like they've stopped 
providing the information online.  Is there an easy source that I'm 
missing, or do I need to call them and have them mail me a copy every 
few months?

Scott

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Re: [Asterisk-Users] ADPCM 4-bit, 6 kHz

[Steven Critchfield]

On Tue, 2004-04-06 at 10:49, Steve Underwood wrote:
 Why do people get this uncontrollable urge to post, when the don't know 
 the correct answer? :-)

Having the absolute correct answer isn't always important if it steers
the requester in the right direction of self enlightenment. 

Don't discourage new users too much from answering questions as we need
new blood to carry on the fight of educating the even newer users. I
know I for one have been pretty burnt out on it and skip all but a small
handful of questions a week.
-- 
Steven Critchfield  [EMAIL PROTECTED]

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RE: [Asterisk-Users] H.323 Seg faulting

[Derek Samford]

Ive placed a bounty on my bug. See http://bugs.digium.com/bug_view_page.php?bug_id=0001334















From: Derek Samford 
Sent: Wednesday, April 07, 2004
4:26 PM
To:
[EMAIL PROTECTED]
Subject: [Asterisk-Users] H.323
Seg faulting





Can someone take a look, tell me if this is a bug, a
possible resources issue, or my own damn fault?



http://bugs.digium.com/bug_view_page.php?bug_id=0001381





Thanks,

Derek

Re: [Asterisk-Users] Agi and bridging problem when codecs differ

[Steven Critchfield]

On Tue, 2004-04-06 at 03:23, Ron McMillin wrote:
 Hi all,
I have encountered this problem: if the caller is connected to the
 callee using Dial() command called from extensions in extensions.conf,
 there is no problem.  But if the same caller and callee are connected
 using an AGI-exec('Dial'...), the line is disconnected when asnwer.
 There's a problem bridging.  If the codecs are the same on both ends
 then there is no problem.
   Is it different to call Dial from extension.conf than to call from
 AGI-exec('Dial'...)?

While not a solution to the real problem, here is a work around for
similar functionality.

Since the dial should effectively kill the agi app, just set the
context, exten and priority to continue on via the agi interface and
exit. You will then be dumped into the dial plan at the appropriate dial
function to fix your codec problem.

From the technical standpoint, I wonder if this is maybe a EAGI app, or
if there had been some form of recording going on that changed the codec
and then during the exec(dial) you don't reset the codec and it assumes
it is one thing when it isn't anymore. This is totally a guess and not
necessarily fact.
-- 
Steven Critchfield  [EMAIL PROTECTED]

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Re: [Asterisk-Users] IAX call traceability

[Steven Critchfield]

On Tue, 2004-04-06 at 07:44, Velimir Novkovic wrote:
 Hello everyone!
 
 On IAX topic.
 
 Is there a way to know from which * box the call has originated and onto
 which box the call is terminating before call terminates? Can the call
 be trapped efficiently (from dial-plan or such) before leaving network
 of * servers to PSTN (e.g. voice prompt Your calling party is outside
 of free-phone domain. Do you want to proceed?)
 
 Suggestions, thoughts - appreciated.

I think that depends on how you implement your network. 

If you blindly dial from one side to the other, then no because you are
transfering control to the remote machine to take control. 

If you use switch statements, you effectively share your dialplan with
the remote machines and they then can query if you have matches. You
could implement a new application that looked for a dialplan match
without doing a real dial. Then you could then implement your dialplans
with a free context and a non-free context which are linked to your
local dialplans via switch statements. You can then testdial a extention
in your free context to see if it comes back with a match, then test
dial using the non-free context.

I think you will want to look at pbx_find_extension in pbx.c
-- 
Steven Critchfield  [EMAIL PROTECTED]

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Re: [Asterisk-Users] RE: Problems with ADIT 600 - latency, loss, etc

[Ralph Forsythe]

Thanks for the input!

This morning we received an email from carrier saying that he was quite
sure the line was perfectly fine.  He saw a few drops but thought it was
related to something he called a priority 4 queue (which I believe is
carrier-speak for I don't know).

When asked for clarification on what a few drops means, we came to learn
that they registered over *900* drops OVERNIGHT.  This is beyond
ridiculous and I am failing to understand how any competent technician
could consider that perfectly fine and within operating specs.

I have asked for an end to end line test, and am of course getting a less
than enthusiastic response about this (which makes me wonder if they're
online reading the T-berd manual as I type this, or perhaps T-1's For
Dummies).

Anyway it seems this is is not an ADIT problem anymore...  Hopefully they
get the hint here, otherwise another carrier may be getting a sales call
soon.

Thanks again,
Ralph

On Wed, 7 Apr 2004, JR Richardson wrote:

 Ralph,

 My experience with the Adit (going on 3 years), it's a solid platform.  I've
 also had one backed into * using MGCP to CMG card for a bit with great
 success.

