date:20040507

Also, a better alternative is MAD player. And there is a patch for
Asterisk that adds support for it.
http://bugs.digium.com/bug_view_page.php?bug_id=0001365

Michael.

brian k. west wrote:
We find that mpg123 0.59r works best.  mpg123 0.59s-mh4 = the devil.
 
What versions does everyone use without problems.
 
0.59r is PERFECT
 
bkw


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Re: [Asterisk-Users] asterisk-oh323, new version 0.6.1

Le jeu 06/05/2004 à 18:52, Michael Manousos a écrit :
 This new version (0.6.1) of asterisk-oh323 fixes the one-way audio
 problem of the previous release.

Hi, what is the difference between chan_h323 and asterisk-oh323? Are
they mutually exclusive? Is one better than the other?
chan_h323 came directly with my .deb package, and I am currently
compiling the CVS version of *, to test ast-oh323, so I may get some
answers then :-)
Thanks,
Paul

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Re: [Asterisk-Users] asterisk-oh323, new version 0.6.1



Paul Berger wrote:
Le jeu 06/05/2004  18:52, Michael Manousos a crit :

This new version (0.6.1) of asterisk-oh323 fixes the one-way audio
problem of the previous release.


Hi, what is the difference between chan_h323 and asterisk-oh323? Are
they mutually exclusive? Is one better than the other?
They are mutually exclusive because they try to do the same thing.

chan_h323 came directly with my .deb package, and I am currently
compiling the CVS version of *, to test ast-oh323, so I may get some
answers then :-)
OK, try it and let me know what you think.

Thanks,
Paul
Michael.

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Re: [Asterisk-Users] PRI to PRI fax pass through

2004-05-07 Thread Konrad Gorski

[EMAIL PROTECTED] wrote:



Hi

I have Asterisk with two E100P cards
One connected to PSTN and other to my local PBX
I'm running into problem with faxes.

Faxes are connected to PBX and asterisk should  just bridge the fax call
from one span to another.
Problem is that even if the fax image reach the over end 90% the fax
receive the transsmit error.
Its happend for incomming (PSTN-asterisk-PBX-fx) and outgoung
(fax-PBX-asterisk-PSTN) directions
dial string include the c option (clear channel)

TRUNK_El=Zap/g1

exten = _[0 1-9].,1,SetCallerID(${MyPreffix}${CALLERIDNUM})
exten = _[0 1-9].,2,Dial(${Zap/g1}/$[EXTEN],100,c)
exten = _[0 1-9].,3,Congestion
And also i have sometimes the echo (even the echocansel is yes in
zapata.conf)
May be it related to the fax also?
Any ideas is appreciated

Juri
 

maybe slips ?  I had similar problem.
check timing parameter of  E100P (zaptel.conf),
check interrupts, sharing is not allowed  (/proc/interrupts)
my 0.02$
Konrad


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Re: [Asterisk-Users] asterisk-oh323, new version 0.6.1

Le ven 07/05/2004 à 11:59, Michael Manousos a écrit :
 They are mutually exclusive because they try to do the same thing.

Why 2 different projects for the same goal? (i hope I'm not starting a
flame war :-))

 OK, try it and let me know what you think.

I seem to have a problem when compiling ast-oh323:

make[1]: Entering directory
`/home/paul/asterisk/asterisk-oh323-0.6.1/asterisk-driver'
gcc -Wall -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE
-I/home/paul/asterisk/asterisk-cvs-2004-05-07/include -I../wrapper -g -c
-o chan_oh323.o chan_oh323.c
chan_oh323.c: In function `oh323_call':
chan_oh323.c:1135: error: too few arguments to function
`ast_queue_hangup'
chan_oh323.c:1162: error: too few arguments to function
`ast_queue_control'
chan_oh323.c: In function `oh323_hangup':
chan_oh323.c:1193: error: too few arguments to function
`ast_queue_hangup'
chan_oh323.c: In function `oh323_read':
chan_oh323.c:1599: error: too few arguments to function
`ast_dsp_process'
chan_oh323.c: In function `oh323_answer':
chan_oh323.c:1826: error: too few arguments to function
`ast_queue_control'
make[1]: *** [chan_oh323.o] Error 1
make[1]: Leaving directory
`/home/paul/asterisk/asterisk-oh323-0.6.1/asterisk-driver'
make: *** [subdirs_all] Error 1

Any hint?
Thanks,
Paul

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[Asterisk-Users] Trouble compiling latest CVS

Hi there,

since a couple of days I can't seem to be able to compile CVS HEAD on 
RH7.2. On a RH7.3 machine with bison-1.35-1 it appears to be fine 
though... any advice?

