date:20051124


Wayne Gemmell wrote:
I'm having trouble receiving faxes using rxfax. Could somebody please browse 
my log file and give me a swift kick in the right direction? I've also added 
my zapata.conf file at the end. 


have you tried using direct indialing, to see if rxfax works? (I assume 
you are now using fax-detection) That way we know if the detection is 
failing or the receiving itself.. (or both :))


cheers

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Re: [Asterisk-Users] SIP Problem

2005-11-24 Thread Elmar Haneke


On Upgrading do 1.2 I do have an PRoblem with my VOIP-Provider.

Making outbound calls does result in Error 400 - exept if I do call my 
own phonenumber.





I dind find the solution th this problem in current CVS source, 
chan_sip.c has to be updated.


Elmar

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[Asterisk-Users] hint problem

2005-11-24 Thread René Enskat [Teamware GmbH]




i enabled hint for
some of my SIP and SCCP lines but on all i have Temp Fail and unavailable line
status.
when i make: SIP
SHOW SUBSCRIPTIONS nothing is shown 

call-limit =
2useclientcode=yesnotifyringing=yes

is in the
config.

Somebody can help
me?

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RE: [Asterisk-Users] Eicon Diva Server query

2005-11-24 Thread David Waugh

Hi Avi,

I added a bit the Asterisk wiki to explain hopefully more clearly how to get it 
installed.

Please have a look at:
http://www.voip-info.org/wiki-Asterisk+Eicon+Diva+CAPI+ISDN

Any feedback appreciated.

After you have installed the Diva Server drivers, please install the 
chan_capi_cm channel.

Cheers
David

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Avi Miller
Sent: 23 November 2005 20:52
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Eicon Diva Server query


David Waugh wrote:
 Yes, you can use the Eicon Diva Range with 2.6 Kernels

Another question, considering the card should arrive tomorrow and I'd 
like to try my hand at setting it up this weekend: Do I need to BRIstuff 
Asterisk to get the Eicon Diva V-4BRI to work, or should I just need 
chan_capi-cm?

Thanks,
Avi

-- 
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[Asterisk-Users] 1.2.0 using 1G of RAM

2005-11-24 Thread Gavin Hamill

Hi, this morning I've switched our 'in-line' Asterisk system (between 
legacy PBX and PSTN) live after a few false starts with the PBX 
configuration.


I've been executing 'show channels' probably hundreds of times, and I 
wanted to see show channel Zap/64-1 So I start with 'show cha' and 
pressed TAB. asterisk then segfaulted, but I was able to immediately 
reconnect to the console with -r.


A minute or two later, response from the console was very sluggish, and 
calls stopped were dropped from the PBX. I ctrl-c'd back to the console 
and found that every byte of 640M swap space was being used as well as 
all of the this 512MB RAM machine, and asterisk's entry in 'top' was:


32021 root  25   0 1112m 473m 7476 S  0.0 93.8   0:00.10 asterisk

I have not yet killed this process and I wondered how I could determine 
where the memory leak is occurring. It's 1.2.0 final, compiled and 
installed with 'make install' - I have not stripped any symbols, etc.


Any help would be greatly welcomed since I dearly want a stable and 
modern asterisk system! :)


Cheers,
Gavin.

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Re: [Asterisk-Users] Eicon Diva Server query

2005-11-24 Thread Avi Miller


David Waugh wrote:

I added a bit the Asterisk wiki to explain hopefully more clearly how to get it 
installed.


Great, thanks. Once I have a working /etc/asterisk/capi.conf for the 
V-4BRI, I'll be sure to add that to the Wiki page for future reference.


cYa,
Avi

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RE: [Asterisk-Users] Eicon Diva Server query

2005-11-24 Thread David Waugh

Hello again.
With the Diva Server 4BRI card - remember that these are in effect 4 CAPI 
controllers. Eg CAPI controller 1, controller 2, Controller 3 and Controller 4.

Therefore you should have 4 sections in your CAPI conf as follows.


; CAPI config 
; 
; 
general 
nationalprefix=0 
internationalprefix=00 
rxgain=0.8 
txgain=0.8 
; If you are in Europe use alaw. Use ulaw otherwise
alaw=yes 

V4BRI-1

;Define CAPI Controller 1 of V-4BRI-8M 
;mode=immediate 
isdnmode=DID 
ntmode=yes; 
incomingmsn=* 
controller=1
softdtmf=0 
accountcode= 
context=isdn 
callgroup=1 
devices=2
echocancel=yes 

V4BRI-2

;Define CAPI Controller 2 of V-4BRI-8M 
;mode=immediate 
isdnmode=DID 
ntmode=yes; 
incomingmsn=* 
controller=3
softdtmf=0 
accountcode= 
context=isdn 
callgroup=1 
devices=2
echocancel=yes 

V4BRI-3

;Define CAPI Controller 2 of V-4BRI-8M 
;mode=immediate 
isdnmode=DID 
ntmode=yes; 
incomingmsn=* 
controller=3
softdtmf=0 
accountcode= 
context=isdn 
callgroup=1 
devices=2
echocancel=yes 

V4BRI-4

;Define CAPI Controller 3 of V-4BRI-8M 
;mode=immediate 
isdnmode=DID 
ntmode=yes; 
incomingmsn=* 
controller=4
softdtmf=0 
accountcode= 
context=isdn 
callgroup=1 
devices=2
echocancel=yes 
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Re: [Asterisk-Users] Asterisk as Softswitch

2005-11-24 Thread trixter aka Bret McDanel

On Thu, 2005-11-24 at 10:09 +0100, Olle E. Johansson wrote:
 Somesh S Shanbhag wrote:
  Hi,
  
  I had doubt like can asterisk talk ISUP over SS7 which
  the normal PSTN
  softswitches talk with other switches?
  
  It becomes *necessary* that asterisk *should* talk
  with other softswitches
  in PSTN using ISUP/SS7 ??
  
 At this point, the standard version of Asterisk can't talk SS7.
 There are a number of commercial and open source additions available
 that adds SS7 to Asterisk, I am not aware whether they support ISUP yet,
 but I would guess so.

one would hope that libisup does.  Its commercial but its also been
through testing and acceptance in europe and asia.  Something you are
not likely to get with free software due to the fact that many people
are less willing to let open source software connect to such networks
for fear that someone will tweak something and hose the network.

http://www.voip-info.org/wiki/index.php?page=Asterisk+ss7+LIBISUP

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US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378
http://www.sacaug.org/ Sacramento Asterisk Users Group


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[Asterisk-Users] PRI problems again - What should I do ?

2005-11-24 Thread Julian Lyndon-Smith


I know that's a real newbie question, but I have a problem.

I keep getting frame rejects, and a D-channel bouncing up and down. BT 
say that it is at my end. If I stop asterisk, stop the zaptel service 
and restart, things seem ok for a while.


I posted a similar problem a couple of days ago, and one of the 
responses suggested that the TE4xxP may be on it's way out.


Is there any way of testing this card to see if that may be the case ?

I was thinking of buying a sangoma a102 as a fall-over - are there any 
issues with the sangoma cards, or should I buy another te4xxp as a backup ?


I was also thinking of moving the * server to a dell 2850 (2x3.06 
processors, 2GB ram, 2x146gb hdd) - again, any gotchas ?


Sorry for so many questions, but we are placing / receiving near on 3000 
calls a day now and my butt is getting sore from all the kicking I've 
received :)


Many thanks for the anticipated (and needed) help :)

Julian
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[Asterisk-Users] Re: Not receiving fax

2005-11-24 Thread Wayne Gemmell

On Thursday 24 November 2005 11:37, Kristof Hardy wrote:
 have you tried using direct indialing, to see if rxfax works? (I assume
 you are now using fax-detection) That way we know if the detection is
 failing or the receiving itself.. (or both :))
I'm not sure what you mean, are you saying that I should some how circumvent 
the menu system to make calls go directly to the fax? Then I should listen 
for noises?

Btw, I've changed my zone to za and my log output has changed slightly. It 
seems that the problem is in dsps busy tone detection.