 The thing you said about the bandwidth usage and call quality sucks for both
 300KB and 1.5Meg is the clue to this problem being a layer 2 or layer 1
 (physical) problem.  I would demand a T1 end-to-end, test set to test set
 check.  If there are bipolar violations, check timing along the path, if CRC
 errors check copper cross connects along the path.  Another common problem
 with inexperienced communications companies is improperly provisioned
 transport equipment, i.e. one leg of the ckt set to AMI and another leg set
 to B8ZS, the path may still work but have errors.  Bottom line, escalate the
 problem, if they want to keep your business, they'll get it fixed.

 Good luck.

 JR



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[Asterisk-Users] External access to voicemail

[Steven Kokinos]

in my setup i have 
several users with DID lines coming in from various sip/iax providers. within 
our old phone system, a user could call their own DID line, then hit the * key 
when they hear their voicemail greeting and be prompted for their password. 


is there any way 
this could be replicated within asterisk? i'm having trouble figuring it out 
since it steps through things sequentially, whereas i want to scan for input 
during the playback. 

any help would be 
greatly appreciated.

regards,

-steve

[Asterisk-Users] RE: [Asterisk-Users] Réf. : Re: [Asterisk-Users] Fritz ISDN PCI v2 and CAPI

[Roger James]

As far as I am aware the only drivers for the avm fritz card that
provide capi support are the avm fcpci driver and the new modular ISDN
driver mISDN. The generic HiSaX driver is not compatible with CAPI.

If you want to use chan_capi you must have a properly configured capi
middleware system running in kernel and a compatible driver underneath
it.

I.e.
Chan_capi user space
-
capi middleware  kernel space
capi compliant driver (fcpci.o or mISDN.o)

However I think that there are still some unresolved issues with the
chan_capi/fcpci combination with Fritz V2 cards, I seem to remember some
messages on this list a while ago.

Whilst I am here, has anyone found a real fix for the dreaded,

kernel: capidrv-1: controller dead ??
kernel: capidrv-1: listen_change_state state=3 event=1 

messages, which seems to be related to power save mode in BT System Y
Exchanges.

Roger

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
 Sent: 08 April 2004 11:10
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] Réf. : Re: [Asterisk-Users] Fritz ISDN PCI
v2
 and CAPI
 
 
 I tried that but it still doesn't work... I think I don't have the
correct
 approach. Have I to install any ISDN drivers (=modules ?) BEFORE
dealing
 with CAPI ?
 
 If yes, why shouldn't I use the hisax drivers (which are kernel ones)
 instead of fcpci drivers (which doesn't seems to work, by the way...)
 
 And finally, how is it possible to link the two modules together?
 
 As you can see, I am drowning in these difficulties and a help would
be
 very welcomed!
 
 Cheers!
 



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Re: [Asterisk-Users] Local Calling Area database?

[William C. Ray]

Hey i was just on there site looking for the same info.

here is a link
http://www22.verizon.com/CallingAreas/LocalCallFinder/LocalCallFinderSAS.htm
William



- Original Message - 
From: Scott Laird [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, April 08, 2004 2:04 PM
Subject: [Asterisk-Users] Local Calling Area database?


Is there an easy way to get information about local calling areas out of 
telcos?  I'm trying to get a list of area codes and prefixes in my local 
calling area out of Verizon, and it looks like they've stopped providing 
the information online.  Is there an easy source that I'm missing, or do I 
need to call them and have them mail me a copy every few months?

Scott

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RE: [Asterisk-Users] Local Calling Area database?

[Steven Sokol]

 Is there an easy way to get information about local calling areas out
 of telcos?  I'm trying to get a list of area codes and prefixes in my
 local calling area out of Verizon, and it looks like they've stopped
 providing the information online.  Is there an easy source that I'm
 missing, or do I need to call them and have them mail me a copy every
 few months?

No.  I actually filed a complaint with my ILEC, my PUC and with the FCC, all
to no avail.  The response was universally You can purchase that
information from a number of 3rd party sources. Which generally meant
TelCordia (evil Belcore) who charges insane amounts for the information.

rantIf you find a source for this, PLEASE LET ME KNOW.  Personally I think
it should be criminal for them NOT to provide anybody who asks with a list
of what is and is not local.  You should be able to ask, in an automated
way, before each call./rant

-S


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Re: [Asterisk-Users] Fwd: Sasquatch, the Loch Ness Monster, UFOs and...

[Olle E. Johansson]

John Todd wrote:

10) Encryption

I'd love to see TLS/SRTP built into the SIP stack, to support the Zultys 
and Sipura devices which now handle crypto natively.  More clients will 
support this functionality; time to start building Asterisk to work with 
them.  Additionally, IAX2 would be much cooler if it had a full-channel 
encryption method, which I know is at least being thought about (the aes 
header files have appeared in the CVS distro.)