Philipp


System: RH 7.2
bison-1.28-7

Related issue:
http://rpm.pbone.net/index.php3/stat/4/idpl/411535/com/bison-1.35-
6.i386.rpm.html

The error message:

bison ast_expr.y --name-prefix=ast_yy -o ast_expr.c
ast_expr.y:103: unrecognized: %locations
ast_expr.y:103:Skipping to next %
ast_expr.y:134: invalid @-construct
ast_expr.y:134: $. is invalid
ast_expr.y:134: invalid @-construct
ast_expr.y:134: $. is invalid
...
ast_expr.y:148: invalid @-construct
ast_expr.y:148: $. is invalid
ast_expr.y:148: invalid @-construct
ast_expr.y:148: $. is invalid
ast_expr.y:148: invalid @-construct
ast_expr.y:148: $. is invalid
ast_expr.y:148: invalid @-construct
ast_expr.y:148: $. is invalid
make: *** [ast_expr.c] Error 1


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[Asterisk-Users] Cisco 7940 microphone volume

2004-05-07 Thread Frederic Steinfels

When talking to me, people are complaining the volume was not high enough.

The phone only allows to change the volume of the speaker/earpiece. Is 
there an alternative solution? Is it possible to increase the volume in 
asterisk?

Frederic
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Re: [Asterisk-Users] Trouble compiling latest CVS

Hm...

 since a couple of days I can't seem to be able to compile CVS HEAD on 
 RH7.2. On a RH7.3 machine with bison-1.35-1 it appears to be fine 
 though... any advice?

Actually this doesn't seem to be related to bison - I can't even compile 
my old CVS-HEAD-05/03/04-19:58:33 anymore, getting the same error. 
Except for a CURL upgrade there was not major change on the system, at 
least not that I know of... I did do a clean checkout, but no 
improvement... am a bit puzzled...

Philipp


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Re: [Asterisk-Users] asterisk-oh323, new version 0.6.1



Paul Berger wrote:
Le ven 07/05/2004  11:59, Michael Manousos a crit :

They are mutually exclusive because they try to do the same thing.


Why 2 different projects for the same goal? (i hope I'm not starting a
flame war :-))
Because there are two (or even more) ways to solve the problem.
This topic has been discussed in the past several times. Check the
archives for details.


OK, try it and let me know what you think.


I seem to have a problem when compiling ast-oh323:
Use latest CVS asterisk.

make[1]: Entering directory
`/home/paul/asterisk/asterisk-oh323-0.6.1/asterisk-driver'
gcc -Wall -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE
-I/home/paul/asterisk/asterisk-cvs-2004-05-07/include -I../wrapper -g -c
-o chan_oh323.o chan_oh323.c
chan_oh323.c: In function `oh323_call':
chan_oh323.c:1135: error: too few arguments to function
`ast_queue_hangup'
chan_oh323.c:1162: error: too few arguments to function
`ast_queue_control'
chan_oh323.c: In function `oh323_hangup':
chan_oh323.c:1193: error: too few arguments to function
`ast_queue_hangup'
chan_oh323.c: In function `oh323_read':
chan_oh323.c:1599: error: too few arguments to function
`ast_dsp_process'
chan_oh323.c: In function `oh323_answer':
chan_oh323.c:1826: error: too few arguments to function
`ast_queue_control'
make[1]: *** [chan_oh323.o] Error 1
make[1]: Leaving directory
`/home/paul/asterisk/asterisk-oh323-0.6.1/asterisk-driver'
make: *** [subdirs_all] Error 1
Any hint?
Thanks,
Paul


Michael.

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Re: [Asterisk-Users] asterisk-oh323, new version 0.6.1

Can you send me off-list a full debug (-vvvcd) output of the call?

Michael.

Michael Niehren wrote:
Still one-way audio problems with version V0.6.1.

Hi Michael,

using asterisk as ISDN2H323-Gateway. Call from ISDN - Asterisk - H323 is now
ok, but in the other direction there is still only one-way audio. I hear 
nothing on the H323 side. 
The 2. thing is after cleareing the 1. Call i try again to get the phone 
number, but asterisk did not dial as you can see in the log.