Nov 24 11:12:04 VERBOSE[18523] logger.c: -- Executing RxFAX(Zap/1-1, 
/var/spool/asterisk/fax/1132823516.1.tif) in new stack
Nov 24 11:12:43 DEBUG[18523] dsp.c: ast_dsp_busydetect detected busy, avgtone: 
469, avgsilence 494
Nov 24 11:12:43 DEBUG[18523] dsp.c: Requesting Hangup because the busy tone 
was detected on channel Zap/1-1
Nov 24 11:12:43 DEBUG[18523] app_rxfax.c: Got hangup
Nov 24 11:12:43 DEBUG[18523] app_macro.c: Extension s, priority 3 returned 
normally even though call was hung up
Nov 24 11:12:43 DEBUG[18523] pbx.c: Extension in_fax, priority 2 returned 
normally even though call was hung up



-- 
Regards

Wayne Gemmell

Work: +27 11 894 2530
Fax : +27 11 894 4081
Cell: +27 83 666 3325
Email   : [EMAIL PROTECTED]
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[Asterisk-Users] [Asterrik-Users] Bristuff for Asterisk 1.2 error

I have download the bristuff-0.3.0-PRE-1.tar.gz and I followed the instruction in INSTALL:  1. make menuconfig  2. make dep  3. ./install.sh  4. copy the file zapata.cong and zaptel.conf  5. modprobe zaptel  6.   But when I doing insmod qozap.o   and ztcfg don't start because in /qozap directory I don't have qozap.o files. Why?I have a quadBRI cardHELP!!!Thanks
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Re: [Asterisk-Users] astman make error

2005-11-24 Thread Tzafrir Cohen

On Thu, Nov 24, 2005 at 12:00:18AM +0100, Fred Blaise wrote:
 On Wed, 2005-11-23 at 19:58 +0100, Fred Blaise wrote:
  Hi all
  
  I am having an issue when trying to 'make astman'. Asterisk 1.2.0 here,
  from source, on debian sarge. Everything else working fine (only SIP
  setup anyway)
  
  deafneuron:/opt/asterisk-1.2.0/utils# make astman
  cc -DNO_AST_MM   -c -o astman.o astman.c
  In file included from /usr/include/asterisk/manager.h:28,
   from astman.c:41:
  /usr/include/asterisk/lock.h: In function `ast_mutex_init':
  /usr/include/asterisk/lock.h:517: error: `PTHREAD_MUTEX_RECURSIVE'
  undeclared (first use in this function)
  /usr/include/asterisk/lock.h:517: error: (Each undeclared identifier is
  reported only once
  /usr/include/asterisk/lock.h:517: error: for each function it appears
  in.)
  make: *** [astman.o] Error 1
 I added -D_GNU_SOURCE to the CFLAGS.. did the trick.

I believe that the basic problem is because make was run directly from 
a subdir, whereas the author has expected make to be run from the top 
directory.

-- 
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[EMAIL PROTECTED] |   |  best
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Re: [Asterisk-Users] Asterisk 1.2 + Debian Sarge

2005-11-24 Thread Tzafrir Cohen

On Thu, Nov 24, 2005 at 03:16:45PM +0800, Dulmandakh Sukhbaatar wrote:
 Tzafrir Cohen wrote:
 
 On Tue, Nov 22, 2005 at 02:00:11PM -0700, Matt wrote:
  
 
 Looks like you need to install the kernel headers  package. While you 
 are at it be sure that you have the kernel source package installed also.

 
 
  apt-get install kernel-headers-`uname -r`
 
 should suffice.
 
  
 
 my $0.02
 
 Juanjo Portela wrote:
 

 
 Dear Collegues
 
 I am trying to compile the new version (Asterisk.1.2) with my debian
 box and i get the following error when i compile the zaptel package:
 
 radio:/usr/src/asterisk-1.2/zaptel-1.2.0# make
 make: Warning: File `Makefile' has modification time 3.1e+08 s in the 
 future
  
 
 
 Hmmm... better setup your clock properly
 
  
 
 And you should create  symlink to your kernel header
 ln -s kernel-headers-`uname -r` linux

No, this should not be needed. At least not with 1.0.10, 1.2.0 and
recent debs in Unstable. Note that the kernel-headers package contain
the symlink /lib/modules/VERSION/build .

Hence you can set up either KSRC or KVERS.

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RE: [Asterisk-Users] Help need to reset Adit 600 for Asterisk install

On Wed, 2005-11-23 at 08:30 -0700, [EMAIL PROTECTED] wrote:
 Does anyone know of a brute force that will work on a serial interface like
 hyperterminal?

Look at expect... you should be able to throw something simple together
using a shell + expect script...

ie, connect and
expect login: 
send  
expect Incorrect
send 1112
etc
Log each attempt to a file, the last number in the file is the right
number ( +/- 1 depending on your logging and logic).

Regards,
Adam

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Re: [Asterisk-Users] Re: Not receiving fax


Wayne Gemmell wrote:
I'm not sure what you mean, are you saying that I should some how circumvent 
the menu system to make calls go directly to the fax? Then I should listen 
for noises?


Yes, make a 'default' to go directly to your fax-receive macro. (rxfax 
witht the parameters)


At least you should hear a 'fax' answering.

Hm, you could try enabling the busy detection in your zapata file..

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Re: [Asterisk-Users] Call parking on Polycom IP501

On Wed, 2005-11-23 at 12:53 -0800, Anthony Rodgers wrote:
 Hi Dave,
 
 exten = callpark,1,Dial(SIP/1000) didn't work - invalid extension

What about:
exten = callpark,1,Dial(Local/[EMAIL PROTECTED])

Regards,
Adam


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Re: [Asterisk-Users] [Asterrik-Users] Bristuff for Asterisk 1.2 error


asterisk183 wrote:

5. modprobe zaptel
6.
  But when I doing insmod qozap.o
  and ztcfg don't start because in /qozap directory I don't have qozap.o 
files. Why?


what is the output you get 'after' you do: modprobe qozap
After this, what is the output you get after: ztcfg -v

Also, the qozap.ko file should have been copied to your 
/lib/modules/your kernel/misc directory. (this is being done by the 
install script)


Cheers
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[Asterisk-Users] Re: Not receiving fax

2005-11-24 Thread Wayne Gemmell

On Thursday 24 November 2005 12:49, Kristof Hardy wrote:
 Yes, make a 'default' to go directly to your fax-receive macro. (rxfax
 witht the parameters)

 At least you should hear a 'fax' answering.
Thanks, I'll try that.

 Hm, you could try enabling the busy detection in your zapata file..
As of my last posted log it was enabled and set to 15.


-- 
Regards

Wayne Gemmell

Work: +27 11 894 2530
Fax : +27 11 894 4081
Cell: +27 83 666 3325
Email   : [EMAIL PROTECTED]
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[Asterisk-Users] Fax sending problems

2005-11-24 Thread Lee Archer

Title: Fax sending problems






Hi, I've got iaxmodem setup but I'm getting failed fax sending. When I send a fax it is spooled through the system and I hear the destination fax machine pick up, it's sat near me, and the transfer starts. However after about 30 seconds the line drops and the fax machine reports an error. The faxsend process still thinks it is running and has to be killed. The logs report

Nov 24 10:50:12.89: [ 8222]: SEND send frame number 132

Nov 24 10:50:12.89: [ 8222]: SEND send frame number 133

Nov 24 10:50:12.89: [ 8222]: MODEM set XON/XOFF/FLUSH: input interpreted, output disabled

Nov 24 10:50:12.89: [ 8222]: DELAY 200 ms

Nov 24 10:50:13.09: [ 8222]: -- [11:AT+FTM=146\r]

Nov 24 10:50:13.10: [ 8222]: -- [7:CONNECT]

Nov 24 10:50:13.10: [ 8222]: DELAY 400 ms

Nov 24 10:50:13.50: [ 8222]: -- data [1025]

Nov 24 10:50:13.50: [ 8222]: -- data [1027]

Nov 24 10:50:13.50: [ 8222]: -- data [1029]

Nov 24 10:50:13.50: [ 8222]: -- data [1029]

Nov 24 10:50:15.02: [ 8222]: -- data [1029]

Nov 24 10:50:15.02: [ 8222]: -- data [1033]

Nov 24 10:50:15.02: [ 8222]: -- data [1034]

Nov 24 10:50:15.02: [ 8222]: -- data [1031]

Nov 24 10:51:15.01: [ 8222]: MODEM TIMEOUT: writing to modem

Nov 24 10:51:15.01: [ 8222]: MODEM WRITE SHORT: sent 1031, wrote 984

Nov 24 10:51:15.01: [ 8222]: SEND end page

Nov 24 10:51:15.01: [ 8222]: Unspecified Transmit Phase C error

Nov 24 10:51:15.01: [ 8222]: -- [9:AT+FTH=3\r]

Nov 24 10:51:22.57: [ 8222]: MODEM TIMEOUT: sending HDLC frame

Nov 24 10:51:22.57: [ 8222]: MODEM input buffering enabled

Nov 24 10:51:22.57: [ 8222]: -- [5:ATH0\r]


Is this an error with my modem or the receiving fax machine?


Regards


Lee



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Re: [Asterisk-Users] [Asterrik-Users] Bristuff for Asterisk 1.2 error

2005-11-24 Thread Tzafrir Cohen

On Thu, Nov 24, 2005 at 11:28:00AM +0100, asterisk183 wrote:
  I have download the bristuff-0.3.0-PRE-1.tar.gz and I followed the 
 instruction in INSTALL:
   1. make menuconfig
   2. make dep
   3. ./install.sh
   4. copy the file zapata.cong and zaptel.conf
   5. modprobe zaptel
   6.
 But when I doing insmod qozap.o
 and ztcfg don't start because in /qozap directory I don't have qozap.o 
 files. Why?
   