Also I would love to see TLS on the manager port.

/O
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RE: [Asterisk-Users] FreeBSD port of asterisk

[Andrew Thompson]

David W. Chapman Jr. wrote:
 In our FreeBSD port of Asterisk, we have a lot of local patches and I
 was wondering if it were possible to get some of them merged into the
 Asterisk source base.
 
 Thanks

You need to ask on asterisk-dev, not asterisk-users.

My guess would be break them down to manageable pieces with explanations and
post them to bugs.digium.com.

-
Andrew Thompson
http://aktzero.com/ 


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Re: [Asterisk-Users] Local Calling Area database?

[Austin M. Brower]

On Thu, Apr 08, 2004 at 02:00:20PM -0500, Steven Sokol wrote:
 rantIf you find a source for this, PLEASE LET ME KNOW.  Personally I think
 it should be criminal for them NOT to provide anybody who asks with a list
 of what is and is not local.  You should be able to ask, in an automated
 way, before each call./rant
 
 -S

http://users.dandy.net/~czg/search.html

http://www.telcodata.us/

Those are the closest places I've found that have the data you're
looking for.  There may be some gaps in what they cover and there may be
some errors.

Austin
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Re: [Asterisk-Users] res_motv: Request for Comment

[Steven Critchfield]

I'm sure I haven't read all the relevant replies yet as I only completed
those who didn't break the thread.

My mindset would be to create a command line switch that returned the
int you mentioned that has the version details. This int could then be
fed to any app that could query your bugs database. This would help in
the automation of large deployments, or the integration with other
monitoring software.

The next step I would see is some form of SOAP call that could return
the data you mentioned. This allows a person who has collected the
version numbers for each of the machines they are running to then create
whatever monitor app they want in whatever language they feel happy
with.

As I see it fitting into my companies current monitoring schemes would
be a package is created at each of the deployed asterisk servers that
contained the version details. This package is routed however we have
configured it to reach a central repository. At this central repository,
we gather them all together and then issue our calls to your website
consolidating all the version numbers necessary into the least number of
calls out to you. We then filter on what is important for each of the
machines in question and register the update information with our
notification service. Then our notification service could notify those
who have registered interest in the machine, level, or event and be
notified appropriately. 

Of course, I would also wish to have the levels be augmented by the
subsystem as has been suggested elsewhere. I only care about Zap and IAX
channels,  core and agi apps. The rest are not interesting at this
moment. This could greatly reduce what I pull back from your servers and
reduce my tossing of records due to lack of interest.

Of course, I have described how it would fit into our monitoring
activity, I could also see a nice web front end built on top of the same
exact SOAP calls, and even run solely from the browser via the built in
Mozilla Javascript SOAP bindings.

The big point is build tools, not necessarily solutions. Tools can be
strung together for the solution, but it can also be broke down and
rebuilt easily enough from the outside.   
-- 
Steven Critchfield  [EMAIL PROTECTED]

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Re: [Asterisk-Users] Local Calling Area database?

[Scott Laird]

On Apr 8, 2004, at 12:00 PM, Steven Sokol wrote:
Is there an easy way to get information about local calling areas out
of telcos?  I'm trying to get a list of area codes and prefixes in my
local calling area out of Verizon, and it looks like they've stopped
providing the information online.  Is there an easy source that I'm
missing, or do I need to call them and have them mail me a copy every
few months?
No.  I actually filed a complaint with my ILEC, my PUC and with the 
FCC, all
to no avail.  The response was universally You can purchase that
information from a number of 3rd party sources. Which generally meant
TelCordia (evil Belcore) who charges insane amounts for the 
information.

rantIf you find a source for this, PLEASE LET ME KNOW.  Personally I 
think
it should be criminal for them NOT to provide anybody who asks with a 
list
of what is and is not local.  You should be able to ask, in an 
automated
way, before each call./rant
Well, I sort of have a source now.  Which is to say, I have a script 
that'll screen-scape a CLEC's website for information for a specified 
number, and then spit out the matching area code/exchange pairs in a 
form suitable for inclusion into a dial plan.  Odds are, the CLEC will 
cut things off eventually, but it works today and seems correct for my 
home phone number, even though I'm not remotely a customer of theirs.

You can grab a copy at 
http://svn.scottstuff.net/project/asterisk-lca-map/.  It's written in 
Ruby, so you'll need to install a ruby interpreter, but there's almost 
certainly one pre-built for whichever Linux distribution you're 
running.

Scott

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[Asterisk-Users] Hangup on SIP unreachable?