Here is the log, maybe it help's:

Asterisk Ready.
*CLI -- Executing Dial(OH323/R32, Modem/ttyI1:892877) in new stack
-- Called ttyI1:892877
-- Modem[i4l]/ttyI1 answered OH323/R32
May  6 19:46:31 DEBUG[11276]: channel.c:2544 ast_channel_bridge: Got a 
FRAME_CONTROL (4) frame on channel OH323/R32
May  6 19:46:31 DEBUG[11276]: channel.c:2606 ast_channel_bridge: Bridge stops 
bridging channels OH323/R32 and Modem[i4l]/ttyI1
May  6 19:46:31 DEBUG[11276]: res_parking.c:423 ast_bridge_call: Read from 
OH323/R32 (4,4)
May  6 19:46:46 DEBUG[11276]: chan_modem_i4l.c:394 i4l_read: Value of escape 
is ^ (3)...
May  6 19:46:46 DEBUG[11276]: channel.c:2536 ast_channel_bridge: Didn't get a 
frame from channel: Modem[i4l]/ttyI1
May  6 19:46:46 DEBUG[11276]: channel.c:2606 ast_channel_bridge: Bridge stops 
bridging channels OH323/R32 and Modem[i4l]/ttyI1
-- Hungup 'Modem[i4l]/ttyI1'
  == Spawn extension (voip-h323, s, 1) exited non-zero on 'OH323/R32'
  0:26.435  H225 Answer:80fab28 H225Read error (0): 
  0:26.505 H323 Cleaner H323Connection 
ip$192.168.70.1:42812/32 terminated.
-- H.323 call 'ip$192.168.70.1:42812/32' cleared, reason 1 (Cleared by 
local user)
-- Hungup 'OH323/R32'

-- SECOND CALL
-- Executing Dial(OH323/R35, Modem/ttyI1:) in new stack
-- Called ttyI1:
  3:59.764  H225 Answer:80f9958 H225Read error (0): 
  3:59.794 H323 Cleaner H323Connection 
ip$192.168.70.1:42825/35 terminated.
-- H.323 call 'ip$192.168.70.1:42825/35' cleared, reason 7 (Remote user 
stopped calling)
-- Hungup 'Modem[i4l]/ttyI1'
  == Spawn extension (voip-h323, s, 1) exited non-zero on 'OH323/R35'
-- Hungup 'OH323/R35'

Greetings,
  Michael
Am Donnerstag, 6. Mai 2004 18:52 schrieb Michael Manousos:

Hello all,

This new version (0.6.1) of asterisk-oh323 fixes the one-way audio
problem of the previous release.
Download from the usual location:
http://www.inaccessnetworks.com/projects/asterisk-oh323
Regards,
Michael.
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--
./M
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[Asterisk-Users] modem (56k) call to PSTN

2004-05-07 Thread Harald B.

Hello,
i have a Winmodem (Softwaremodem) i know, this is a problem under linux.
But asterisk loaded a Modem channel.
What i wanted to know is, can i use this channel to make a PSTN call??
If yes, how can i do that. Which *.conf files do i have to change??
Has someone experience with that??
Kindly Regards
Harald 

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Re: [Asterisk-Users] mpg123 versions ?

 We find that mpg123 0.59r works best.  mpg123 0.59s-mh4 = the devil.
  
 What versions does everyone use without problems.
  
 0.59r is PERFECT

Been using mpg123-0.59q-1.i386.rpm since late September with no problems.

Not had a need to change; ain't broke, don't fix it.



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Re: [Asterisk-Users] asterisk-oh323, new version 0.6.1

Le ven 07/05/2004 à 13:18, Michael Manousos a écrit :
 Because there are two (or even more) ways to solve the problem.
 This topic has been discussed in the past several times. Check the
 archives for details.

Sorry, I should have started there...

 Use latest CVS asterisk.

I was using the latest CVS (v1-0_stable branch).
I'll try the dev CVS and let you know how it goes...
Thanks,
Paul

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[Asterisk-Users] PSTN line tests

2004-05-07 Thread Clif Jones

Has anyone found any good online resources for performing transmission
tests for POTS lines?  There is plenty of info on this list about 
adjusting gains
on X100 cards, etc. but I am looking for test procedures using test sets.
I'm talking about tests for echo loss, distortion, etc.  Thanks in advance
for you help!

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Re: [Asterisk-Users] Cisco 7940 microphone volume

2004-05-07 Thread John Fraizer

Frederic Steinfels wrote:

When talking to me, people are complaining the volume was not high enough.