   I have a quadBRI card

kernel 2.4 or 2.6? On 2.6 you should have qozap.ko

-- 
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Re: [Asterisk-Users] [Asterrik-Users] Bristuff for Asterisk 1.2 error

Show:  insmod: qozap.o: No such file or directoryKristof Hardy [EMAIL PROTECTED] ha scritto:   asterisk183 wrote: 5. modprobe zaptel 6.   But when I doing insmod qozap.o   and ztcfg don't start because in /qozap directory I don't have qozap.o  files. Why?what is the output you get 'after' you do: modprobe qozapAfter this, what is the output you get after: ztcfg -vAlso, the qozap.ko file should have been copied to your /lib/modules//misc directory. (this is being done by the install script)Cheers___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing
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Re: [Asterisk-Users] PRI problems again - What should I do ?

On Thu, 2005-11-24 at 10:26 +, Julian Lyndon-Smith wrote:

 I posted a similar problem a couple of days ago, and one of the 
 responses suggested that the TE4xxP may be on it's way out.
 Is there any way of testing this card to see if that may be the case ?

Speak to digium and ask them how to run a pattern looptest... You need a
custom cable (not a PRI crossover cable) and some software, but I forget
the details... else search on the wiki for patlooptest or similar
terms...

 I was thinking of buying a sangoma a102 as a fall-over - are there any 
 issues with the sangoma cards, or should I buy another te4xxp as a backup ?

I would suggest keeping identical hardware for your backup use... since
if you use different hardware, you need a different config, and hence
are not testing/isolating the source of the problem...

 I was also thinking of moving the * server to a dell 2850 (2x3.06 
 processors, 2GB ram, 2x146gb hdd) - again, any gotchas ?

Have no idea, but personally I don't like dell :) The only other thing I
would suggest is not to try changing too many things at the same time.

 Sorry for so many questions, but we are placing / receiving near on 3000 
 calls a day now and my butt is getting sore from all the kicking I've 
 received :)

Try and call digium when you can do some proper testing (ie, outside of
your general usage hours, or off-peak hours or whatever... or, if you
can, just schedule an outage time.

Digium provide warranty + support on their products, so best to call
them and find out.

Regards,
Adam


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Re: [Asterisk-Users] [Asterrik-Users] Bristuff for Asterisk 1.2 error

kernel 2.4  Tzafrir Cohen [EMAIL PROTECTED] ha scritto:  On Thu, Nov 24, 2005 at 11:28:00AM +0100, asterisk183 wrote:  I have download the bristuff-0.3.0-PRE-1.tar.gz and I followed the instruction in INSTALL:   1. make menuconfig   2. make dep   3. ./install.sh   4. copy the file zapata.cong and zaptel.conf   5. modprobe zaptel   6. But when I doing insmod qozap.o and ztcfg don't start because in /qozap directory I don't have qozap.o files. Why?  I have a quadBRI cardkernel 2.4 or 2.6? On 2.6 you should have qozap.ko-- Tzafrir Cohen | [EMAIL PROTECTED] | VIM ishttp://tzafrir.org.il |   | a Mutt's  [EMAIL PROTECTED] | 
 
  |  bestICQ# 16849755 |   | friend___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] [Asterrik-Users] Bristuff for Asterisk 1.2 error


asterisk183 wrote:

  and ztcfg don't start because in /qozap directory I don't have
qozap.o files. Why?


If you don't have qozap.o files, then your qozap is not compiled 
correctly. Try (in qozap dir) a 'make clean' and 'make all' and see if 
this produces an error.

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[Asterisk-Users] Sip dosenot fall to default 's' , STRANGE?

2005-11-24 Thread vivek

Hello friends, 
 I have a strange problem. I am using asterisk 1.2 and asterisk addons 1.2. I 
have three SIP phones and one H323 phones connected to asterisk. The problem is 
that when I dial an invalid extension from H323 phones, I get the invalid 
extension message with exten = i... in that context but this does not happen 
with the SIP phones. All I get is something like an engaged tone from the SIP 
phones. Also I am able to dial and transfer between SIP and H323 phones. I am 
not able to figure out whats wrong. None of them are behind the NAT. All of 
them and the asterisk server are on private-ip.
 I also tried  sip debug from the command line and dial an invlaid extension 
from the SIP phone and get nothing but a 
SIP/2.0 404 Not Found in the o/p. But it then dosent fall to the exten = i 
or exten = s.
My conf. files are as under:-

extensions.conf:-
[incoming]
exten = s,1,Answer ; Answer the line.
exten = s,n,Set(TIMEOUT(digit)=5)  ; Set Digit Timeout to 
5 seconds.
exten = s,n,Set(TIMEOUT(response)=10)  ; Set Response Timeout 
to 10 seconds.
exten = s,n(restart),BackGround(demo-congrats) ; Play a congratulatory 
message.
exten = s,n,WaitExten(5)   ; Wait for an extension 
to be dialed.
exten = s,n,Dial(SIP/192.168.1.196,100,t)  , Dial the operator.

exten = i,1,Playback(invalid)  ; That's not valid, 
try again.

[default]
include = incoming ; Instead of demo in 
the sample, there is incoming.

[testing]
include = parkedcalls

exten = s,1,Playback(invalid)  ; When this is present, 
invalid extension from h323 comes here or 
;;; exten = i,1,Playback(invalid)  ;;;even this did not work.   
;;  H323 Phones  ;;
exten = 61,1,Dial(OOH323/192.168.1.194,20|t)  ;ip=h323
;;  SIP Phones   ;;
exten = 62,1,Dial(SIP/62,20|t);new-gray=sip
exten = 63,1,Dial(SIP/63,20|t);old-gray=sip
exten = 64,1,Dial(SIP/64,20|t);ip=sip

ooh323.conf:-
context=testing
disallow=all
allow=ulaw
allow=alaw
dtmfmode=h245alphanumeric
[61]
type=friend
ip=192.168.1.194
context=testing

sip.conf:-
[general]
context=default
bindport=5060 
bindaddr=0.0.0.0
srvlookup=yes
disallow=all   
allow=alaw
allow=ulaw  
musicclass=default
dtmfmode = rfc2833

[63]
type=friend
context=testing ; context above where the extensions dialable by this 
are defined. 
username=63
secret=1234
host=dynamic
defaultip=192.168.1.192 ; ip address of this phone
canreinvite=no
callgroup=1 ; We are in caller groups 1
pickupgroup=1   ; We can do call pick-p for call group 1
;; rest of the sip users are configured in the same way.

Help will be very much appreciated. Kindly help. I am totally confused as to 
where the fault is. 






With warm regards.

Vivek J. Joshi.

[EMAIL PROTECTED]
Trikon electronics Pvt. Ltd.

--Truth springs from argument amongst friends.


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Re: [Asterisk-Users] Call parking on Polycom IP501

What firmware version did you use for the polycom phone ??

I just tried it on my IP600, and when I press the park button, it waits
for me to dial an extension number, then I press park again, and it just
hangs up the call.

Thanks,
Adam

On Tue, 2005-11-22 at 13:56 -0800, Anthony Rodgers wrote:
 Hi there,
 
 Instead of asking a question, I thought I'd post an answer. I got the  
 Polycom IP501 'Park' softkey working with * by doing the following:
 
 features.conf:
 
 [general]
 parkext = 1000
 parkpos = 1001-1009
 context = parkedcalls
 parkingtime = 120
 transferdigittimeout = 3
 courtesytone = beep
 
 Nothing unusual there. Here's the neat bit:
 
 extensions.conf:
 
 [internal] ; or whatever the relevant context is for you - it's usually  
 wherever your Polycom lives
 include = parkedcalls
 exten =  
 callpark,1,ParkAndAnnounce(pbx-transfer:PARKED|120|SIP/ 
 ${DIALEDPEERNUMBER}|internal,${DIALEDPEERNUMBER},1)
 
 By using SIP DEBUG, I discovered that the Polycom attempts to re-invite  
 the call to an extension called callpark. I couldn't get Park() to work  
 (it announces the stall number to the parked caller, instead of the  
 parker, for some reason), but using ParkAndAnnouce puts the parked call  
 on hold, hangs up the parker and then immediately calls them back with  
 an announcement of the stall number.
 
 Hope this helps someone out..
 
 Regards,

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[Asterisk-Users] Compatibilidade com PABX Intelbras

2005-11-24 Thread thiago

Bom dia,
 Gostaria de saber se alguém já conseguiu utilizar o pabx intelbras como
conexão pstn para o asterisk?!?!
 Já li algumas coisas sobre alterar o wcfxs.c mas nada que tenha surtido
efeito.

Agradeço a atenção.

Att,

 Thiago Rodrigues
-- 
#
#  THIAGO RODRIGUES DE SA
#  Assistencia Tecnica / Suporte
#  OPTICA TELECOM
#
#  Tel:. 55 24 2102-0800
#
#  http://www.opticatelecom.com.br
#

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Re: [Asterisk-Users] Poor sounds on Adtran 750

2005-11-24 Thread Chris Mason (Lists)

I have had my linesman go over the lines on the pole and manhole and 
remake the connections, I have played with the milliwatt generator to 
dial out and back to another number, measuring with ztmonitor to 
establish the levels, and I have played with the echo canceller 
settings. I still get hum and crappy audio with lots of  echo. When I 
listen to the lines with a butt set they sound clear, no hum.
Has any successfully used an Adtran 750 with PSTN lines? I have several 
and changing the unit or cards does not improve the quality.