[Scott Laird]

I've noticed a little problem with my setup.  I've been using a flaky 
version of X-Lite for testing, and it tends to crash every few phone 
calls.  Since I'm just using it for testing, I don't really care, but 
it's exposed a problem: when the SIP client goes away, their calls are 
left in limbo.  I just had to soft-hangup a multi-hour outgoing call 
that had belonged to my X-Lite client.  Is there a way to treat SIP 
UNREACHABLE as a hangup?  I'm running ~2 week old CVS code right now.

Scott

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[Asterisk-Users] RE: Asterisk-Users digest, Vol 1 #3373 - 14 msgs

[Jain, Sonal]

Can anybody recommend a good web interface for asterisk that actually works.
I am looking for a web interface that can show how many callers are on the phone, 
should be able to transfer the calls and disconnect. I have tried using the flash 
operator but has been unsuccessful in making it work.
thanks

 -Original Message-
From:   [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]  On Behalf Of [EMAIL PROTECTED]
Sent:   Thursday, April 08, 2004 3:30 PM
To: [EMAIL PROTECTED]
Subject:Asterisk-Users digest, Vol 1 #3373 - 14 msgs

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than Re: Contents of Asterisk-Users digest...


Today's Topics:

   1. Re: can't hear vm audio (Mark Phillips)
   2. Re: Zapata required? (Mark Phillips)
   3. Re: TigerJet ISDN card (Michael Welter)
   4. Using Skinny with a 7905G phone (Chris Barnett)
   5. Re: Asterisk  3com nbx 100 support (Eric Wieling)
   6. Re: Siemens EWSD 13 (CW_ASN)
   7. Auto Attendant?? (James Moran)
   8. Re: TigerJet ISDN card (Michael Welter)
   9. FreeBSD port of asterisk (David W. Chapman Jr.)
  10. TDM Stater kit all working - WOOHOO - wondering about Asterisk FAX Support (Kyle 
Hagan)
  11. Re: RE: RxFax/spandsp: not disconnecting (Derek)
  12. Local Calling Area database? (Scott Laird)
  13. Re: ADPCM 4-bit, 6 kHz (Steven Critchfield)
  14. RE: H.323 Seg faulting (Derek Samford)

--__--__--

Message: 1
Date: Thu, 8 Apr 2004 13:03:59 -0400 (EDT)
Subject: Re: [Asterisk-Users] can't hear vm audio
From: Mark Phillips [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]

OK, this is too damn freaky!

Now it work but I didn't do anything.

Actually, that's not quite true. I found that if I do modprobe zaptel then
modprobe wcfxo then ztcfg the vm audio works. If I do service zaptel start
it doesn't.

What gives?


 So I've been fighting to get the X100P working. A battle which I've kinda
 won but not without a cost.

 Before I won the Zaptel battle I was able to hear all of the messages that
 asterisk plays. For example, when I'm accessing VoiceMail I would have
 been requested to input my password. This did work but now it doesn't.
 Asterisk does show that it is playing the file but no audio is heard.

 I have audio on regular SIP based calls as well as IAX based ones. I'me
 not getting and audio when I make a ZAP call.

 Ideas?

 Mark



 G7LTT/KC2ENI
 Mark Phillips



 G7LTT/KC2ENI
 Mark Phillips
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G7LTT/KC2ENI
Mark Phillips

--__--__--

Message: 2
Date: Thu, 8 Apr 2004 13:05:06 -0400 (EDT)
Subject: Re: [Asterisk-Users] Zapata required?
From: Mark Phillips [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]

You did unhash the ztdummy in the Makefile before compiling it right?

 Steven Kokinos wrote:

Ho do I go about loading the ztdummy driver after unloading zap?



 $ su -
 # modprobe ztdummy


Thanks,

-Steve




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G7LTT/KC2ENI
Mark Phillips

--__--__--

Message: 3
Date: Thu, 08 Apr 2004 11:23:31 -0600
From: Michael Welter [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] TigerJet ISDN card
Reply-To: [EMAIL PROTECTED]

In Linux config, CAPI2.0 support is under the Active ISDN Cards 
category.  Does this mean CAPI will not work with static cards?

Thanks,

Mark Phillips wrote:

 Is it CAPI compliant? if so yes
 
 
 
Is there any Linux/* support for the TigerJet ISDN card?

-brian

 
 
 
 G7LTT/KC2ENI
 Mark Phillips
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-- 
Michael Welter
Introspect Telephony Corp.
Denver, Colorado
+1 303 674 2575
[EMAIL PROTECTED]
www.introspect.com



--__--__--

Message: 4
Date: Thu, 8 Apr 2004 18:31:07 +0100
From: Chris Barnett [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Using Skinny with a 7905G phone
Reply-To: [EMAIL PROTECTED]

Hi All,

I'm trying to get a Cisco 7905g IP Phone to work with our Asterisk
server but I'm having problems getting the phone to