The phone only allows to change the volume of the speaker/earpiece. Is 
there an alternative solution? Is it possible to increase the volume in 
asterisk?

Frederic
___
Sounds like either you're not holding the handset in the proper position 
or your handset is faulty.  I've never seen a 7960/7940 that didn't 
provide ample mic gain.

John
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[Asterisk-Users] Re: quadBRI ISDN telephone

2004-05-07 Thread Matthias Cramer

On Fri, 7 May 2004 10:51:56 +0200
Pedro Vela [EMAIL PROTECTED] wrote:

 Hello,
 
 We have a quadBRI in NT mode with bri_cpe_ptmp signalling and when connect a
 ISDN telephone to this nothings happen.

Do you have a power feeding module for the quadBRI, I think you need that, but not 
shure.

Best regards

  Matthias


-- 
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/_.  \   Dolphins Network Systems AGPhone +41-44-847'45'45
   |/ -\ .)  Libernstrasse 24   Fax   +41-44-847'45'49
 -'^`-   \;  CH-8112 Otelfingen http://www.dolphins.ch/
GnuPG 1024D/2D208250 = DBC6 65B6 7083 1029 781E  3959 B62F DF1C 2D20 8250


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Description: PGP signature

RE: [Asterisk-Users] 5 seconds delay with Macros

2004-05-07 Thread Uriel Carrasquilla

I have noticed that when I switched to macros in my extensions.conf, there
is now a 5 second delay.
The macro starts with an announcement and then voicemail.
Has anybody noticed the same?
is it a feature?
URiel


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[Asterisk-Users] SIP: Trouble with Moved temporarily (302)

Hi folks,

this does look like a bug to me: Asterisk replaces the @63.214.186.6 by 
@context which obviously leads to a failure. Any comments, do I have a 
configuration issue on my side that I missed?

Cheers, Philipp

-- Executing Dial(SIP/philipp-bd5f, SIP/[EMAIL PROTECTED]
out|90) in new stack
-- Called [EMAIL PROTECTED]
-- Got SIP response 302 Moved Temporarily new 
sip:[EMAIL PROTECTED] back from 63.214.186.6
-- Now forwarding SIP/philipp-bd5f to '[EMAIL PROTECTED]' (thanks to 
SIP/nikotel-out-c286)
May  7 14:20:54 NOTICE[18450]: chan_local.c:362 local_alloc: No such 
extension/context [EMAIL PROTECTED] creating local channel
May  7 14:20:54 NOTICE[18450]: app_dial.c:204 wait_for_answer: Unable to 
create local channel for call forward to '[EMAIL PROTECTED]'


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[Asterisk-Users] zaptel.conf question

2004-05-07 Thread James Bean


Sorry very very very newbie here,

I just started setting up a asterix box as a test environment for my
work to see if it is a viable solution.

I have a standard TMD400P Development Kit with a FXS and FXO module on
it, and a standard analog handset plugged into the FXS module and a
Analog phone line plugged into a FXO.

My hope is to setup asterix to communicate with an existing OKI VoIP
network. No NAT required, all communication is by dedicated secured VPN.

Sorry for my lack of knowledge in this area but if someone could point
me in the right direction or send me a zaptel.conf and zaptela.conf that
would work in my situation it would be very much appreciated, some of
the basic text files I am finding on the net seem a little
contradictory.

James
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[Asterisk-Users] meetme conf-background.agi

2004-05-07 Thread Atif Rasheed









Hello there!



Somebody tried the meetme|b which initiates the conf-background
AGI

Actually I want to originate another call from a conferencemy
AGI originates the call and connects it to the conference, but the call is nowhere



My extension

exten = 21,1,meetme(21|pb)



and my AGI



#!/usr/bin/perl -w



$aginame=conf-background.agi;

use File::Copy cp;

use Asterisk::AGI;

$AGI = new Asterisk::AGI;

my %input = $AGI-ReadParse();





print STDERR Dialing your number\n;



$srcfile=/tmp/mycall;

$dstfile=/var/spool/asterisk/outgoing/mycall;

open(MYCALL,$srcfile) || die Cant't
open file :$srcfile $!\n;

print MYCALL Channel:Zap/1/13\n;

print MYCALL MaxRetries:2\n;

print MYCALL RetryTime:60\n;

print MYCALL WaitTime:30\n;

print MYCALL Context:default\n;

print MYCALL Extension:22\n;

print MYCALL Priority:1\n;

close MYCALL;

cp($srcfile,$dstfile);



#used to hold the AGI, otherwise it quits

$AGI-get_data('ccs-getnumber','1','2');



print STDERR dialing complete...\n;





Some one can sort out, where things are going wrong

Thank you

Atif


35,1 Top

[Asterisk-Users] WI FI IP phones??