--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 


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[Asterisk-Users] Re: Asterisk SIP architecture question

2005-11-24 Thread David Thomas

Does asterisk have support for SIP session timers?

David

On 11/24/05, Olle E. Johansson [EMAIL PROTECTED] wrote:
 Matt Riddell wrote:
  Kevin P. Fleming wrote:
 
 Matt Riddell wrote:
 
 
 So how does Asterisk know that the media stream has been disconnected
 between
 the two remote hosts?
 
 It doesn't... nor does any other SIP softswitch. See my other reply for
 a possible solution.
 
 
  I agree that you could code a fix, but saying my advice is bogus because
 you
  could code a fix for Asterisk to avoid it is slightly wrong.
 
  The fact remains, if you need *very* accurate cdr's then you either don't
 do
  canreinvite=yes for the peer or you code something so that Asterisk
 notices
  that the rtp has stopped.  The fact remains that without these, the most
  accurate CDR is going to come from the provider.
 

 If the audio goes through asterisk without re-invites, you could use the
 rtptimeouts to detect a dead phone.

 /O
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RE: [Asterisk-Users] Eicon Diva Server query

2005-11-24 Thread Erik Slooff

written by David Waugh
 Hi Avi,

 I added a bit the Asterisk wiki to explain hopefully more clearly how to
 get it installed.

 Please have a look at:
 http://www.voip-info.org/wiki-Asterisk+Eicon+Diva+CAPI+ISDN

 Any feedback appreciated.

 After you have installed the Diva Server drivers, please install the
 chan_capi_cm channel.

 Cheers
 David

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Behalf Of Avi Miller
 Sent: 23 November 2005 20:52
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Eicon Diva Server query


 David Waugh wrote:
 Yes, you can use the Eicon Diva Range with 2.6 Kernels

 Another question, considering the card should arrive tomorrow and I'd
 like to try my hand at setting it up this weekend: Do I need to BRIstuff
 Asterisk to get the Eicon Diva V-4BRI to work, or should I just need
 chan_capi-cm?

 Thanks,
 Avi

I would like to suggest one small addition for clarity:
you will *need* to have isdn4linux and capi4linux installed on your system
in order to get chan_capi-cm installed.

Erik

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Re: [Asterisk-Users] Eicon Diva Server query

2005-11-24 Thread Sergio Chersovani


Erik Slooff ha scritto:


I would like to suggest one small addition for clarity:
you will *need* to have isdn4linux and capi4linux installed on your system
in order to get chan_capi-cm installed.
 


You just need the capi20 lib in order to use the chan_capi
wget 
ftp://ftp.isdn4linux.de/pub/isdn4linux/CVS-Snapshots/isdn4k-utils-CVS-2005-10-28.tar.bz2


tar xvjf isdn4k*bz2
cd isdn4*
./configure
make
make install

that's all

Sergio
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Re: [Asterisk-Users] Eicon Diva Server query

2005-11-24 Thread Erik Slooff

I would like to suggest one small addition for clarity:
you will *need* to have isdn4linux and capi4linux installed on your
 system
in order to get chan_capi-cm installed.


 You just need the capi20 lib in order to use the chan_capi
 wget
 ftp://ftp.isdn4linux.de/pub/isdn4linux/CVS-Snapshots/isdn4k-utils-CVS-2005-10-28.tar.bz2

 tar xvjf isdn4k*bz2
 cd isdn4*
 ./configure
 make
 make install

 that's all

 Sergio

Great, that's clear for me now.
Maybe a good idea to add this to the wiki page.

Erik

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Re: RE : [Asterisk-Users] In France asterisk never detect hang up. Why ?

2005-11-24 Thread Faris Raouf


[EMAIL PROTECTED] wrote:

Hello everybody  :-)

This are my first line french zapata.conf settings.
I have 3 like this, with only rx/tx gain a little bit different levels.
Running well.
Best Regards,
Francois BERGERET,
France.

usecallerid=yes
hidecallerid=no
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
rxgain=2
txgain=6
group=1
callgroup=1
pickupgroup=1
immediate=no
busydetect=yes
busycount=3
busypattern=500,500
signalling = fxs_ks
channel = 1

-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de asterisk user
dupont
Envoyé : vendredi 18 novembre 2005 13:33
À : asterisk-users@lists.digium.com
Objet : [Asterisk-Users] In France asterisk never detect hang up. Why ?


Hello.

I am sorry my english is not good at all.

When i have a call from a fxo port of a tdm400p, asterisk waits one minute
before detecting that the caller has hang up his phone.

I have in my extension conf :
answer
background  (the prompt is 40 second long)
dial (on fxs port)  confgured for 30 seconds ringing.

if the caller hang up at the begining of the background prompt, asterisk
waits until he make ring the phone on the dial command for the all 30
secondes before detecting the hang up.

Do you know if there is a way to repair that ?

here is what i see on asterisk when the caller hang up IMMEDITALY after the
test prompt begins :

*CLI -- Starting simple switch on 'Zap/4-1'
-- Executing Answer(Zap/4-1, ) in new stack
-- Executing NoOp(Zap/4-1, 0675458745) in new stack
-- Executing Set(Zap/4-1, TIMEOUT(response)=20) in new stack
-- Response timeout set to 20
-- Executing BackGround(Zap/4-1, barge) in new stack
-- Playing 'test' (language 'fr')
-- Executing Dial(Zap/4-1, Zap/2|0675458745|30) in new stack
-- Called 2
-- Zap/2-1 is ringing
-- Zap/2-1 is ringing
-- Zap/2-1 is ringing
-- Zap/2-1 is ringing
-- Zap/2-1 answered Zap/4-1
-- Attempting native bridge of Zap/4-1 and Zap/2-1
-- Hungup 'Zap/2-1'
  == Spawn extension (reseau, s, 5) exited non-zero on 'Zap/4-1'
-- Executing Hangup(Zap/4-1, ) in new stack
  == Spawn extension (reseau, h, 1) exited non-zero on 'Zap/4-1'
-- Hungup 'Zap/4-1'


In my zapata.conf i have :

language=fr
default=fr
relaxdtmf=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
cidsignalling=v23
usecallerid=yes
group = 1
context=reseau
signalling=fxs_ks
callprogress=yes
busydetect=yes
callerid=asreceived
busycount=5
pulse=yes

In my zaptel.conf i have :

loadzone=fr
defaultzone=fr
fxoks=1-3
fxsks=4


If anyone can see what is wrong he will really help me.

thank you.


Your English is better than my French :-)

Making the TDM400p detect hangups can be hard. I had it working OK with 
pre-1.2 versions, but now in 1.2 stable I'm also having some problems 
again. I'll investigate in more details eventually.


For now, the only thing I can suggest is that you add:

hanguponpolarityswitch=yes

in your zapata.conf

In the UK, hangups are signaled by a polarity switch, and since 
sometimes the UK and Europe do the same thing, I'm hoping this will be 
the case for you too.


However, even with this option enabled, like I say, I'm having some 
small problems with 1.2 stable. I hope to have time this weekend to 
investigate and see what is going on.


Faris.



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Re: [Asterisk-Users] Eicon Diva Server query

2005-11-24 Thread Sergio Chersovani

Ok, pay attention to /dev/capi20 device it must exists with the right 
permissions



You just need the capi20 lib in order to use the chan_capi
wget
ftp://ftp.isdn4linux.de/pub/isdn4linux/CVS-Snapshots/isdn4k-utils-CVS-2005-10-28.tar.bz2

tar xvjf isdn4k*bz2
cd isdn4*
./configure
make
make install

that's all

Sergio
   



Great, that's clear for me now.
Maybe a good idea to add this to the wiki page.
 



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RE: [Asterisk-Users] Eicon Diva Server query

2005-11-24 Thread David Waugh

Hi Erik,
You'll have to excuse my ignorance here. But why is this?

I don't have isdn4linux and capi4linux installed but do have 
isdn4k-utils-devel-3.2-13.p1.1
isdn4k-utils-3.2-13.p1.1

installed.

Is this for the capi20.h needed for chan_capi to compile?

thanks David
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[Asterisk-Users] HELP! on disconnecting stale calls.