Are there any other wireless IP phones out there other then the Cisco
7920??
-- 
James Moran [EMAIL PROTECTED]
Potential Technologies

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Re: [Asterisk-Users] zaptel.conf question

2004-05-07 Thread Togan Muftuoglu

* James Bean; [EMAIL PROTECTED] on 07 May, 2004 wrote:
Sorry for my lack of knowledge in this area but if someone could point
me in the right direction or send me a zaptel.conf and zaptela.conf that
would work in my situation it would be very much appreciated, some of
the basic text files I am finding on the net seem a little
contradictory.
Look the Development kit configurations here 

http://www.digium.com/index.php?menu=documentation



--

Togan Muftuoglu

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Re: [Asterisk-Users] Trouble compiling latest CVS

2004-05-07 Thread Denis E. Pilon

Upgrade bison...i had the same problems until i upgraded bison.

On Fri, 2004-05-07 at 07:17, Philipp von Klitzing wrote:
 Hm...
 
  since a couple of days I can't seem to be able to compile CVS HEAD on 
  RH7.2. On a RH7.3 machine with bison-1.35-1 it appears to be fine 
  though... any advice?
 
 Actually this doesn't seem to be related to bison - I can't even compile 
 my old CVS-HEAD-05/03/04-19:58:33 anymore, getting the same error. 
 Except for a CURL upgrade there was not major change on the system, at 
 least not that I know of... I did do a clean checkout, but no 
 improvement... am a bit puzzled...
 
 Philipp
 
 
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[Asterisk-Users] SIP Wokflow diagram

2004-05-07 Thread Ignace CARIA

Hi everybody,

I would like to create SIP call flow Diagram under Windows.  Is anybody 
know a program to perform it?  I have already Ethereal and I would like 
an explicit diagram just to show where something have problems...

Thanks

Ignace

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Re: [Asterisk-Users] WI FI IP phones??

2004-05-07 Thread Paul Tyreman

Why not get a analogue to IP adapter, then use a Digital Cordless phone.

Much cheeper than the 7920 and works wonders for me.

I've got a couple of adapters for sale at the moment, e-mail me if your
interested !

Paul.



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James Moran
Posted At: 07 May 2004 13:59
Posted To: Asterisk-Users
Conversation: [Asterisk-Users] WI FI IP phones??
Subject: [Asterisk-Users] WI FI IP phones??


Are there any other wireless IP phones out there other then the Cisco 7920??
--
James Moran [EMAIL PROTECTED]
Potential Technologies

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Re: [Asterisk-Users] SIP: Trouble with Moved temporarily (302)

2004-05-07 Thread Olle E. Johansson

Philipp von Klitzing wrote:

Hi folks,

this does look like a bug to me: Asterisk replaces the @63.214.186.6 by 
@context which obviously leads to a failure. Any comments, do I have a 
configuration issue on my side that I missed?

We don't support 302 redirects to other hosts/domains now.
It's a bug, we have a plan and I've started coding :-)
Waiting for another fix in app_dial before I can solve it...
/O
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Re: [Asterisk-Users] WI FI IP phones??

2004-05-07 Thread David Liu

Hey there,

There is ZyXel Prestige 2000W Wi-Fi IEEE 802.11b

David

- Original Message - 
From: James Moran [EMAIL PROTECTED]
To: Asterisk [EMAIL PROTECTED]
Sent: Friday, May 07, 2004 8:59 PM
Subject: [Asterisk-Users] WI FI IP phones??


 Are there any other wireless IP phones out there other then the Cisco
 7920??
 -- 
 James Moran [EMAIL PROTECTED]
 Potential Technologies
 
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Re: [Asterisk-Users] SIP Wokflow diagram

 I would like to create SIP call flow Diagram under Windows.  Is anybody 
 know a program to perform it?  I have already Ethereal and I would like 
 an explicit diagram just to show where something have problems...

Might take a look at:
http://www.cisco.com/en/US/products/sw/iosswrel/ps1835/products_programming_reference_guide_book0
9186a0080080221.html

There are some good flow diagrams there.