2005-11-24 Thread Paradise Dove

hi,
how can i hangup such calls without restarting asterisk?
the Zap channel on this case is busy for more than 7 hours
some logs are followed.

thanks,
Paradise Dove
-
Nov 23 16:59:49 NOTICE[3752] chan_sip.c: Disconnecting call
'SIP/2378-740f' for lack of RTP activity in 25788 seconds
Nov 23 16:59:49 NOTICE[3752] chan_sip.c: Disconnecting call
'SIP/2378-740f' for lack of RTP activity in 25788 seconds
Nov 23 16:59:50 NOTICE[3752] chan_sip.c: Disconnecting call
'SIP/2378-740f' for lack of RTP activity in 25789 seconds
Nov 23 16:59:51 NOTICE[3752] chan_sip.c: Disconnecting call
'SIP/2378-740f' for lack of RTP activity in 25790 seconds
Nov 23 16:59:51 NOTICE[3752] chan_sip.c: Disconnecting call
'SIP/2378-740f' for lack of RTP activity in 25790 seconds
Nov 23 16:59:52 NOTICE[3752] chan_sip.c: Disconnecting call
'SIP/2378-740f' for lack of RTP activity in 25791 seconds
Nov 23 16:59:52 NOTICE[3752] chan_sip.c: Disconnecting call
'SIP/2378-740f' for lack of RTP activity in 25791 seconds
Nov 23 16:59:53 NOTICE[3752] chan_sip.c: Disconnecting call
'SIP/2378-740f' for lack of RTP activity in 25792 seconds
Nov 23 16:59:53 NOTICE[3752] chan_sip.c: Disconnecting call
'SIP/2378-740f' for lack of RTP activity in 25792 seconds
-
Channel  Location State   Application(Data)
Zap/15-1 [EMAIL PROTECTED]:1 Up  Bridged 
Call(SIP/2378-740f)
1 active channel
1 active call
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Re: [Asterisk-Users] Eicon Diva Server query

2005-11-24 Thread Sergio Chersovani


David Waugh ha scritto:

I don't have isdn4linux and capi4linux installed but do have 
isdn4k-utils-devel-3.2-13.p1.1

isdn4k-utils-3.2-13.p1.1
 

Those are old packages, I suggest you to uninstall it and manual compile 
the version I posted in a previous release


There are alot of changes in the newer versions

Sergio
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[Asterisk-Users] GUI and Asterisk Realtime

2005-11-24 Thread harry gaillac

Hello,

Is there a GUI to manage sip users and voicemail with
Asterisk Realtime .
Regards
Harry






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[Asterisk-Users] Lag in speech

2005-11-24 Thread Tim Stoop

Hi people,

I just rolled out my first attempt with Asterisk to get a working PBX.
I'm recieving my calls still through my ISDN connection. But I'm
getting a lag of almost 1 second between the both sides of the
conversation. What should be the first things to look at when trying
to solve this lag?

I'm using an Athlon 700MHz with Debian and the default Asterisk
packages from Debian. The connection is made with a HPC-S PCI ISDN
card, directly connected to the NT1 box. I'm using a Grandstream
HandyTone 486 to connect my Philips DECT handset to the VoIP server.
The HandyTone is connected through a 100Mbit switch with the server.

Any advice is appreciated!

--
Gegroet,
Tim
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RE: [Asterisk-Users] Help need to reset Adit 600 for Asterisk install

2005-11-24 Thread Steve Totaro

Did you get it?  I would like to take a whack at it if not.

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
 Sent: Wednesday, November 23, 2005 10:30 AM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: RE: [Asterisk-Users] Help need to reset Adit 600 for Asterisk
 install

 Does anyone know of a brute force that will work on a serial interface
 like
 hyperterminal?

 --Jim

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Steve
Totaro
 Sent: Wednesday, November 23, 2005 8:22 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] Help need to reset Adit 600 for Asterisk
 install

 Is the password limited to four digits like the Adtran 600 (I think)?

 Start plugging in numbers.  Only 10,000 possible combinations.

  -Original Message-
  From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
  Sent: Wednesday, November 23, 2005 9:59 AM
  To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
  Subject: RE: [Asterisk-Users] Help need to reset Adit 600 for
Asterisk
  install

  Thanks Jerry,

  I have called Carrier Access and they can reset the password but for
a
  considerable fee.   We have serial access but after it boots it
  immediately
  asks for a username and password.  We have the username but the
 password
  is
  not what it is suppose to be.   There's a reset switch on the
 faceplate
  but
  I think the LOCAL SET is OFF and that is why it doesn't respond.
 Their
  manual says the Reset switch is not under the control of LOCAL SET,
 yet it
  doesn't seem to work.  Well, we might not know the proper boot
 sequence.
  It
  contains flash memory and there is a timing that important to that
 reset
  procedure.  Anyone's help is much appreciated.

  --Jim

  -Original Message-
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] On Behalf Of Jerry
 Jones
  Sent: Wednesday, November 23, 2005 7:40 AM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [Asterisk-Users] Help need to reset Adit 600 for
Asterisk
  install

  Not sure but are you connecting via serial or ehternet? Seems to be
 the
  serial had a way to do this easily on bootup. Otherwise I would be
  interested for future reference. Carrier Access does have a good
 support
  team, just need to know your serial number.

  On Nov 23, 2005, at 12:13 AM, [EMAIL PROTECTED] [EMAIL PROTECTED]
 wrote:

   Looking for a way to hard reset a ADIT 600 just purchased used.
   But it
   seems to have a master password already set.  We've tried the
front
   reset but maybe we don't have the right sequence of boot order.
Any
   help would be much appreciated?  - Jim

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Re: [Asterisk-Users] app_icd anyone? on 1.2?

2005-11-24 Thread Greg Boehnlein

On Tue, 22 Nov 2005, Lenz wrote:
 
 I also have never found anybody running an Asterisk system using app_icd.  
 Maybe app_queue is now after all flexible enough to be used in most cases.  
 Anybody else using different apps for Asterisk call centre applications?

I suspect that since the authors of ICD are no longer really submitting 
patches to Asterisk, that ICD for Asterisk is probably end of life, unless 
someone wants to pick up and take on maintenance of the code. As far as I 
know, ICD was never officially accepted into the mainline tree and was 
always kept as an outside project.

-- 
Vice President of N2Net, a New Age Consulting Service, Inc. Company
 http://www.n2net.net Where everything clicks into place!
 KP-216-121-ST



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Re: [Asterisk-Users] PhoneCALL version 2.7-RC1 Released!

2005-11-24 Thread Doug Lytle


Benoît Mérouze wrote:


Dustin Wildes wrote:


Hello Everyone!
For all of you PhoneCALL users, we have a treat for you today as 
PhoneCALL 2.7-RC1 has been released!



Demo with demo/demo doesn't work.
Is that possible to have a look to PhoneCALL without installing it 
(use the demo or screenshots)?



They must have fixed it, because I just logged in.  Looks nice, will have to 
give it a try this long holiday weekend.

Doug

--
Ben Franklin quote:

Those who would give up Essential Liberty to purchase a little Temporary Safety, 
deserve neither Liberty nor Safety.


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Re: [Asterisk-Users] v1-2 install mkdep loop

2005-11-24 Thread Greg Boehnlein

On Mon, 21 Nov 2005, Bob Knight wrote:

 Just pulled a v1-2 onto a system that was running a v1-0.
 
 Zaptel and libpri, build and install just fine.
 Building asterisk is fine.
 But when I try to do a make install on asterisk, it goes into an
 infinite loop doing on .depend doing: build_tools/mkdep
 
 I did the same thing on another box the other day with a different pull
 and did not have any problems.  Do you think this is something related
 to this box?

Hi Bob! Long live the PM3!
This is an issue that many many people have been running into, and 
has been discussed on the dev list.

Check the following:

http://lists.digium.com/pipermail/asterisk-dev/2005-November/016509.html

I'm not sure there is a specific fix, although there are many suggestions 
in that thread.

-- 
Vice President of N2Net, a New Age Consulting Service, Inc. Company
 http://www.n2net.net Where everything clicks into place!
 KP-216-121-ST



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Re: [Asterisk-Users] Lag in speech

2005-11-24 Thread Thor Atle Rustad

I had the same problem with ISDN. I actually got the last second or so of the previous call played back at the beginning of each call. There is a patch for this problem. I wish I could remember the name of the person who sent it to me. Maybe she will contact you if she sees this post. I will also have a look in my old mail later today.

Thor
On 11/24/05, Tim Stoop [EMAIL PROTECTED]
wrote:
Hi people,I just rolled out my first attempt with Asterisk to get a working PBX.I'm recieving my calls still through my ISDN connection. But I'm
getting a lag of almost 1 second between the both sides of theconversation. What should be the first things to look at when tryingto solve this lag?I'm using an Athlon 700MHz with Debian and the default Asterisk
packages from Debian. The connection is made with a HPC-S PCI ISDNcard, directly connected to the NT1 box. I'm using a GrandstreamHandyTone 486 to connect my Philips DECT handset to the VoIP server.The HandyTone is connected through a 100Mbit switch with the server.
Any advice is appreciated!--Gegroet,Tim___--Bandwidth and Colocation sponsored by
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RE: [Asterisk-Users] Asterisk 1.0/1.2 on cobalt Raq2-4

2005-11-24 Thread Greg Boehnlein

On Mon, 21 Nov 2005, Jonathan k. Creasy wrote:

 I've thought about doing that as I have a few spare also. I would use
 the raq4 I think. 
 
 Let me know if you have any trouble with it.

What you may want to do (I have several of these) is see if you can 
re-install the new Centos + BlueQuartz (GPL'd Raq GUI) ISO onto a drive 
and get it to boot in a Raq.

http://www.nuonce.net/bq-cd.php

-- 
Vice President of N2Net, a New Age Consulting Service, Inc. Company
 http://www.n2net.net Where everything clicks into place!
 KP-216-121-ST



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RE: [Asterisk-Users] v1-2 install mkdep loop

2005-11-24 Thread Lee Archer

I found running a later kernel and source code fixed it.  I had it on
Fedora Core 3 using kernel 2.6.9 but after updating to 2.6.12 the
problem went away.

Lee 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Greg
Boehnlein
Sent: 24 November 2005 14:16
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] v1-2 install mkdep loop

On Mon, 21 Nov 2005, Bob Knight wrote:

 Just pulled a v1-2 onto a system that was running a v1-0.
 
 Zaptel and libpri, build and install just fine.
 Building asterisk is fine.
 But when I try to do a make install on asterisk, it goes into an 
 infinite loop doing on .depend doing: build_tools/mkdep
 
 I did the same thing on another box the other day with a different 
 pull and did not have any problems.  Do you think this is something 
 related to this box?

Hi Bob! Long live the PM3!
This is an issue that many many people have been running into,
and has been discussed on the dev list.

Check the following:

http://lists.digium.com/pipermail/asterisk-dev/2005-November/016509.html

I'm not sure there is a specific fix, although there are many
suggestions in that thread.

-- 
Vice President of N2Net, a New Age Consulting Service, Inc. Company
 http://www.n2net.net Where everything clicks into place!
 KP-216-121-ST



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Re: [Asterisk-Users] PhoneCALL version 2.7-RC1 Released!

2005-11-24 Thread Dustin Wildes


Doug Lytle wrote:

They must have fixed it, because I just logged in.  Looks nice, will 
have to give it a try this long holiday weekend.


Doug

Hey Doug - yes, it was fixed this morning - we'd purged all the old demo 
data  forgot to re-create the demo account.
We've already gotten quite a few feature requests (like real-time status 
events for accounts, fax monitor,  an interface to the backend logging 
security) that we're getting ready to put in place.
Just keep in mind it's an RC1, so there maybe a few remaining 
bugs/issues which we're hoping to gain alot of feedback in the next week 
or so as we prepare for a -stable release.  We'd love to hear your input 
as you try it out!  :-)


Fixes are usually very quick as the codebase is rather easy to 
understand and follow since it's all in PHP/Smarty - all of the core DB 
functions should be (there are few sections that still do DB function 
directly) in the libs/accounts.php class. 

If you want to use Dreamweaver to edit the templates, we posted the 
SMARTY extension we use for Dreamweaver.  It works with both MX  2004 
that we've tried.

You can find it in the '3rd party' section of the downloads.


--Dustin


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Re: [Asterisk-Users] VoIPJet Support Contact

2005-11-24 Thread Mark Hulber

I'm all for criticism where it's due but I'm sure for all the bashing of 
Voipjet going on in this thread I'm sure there are just as many 
non-users who are generally happy with the service they provide and 
the price at which they provide it.


I for one am also a customer of Verizon, a fact I'd rather not 
advertise in case anyone might get the false impression I am happy with 
the service they provide and the price at which they provide it.


I don't think any of the VoIP wholesalers I deal with provide stellar 
customer service.  Contrary to the bigger telco's, when you do finally 
get their attention they do their best to resolve your problem.  Those 
that just really don't get it (remember LiveVoIP?) don't last.  
Otherwise, I think many of them are people like many of us who are 
trying to find a place in a difficult market.


If you want wholesale termination/origination with an SLA attached then 
you're going to have to pay for it.


MARK.

Chris Mason (Lists) wrote:



NON-DISCLOSURE: ALL CUSTOMERS USING VOIPJET'S SERVICE ARE SPECIFICALLY
PROHIBITED FROM DISCLOSING TO OTHERS THAT THEY USE VOIPJET'S SERVICE,
 


I use Voipjet,
I have used Voipjet...

Did I mention I use Voipjet?

I'd like to teach the world to sing (about using Voipjet)...

So sue me Voipjet, or better still, refund the outstanding balance so 
I can use it with a service that doesn't make people agree to stupid 
unenforcable rules. Another LiveVoip in the making.



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[Asterisk-Users] Don't Outgoing call with Zap

I have a QuadBRI card installed, and I received the call incoming, but I don't place call outgoing.  Asterisk show this message:  Executing Dial("SIP/101-a440", "ZAP/g1/3472543320|60") in new stack  -- Requested transfer capability: 0x00 - SPEECH  -- Called g1/3472543320 Nov 24 15:43:14 WARNING[10258]: chan_zap.c:6511 handle_init_event: Detected alar m on channel 2: Red Alarm Nov 24 15:43:14 WARNING[10258]: chan_zap.c:1586 zt_disable_ec: Unable to disable echo cancellation on channel 2 Nov 24 15:43:14 NOTICE[10254]: chan_zap.c:8451 pri_dchannel: PRI got event: Alar m (4) on Primary D-channel of span 1 Nov 24 15:43:14 NOTICE[10254]: chan_zap.c:8458 pri_dchannel: pri_shutdown Nov 24 15:43:14 NOTICE[10258]: chan_zap.c:6506 handle_init_event: Alarm cleared on channel 2 Nov 24 15:43:14 NOTICE[10254]: chan_zap.c:8451 pri_dchannel: PRI got event: No m ore alarm (5) on
  Primary
 D-channel of span 1 Nov 24 15:43:14 WARNING[10258]: chan_zap.c:6511 handle_init_event: Detected alar m on channel 1: No Alarm Nov 24 15:43:14 WARNING[10258]: chan_zap.c:1586 zt_disable_ec: Unable to disable echo cancellation on channel 1 Nov 24 15:43:14 NOTICE[10258]: chan_zap.c:6506 handle_init_event: Alarm cleared on channel 1  -- Hungup 'Zap/1-1'  == No one is available to answer at this time (1:0/0/0)  -- Executing Hangup("SIP/101-a440", "") in new stack  == Spawn extension (local, 3472543320, 2) exited non-zero on 'SIP/101-a440'  Why?  My extension.conf is: [general] static=yes writeprotect=no  [globals] TELEIN=SIP/200  ; ; CONTESTO PER LE CHIMATE IN INGRESSO LOCALI E IN USCITA * ; [local] <
 br>
 exten = _x.,1,Dial(ZAP/g1/${EXTEN},60) exten = _x.,2,Hangup   ;** ; CONTESTO PER LE CHIAMATE IN INGRESSO DA UNIVOICE * ;** [out-sip] exten = 101,1,Dial(${TELEIN},20,rt) exten = 101,3,Hangup  [isdn_incoming] exten = _x.,1,Dial(SIP/200,60) exten = _x.,2,Hangup   My zapata.conf is  [channels] switchtype = euroisdn  signalling = bri_cpe_ptmp pridialplan = local prilocaldialplan = local nationalprefix = 0 internationalprefix = 00  echocancel = yes  context=isdn_incoming group = 1 channel = 1-2 group = 2 channel = 4-5  Thanks 
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[Asterisk-Users] Re: Not receiving fax

2005-11-24 Thread Wayne Gemmell

On Thursday 24 November 2005 12:49, Kristof Hardy wrote:
 Yes, make a 'default' to go directly to your fax-receive macro. (rxfax
 witht the parameters)

 At least you should hear a 'fax' answering.
Yes, I hear a fax answering, so at least I know its working.
-- 
Regards

Wayne Gemmell

Work: +27 11 894 2530
Fax : +27 11 894 4081
Cell: +27 83 666 3325
Email   : [EMAIL PROTECTED]
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[Asterisk-Users] jittering with Iax2 and Meetme on Asterisk 1.2.0

2005-11-24 Thread Steven Langley

Title: jittering with Iax2 and Meetme on Asterisk 1.2.0






Hi

I have been using Asterisk 1.0.9 fairly successfully. I have Iax2 softphones based on the IaxClient library that are dialing into Meetme conferences. I am using a Zaptel card as a timing source.

I am now trying to migrate to Asterisk 1.2.0, mainly because of the alleged improved jitterbuffer implementation. I have installed 1.2.0 (Zaptel and Asterisk) and am running it on a 100 mbit LAN. I am dialing in with the same softphone (as the other server with Asterisk 1.0.9), but experience consistently bad jitter, both when jitterbuffer=no and when jitterbuffer=yes. I have run zttest and am getting pretty much 100% accuracy from the card.

Does anyone have any ideas what the problem could be?

Many thanks

Steven


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Re: [Asterisk-Users] Re: Asterisk SIP architecture question

2005-11-24 Thread Olle E. Johansson

David Thomas wrote:
 Does asterisk have support for SIP session timers?
 
No.

/O
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[Asterisk-Users] What does it mean?

Hello,

I have compiled asterisk cvs under freebsd, no problems.

When starting asterisk, I get :

[res_config_mysql.so] = (MySQL RealTime Configuration Driver)
/libexec/ld-elf.so.1: /usr/lib/asterisk/modules/res_config_mysql.so:
Undefined symbol ast_config_load

What's wrong?

Olivier

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RE: [Asterisk-Users] What does it mean?

2005-11-24 Thread harry gaillac

Hello,

Read the Makefile in apps.
Harry
--- Olivier Taylor [EMAIL PROTECTED] a écrit
:

 Hello,
 
 I have compiled asterisk cvs under freebsd, no
 problems.
 
 When starting asterisk, I get :
 
 [res_config_mysql.so] = (MySQL RealTime
 Configuration Driver)
 /libexec/ld-elf.so.1:
 /usr/lib/asterisk/modules/res_config_mysql.so:
 Undefined symbol ast_config_load
 
 What's wrong?
 
 Olivier
 
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Re: [Asterisk-Users] SIP Extension behind NAT, Asterisk on a public domain

2005-11-24 Thread Tom Rymes


On Nov 23, 2005, at 11:04 AM, Manny A. Wise wrote:

[snip]
Well, as the user stated on the original message, the asterisk  
server is behind a NAT and the client is also behind a NAT….


if you make it work just by opening ports, let me know..I have  
never been able to get it to work, that’s why I don’t use sip, just  
plain iax2 for everything… J


Manny


Manny,

I have this working as I write this. (I just hung up the phone.) In  
fact, I brought a Cisco 7940G to a completely unknown nat-ed network  
the other day, plugged it in and started making calls right away.  
Here's the setup I have for this specific configuration:


1.) Asterisk server behind NAT. (Setup as DMZ on Linksys WRT54G, but  
it's still NAT. I just don't have to forward ports this way)
2.) externip, localnet, nat settings configured in the sip.conf file  
(sip_nat.conf for [EMAIL PROTECTED])
3.) Cisco phone (or whatever SIP UA you choose) configured for NAT  
(via the SIPMAC.cnf file for Cisco)

4.) Lather, rinse, repeat if necessary

Hopefully that will work for you. I'd rather use IAX and avoid these  
problems altogether, but I have yet to find an IAX hardphone I am  
willing to use. In fact, for softphone use, I do indeed use IAX via  
LoudHush for the mac. (Great piece of software, BTW. No connection  
here, just a happy user...)


Tom


Tom Rymes
Cascade Link Systems
www.cascadelinksystems.com
(603) 375-1414

Intelligent technology solutions for small businesses.


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[Asterisk-Users] CallerID not passing through to Polycom 500

2005-11-24 Thread Gary MacKay

I have a basic system working, except for callerid. The Polycom 500 just 
shows call from Business Line on the screen. Business Line is the 
name of the context that line is in. How do I get it to show the 
callerID on the screen instead? Yes, I have CallerID on that line and it 
works on a standard analog phone.

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Re: [Asterisk-Users] Re: sip URL peering

2005-11-24 Thread Klaus Darilion


Wolfgang S. Rupprecht wrote:

Klaus Darilion [EMAIL PROTECTED] writes:


There is a new ietf WG to come which deals with peering issues. It's
called SPEER (formerly VOIPEER)

The list archive is at
http://darkwing.uoregon.edu/~llynch/voipeer/

minutes from last ietf meeting:
http://www3.ietf.org/proceedings/05nov/minutes/voipeer.html



It looks interesting, but these things always seem to be scuttled or
reduced to glacial progress by the telecom interests.

VOIP peering isn't something that should require years of meeting to
make happen.


It's not that easy. If you want to have open SIP URIs (just like email 
is open for everybody) you will receive SPIT calls. E.g. the SPEER group 
tries to define rules for VoIP peering which allows authentication to 
enable open SIP URIs. (I won't open acces to my SIP URI if I can not 
verify the senders URI).


regards
klaus
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[Asterisk-Users] H323 to H323 calls problem

2005-11-24 Thread Javier Oviedo

This is my network scheme:


h323 endpoint1 ... endpint10 = gk1 = gk2 = asterisk

GK1 configuration: routed mode
GK2 configuration: direct mode

How to obtain that rtp channels not through asterisk for h323 to h323 calls

Thanks in advance!

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RE : [Asterisk-Users] What does it mean?

Tes réponses sont aussi sybillines que tes questions :)

Olivier

-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de harry gaillac
Envoyé : jeudi 24 novembre 2005 16:45
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : RE: [Asterisk-Users] What does it mean?


Hello,

Read the Makefile in apps.
Harry
--- Olivier Taylor [EMAIL PROTECTED] a écrit
:

 Hello,
 
 I have compiled asterisk cvs under freebsd, no
 problems.
 
 When starting asterisk, I get :
 
 [res_config_mysql.so] = (MySQL RealTime
 Configuration Driver)
 /libexec/ld-elf.so.1:
 /usr/lib/asterisk/modules/res_config_mysql.so:
 Undefined symbol ast_config_load
 
 What's wrong?
 
 Olivier
 
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[Asterisk-Users] Voicemail notifications alwats sent as [EMAIL PROTECTED]

2005-11-24 Thread Andre Courchesne - Consultant


Hi,

 I have a problem with e-mail notifications. For some reason Asterisk 
does not use the serveremail configuration when sending e-mails 
notifications. it always send it using [EMAIL PROTECTED]


My configuration:
pbxskip=yes ; Don't put [PBX]: in the subject line
[EMAIL PROTECTED]
fromstring=Voicemail System ; Real name of email sender
maxmessage=180  ; max length of vm message
minmessage=3; Minimum length of a voicemail 
message in seconds
maxsilence=5; Wait for 5 silent seconds and end the 
voicemail

silencethreshold=128; What do we consider to be silence
skipms=3000 ; How many miliseconds 
to skip forward/back when rew/ff in message playback
review=yes  ; Allow sender to 
review/rerecord their message before saving it

operator=yes; Allow caller to press 0

Exim log when sending an e-mail:
Nov 24 11:44:21 privalodc exim[8501]: 2005-11-24 11:44:21 
1EfKCr-0002D5-N1 ** [EMAIL PROTECTED] R=dnslookup T=remote_smtp: 
SMTP error from remote mail server after RCPT 
TO:[EMAIL PROTECTED]: host mail.privalodc.com 
[207.115.102.XX]: 504 [EMAIL PROTECTED]: Sender address rejected: need 
fully-qualified address


Any ideas?

Andre

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RE: RE : [Asterisk-Users] What does it mean?

2005-11-24 Thread harry gaillac

Je ne connais pas la signification de sybillines.
Harry
--- Olivier Taylor [EMAIL PROTECTED] a écrit
:

 Tes réponses sont aussi sybillines que tes questions
 :)
 
 Olivier
 
 -Message d'origine-
 De : [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] De
 la part de harry gaillac
 Envoyé : jeudi 24 novembre 2005 16:45
 À : Asterisk Users Mailing List - Non-Commercial
 Discussion
 Objet : RE: [Asterisk-Users] What does it mean?
 
 
 Hello,
 
 Read the Makefile in apps.
 Harry
 --- Olivier Taylor [EMAIL PROTECTED] a
 écrit
 :
 
  Hello,
  
  I have compiled asterisk cvs under freebsd, no
  problems.
  
  When starting asterisk, I get :
  
  [res_config_mysql.so] = (MySQL RealTime
  Configuration Driver)
  /libexec/ld-elf.so.1:
  /usr/lib/asterisk/modules/res_config_mysql.so:
  Undefined symbol ast_config_load
  
  What's wrong?
  
  Olivier
  
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http://lists.digium.com/mailman/listinfo/asterisk-users
  
 
 
 
   
 
   
   

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[Asterisk-Users] Re: Queue Callback - SOLVED

2005-11-24 Thread Tyler

Happy Thanksgiving everyone..  I added the following page to the Wiki
documenting how I solved this problem without having to hack with ICD or
any commercial offerings.

http://www.voip-info.org/wiki/view/Asterisk+Queue+Callback

Hope it can help somebody out.

tf.

 -Forwarded Message-
  From: Tyler [EMAIL PROTECTED]
  To: asterisk-users@lists.digium.com
  Subject: Queue Callback
  Date: Tue, 15 Nov 2005 08:39:27 -0500
  
  Hello,
  
  Does anyone have any information on configuring app_icd (or know of any
  way to do it with the dialplan) that would allow a user holding in a
  queue to hang up, and have the system call them back when their place in
  line comes up next?  
  
  I can (obviously) allow them to '0' out to voicemail or something, but I
  can only find vague references to app_icd and 'OrderlyQ' for doing what
  I want to do...
  
  Anyone?
  
  Bueller? ;-)
  
  Thanks
  
  tf.
  
  

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RE: RE : [Asterisk-Users] What does it mean?

2005-11-24 Thread Dave Cotton

On Thu, 2005-11-24 at 17:54 +0100, harry gaillac wrote:
 Je ne connais pas la signification de sybillines.

http://www.village-justice.com/forum/viewtopic.php?t=1224start=0sid=964c2c9a1cd842eaca284be8899028a8


-- 
Dave Cotton [EMAIL PROTECTED]

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Re: [Asterisk-Users] SIP Extension behind NAT, Asterisk on a public domain

2005-11-24 Thread Bharath

I found out that I have a faulty Belkin Router which was causing the
problem. I tried forwarding ports as well as DMZ'd the Sip device but
still could'nt not hear the voice. So i plugged the sip device directly
to the cable modem it worked fine. So my guess is that though I
have set up the router to forwards port to the sip device there is
something happening at the router that is blocking the RTP ports
(1-2).
Thanks
On 11/24/05, Tom Rymes [EMAIL PROTECTED] wrote:
On Nov 23, 2005, at 11:04 AM, Manny A. Wise wrote:[snip] Well, as the user stated on the original message, the asterisk server is behind a NAT and the client is also behind a NAT…. if you make it work just by opening ports, let me know..I have
never been able to get it to work, that's why I don't use sip, just plain iax2 for everything… J MannyManny,I have this working as I write this. (I just hung up the phone.) In
fact, I brought a Cisco 7940G to a completely unknown nat-ed networkthe other day, plugged it in and started making calls right away.Here's the setup I have for this specific configuration:1.) Asterisk server behind NAT. (Setup as DMZ on Linksys WRT54G, but
it's still NAT. I just don't have to forward ports this way)2.) externip, localnet, nat settings configured in the sip.conf file(sip_nat.conf for [EMAIL PROTECTED])3.) Cisco phone (or whatever SIP UA you choose) configured for NAT
(via the SIPMAC.cnf file for Cisco)4.) Lather, rinse, repeat if necessaryHopefully that will work for you. I'd rather use IAX and avoid theseproblems altogether, but I have yet to find an IAX hardphone I am
willing to use. In fact, for softphone use, I do indeed use IAX viaLoudHush for the mac. (Great piece of software, BTW. No connectionhere, just a happy user...)TomTom Rymes
Cascade Link Systemswww.cascadelinksystems.com(603) 375-1414Intelligent technology solutions for small businesses.___
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[Asterisk-Users] Re: Asterisk not picking up calls.

2005-11-24 Thread David Yat Sin









This is usually a problem one of the pair not physically
disconnected, i.e loose plug/socket connection. Try
to replace your connectors and test your cable



David
Yat Sin

Sangoma
Technologies

(905) 474-1990
x119

(800)
388-2475 x119

Fax:
(905) 474 9223

MSN:
[EMAIL PROTECTED]

Email: [EMAIL PROTECTED]m

Website:
www.sangoma.com








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Re: [Asterisk-Users] :: Success Case = Motorola 62802-51 as FXO device::

2005-11-24 Thread Nestor A. Diaz


Walter Willis wrote:


not work fine


Actually it is recognized as an x100p device:

Nov 21 19:54:34 asterix kernel: Zapata Telephony Interface Registered on 
major 196

Nov 21 19:54:34 asterix kernel: PCI: Found IRQ 5 for device 00:0d.0
Nov 21 19:54:34 asterix kernel: wcfxo: DAA mode is 'FCC'
Nov 21 19:54:34 asterix kernel: Found a Wildcard FXO: Wildcard X100P
Nov 21 19:54:34 asterix kernel: Registered tone zone 0 (United States / 
North America)


i have been able to call from the outside and the default greeting sound 
good, but it can not recognize tones, when program the extension to dial 
an inside line the sound is very bad, too much noise !!! i think the 
problem is with full duplex.


it will be nice to investigate if we can modify the sources to make this 
chipset (62802-52) work with asterisk in a nice way, i have been dealing 
with rxgain and txgain in order to tune the card, but i have failed, the 
sound is still bad.


62802 is one of the chipset that it is still available on the market, it 
is not designed to compete against digium analog card, is designed to 
introduce people on the voip field, for this it is important to be 
supported, think of  PC vs. Apple, the more people will use Asterisk the 
best the business will become.


Somebody have deal with zapata sources in order to make some changes and 
make that chipset works ? does anyone have tried newer intel modem 
chipset with asterisk ? they work ? the only chipset that works for me 
was the ambient md3200, have some echo problems but with echo 
chancelation and training things get better after a few seconds.


What are the requeriments for a modem chipset to be supported on asterisk ?

p.d. i am searching for ambiend md 3200 cards, anybody know where i can 
buy them ? at a reasonable price off course.


Thanks everyone.

--
Nestor A. Diaz
Ingeniero de Sistemas
Tel. +57 1-600-5490 x 211
Cel. +57 316-227-3593
Tel. SIP: sip:[EMAIL PROTECTED]
Email/MSN: [EMAIL PROTECTED]
http://www.tiendalinux.com/
Bogota, Colombia 


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RE : RE : [Asterisk-Users] What does it mean?

SIBYLLIN, INE. adj. Qui appartient aux sibylles. Il n'est guère usité au
sens propre que dans ces locutions : Les oracles, les livres, les vers
sibyllins, Les oracles, les livres, les vers des sibylles. 
Il signifie au figuré Qui est mystérieux obscur, dont le sens est difficile
à saisir. Il m'a répondu en termes sibyllins. Des paroles sibyllines. Un
langage sibyllin.


-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de harry gaillac
Envoyé : jeudi 24 novembre 2005 17:54
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : RE: RE : [Asterisk-Users] What does it mean?


Je ne connais pas la signification de sybillines.
Harry
--- Olivier Taylor [EMAIL PROTECTED] a écrit
:

 Tes réponses sont aussi sybillines que tes questions
 :)
 
 Olivier
 
 -Message d'origine-
 De : [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] De
 la part de harry gaillac
 Envoyé : jeudi 24 novembre 2005 16:45
 À : Asterisk Users Mailing List - Non-Commercial
 Discussion
 Objet : RE: [Asterisk-Users] What does it mean?
 
 
 Hello,
 
 Read the Makefile in apps.
 Harry
 --- Olivier Taylor [EMAIL PROTECTED] a
 écrit
 :
 
  Hello,
  
  I have compiled asterisk cvs under freebsd, no
  problems.
  
  When starting asterisk, I get :
  
  [res_config_mysql.so] = (MySQL RealTime
  Configuration Driver)
  /libexec/ld-elf.so.1:
  /usr/lib/asterisk/modules/res_config_mysql.so:
  Undefined symbol ast_config_load
  
  What's wrong?
  
  Olivier
  
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Re: [Asterisk-Users] Lag in speech

2005-11-24 Thread Thor Atle Rustad

I found the mail from Pauline Middelink!

filename: hfc_pci.c.diff



--- /root/hfc_pci.c Wed Aug  7 15:31:24 2002
+++ /usr/src/linux/drivers/isdn/hisax/hfc_pci.c Thu Oct 31 10:18:05 2002
@@ -270,8 +270,16 @@
if (fifo_state)
cs-hw.hfcpci.fifo_en ^= fifo_state;
Write_hfc(cs, HFCPCI_FIFO_EN, cs-hw.hfcpci.fifo_en);
-   bzt-za[MAX_B_FRAMES].z1 = B_FIFO_SIZE + B_SUB_VAL - 1;
-   bzt-za[MAX_B_FRAMES].z2 = bzt-za[MAX_B_FRAMES].z1;
+   /* Notice the z2 is readonly, and could be active when we enter this
+* function. (I.e. changing.) When we now reset z1 to MAXSIZE, the
+* FIFO thinks there is data and runs it when re-enabled...
+* To prevent this from happening, we make z1 ONE higher than z2, so
+* when the FIFO gets re-enabled, it thinks it only has to send a
+* single byte, which hopefully nobody notices (1/8000 second?)
+* (Pauline Middelink - 2002) */
+   bzt-za[MAX_B_FRAMES].z1 = bzt-za[MAX_B_FRAMES].z2 + 1;
+   if (bzt-za[MAX_B_FRAMES].z1 = B_FIFO_SIZE + B_SUB_VAL)
+   bzt-za[MAX_B_FRAMES].z1 -= B_FIFO_SIZE;
bzt-f1 = MAX_B_FRAMES;
bzt-f2 = bzt-f1;  /* init F pointers to remain constant */
if (fifo_state)
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Re: [Asterisk-Users] TDM400 FXO port 1 only problem.

2005-11-24 Thread Kevin Hanson


Tom Vile wrote:


Everyone,

I have a TDM400 REV I Ver 1 board and am having an issue with 1 of the
4 FXO channels.  FXO 1 always has clicks, pops and echo but the others
are crystal clear all of the time.  The card is on its own IRQ zztest
shows 100% to 99.98%  and is getting 1000 int per second.  Its not
dropping interrupts either.

I ran FXOTUNE and it did nothing to fix the issue

It only is happening on FXO port 1

Anything else to try?

Thanks,
--
Tom Vile
 

I had the same problem w/ two boards.  Called Digium and they said they 
had a batch of bad boards go out where port 1 would exhibit the problems 
you describe.  They rma'd the boards for me and all is well.


Cheers,
Kevin
--
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http://www.optimacycomm.com
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RE : RE : [Asterisk-Users] What does it mean?