Rich


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Re: [Asterisk-Users] SIP: Trouble with Moved temporarily (302)

2004-05-07 Thread Eric Wieling

On Fri, 2004-05-07 at 07:41, Philipp von Klitzing wrote:
 Hi folks,
 
 this does look like a bug to me: Asterisk replaces the @63.214.186.6 by 
 @context which obviously leads to a failure. Any comments, do I have a 
 configuration issue on my side that I missed?
 
 Cheers, Philipp
 
 -- Executing Dial(SIP/philipp-bd5f, SIP/[EMAIL PROTECTED]
 out|90) in new stack
 -- Called [EMAIL PROTECTED]
 -- Got SIP response 302 Moved Temporarily new 
 sip:[EMAIL PROTECTED] back from 63.214.186.6
 -- Now forwarding SIP/philipp-bd5f to '[EMAIL PROTECTED]' (thanks to 
 SIP/nikotel-out-c286)
 May  7 14:20:54 NOTICE[18450]: chan_local.c:362 local_alloc: No such 
 extension/context [EMAIL PROTECTED] creating local channel
 May  7 14:20:54 NOTICE[18450]: app_dial.c:204 wait_for_answer: Unable to 
 create local channel for call forward to '[EMAIL PROTECTED]'

302 Moved is not fully supported by chan_sip.  Personally I like this
because the way Asterisk currently supports 302 Moved will prevent calls
from being forwarded outside of Asterisk's dialplan.  I would just
create an exten = joesmith,1,GoTo(xxx,n) where xxx is the extension you
want the call to do and n is the priority.  See show application goto.


-- 
  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] WI FI IP phones??

We need to have about 30 phones on one floor

On Fri, 2004-05-07 at 09:18, Paul Tyreman wrote:
 Why not get a analogue to IP adapter, then use a Digital Cordless phone.
 
 Much cheeper than the 7920 and works wonders for me.
 
 I've got a couple of adapters for sale at the moment, e-mail me if your
 interested !
 
 Paul.
 
 
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of James Moran
 Posted At: 07 May 2004 13:59
 Posted To: Asterisk-Users
 Conversation: [Asterisk-Users] WI FI IP phones??
 Subject: [Asterisk-Users] WI FI IP phones??
 
 
 Are there any other wireless IP phones out there other then the Cisco 7920??
 --
 James Moran [EMAIL PROTECTED]
 Potential Technologies
 
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-- 
James Moran [EMAIL PROTECTED]
Potential Technologies

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RE: [Asterisk-Users] Cisco 7940/7960 SIP functionality questions

2004-05-07 Thread Jennings, Mike

Title: RE: [Asterisk-Users] Cisco 7940/7960 SIP functionality questions





Is the 7970 still problematic?


-Original Message-
From: Fran Boon [mailto:[EMAIL PROTECTED]]
Sent: Wednesday, April 21, 2004 3:29 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Cisco 7940/7960 SIP functionality
questions



On Wed, 2004-04-21 at 20:20, David Carter wrote:
 I'm considering using Asterisk with some type of Cisco phone, and currently
 considering either the 7940 or 7960 because of its more-complete functionality
 (compared to the 7905).
 I'm currently wondering:
  Do all the expected functions (transfer, conference, voice mail, message
  waiting indicator, etc.) work normally with Asterisk over SIP?


All work great :)


  What caveats are known about using these phones with SIP, as opposed to
  Cisco's proprietary SCCP? If an SCCP module is available for Asterisk,
  how functional is it?


There are 2 SCCP modules chan_sccp  chan_skinny
I've not personally used either yet, but I believe they offer working
basic functionality, but are not as advanced as SIP/IAX or, indeed, SCCP
with CallManager.


  How customizable are the phone menus while using SIP (or if a SCCP
  module is available, using SCCP)?


Services menu is very customisable:
http://www.voip-info.org/wiki-Asterisk+Cisco+79XX+XML+Services


There is even a manager interface for Asterisk available!:
http://asterisk.edihost.co.uk/am-web/


 Cisco doesn't seem to have much documentation online about using these phones
 in SIP mode, so if anyone is using these phones now, I'd appreciate hearing
 about your experiences.


A good resource is:
http://www.voip-info.org/wiki-Asterisk+phone+cisco+79xx


F


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A.G. Edwards & Sons' outgoing and incoming e-mails are electronically
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than the recipient.

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RE: [Asterisk-Users] Mediatrix 1204 (4x FXO)

2004-05-07 Thread Dawid Mielnik




And 
what problem do you have with registering ?
Jeremy 
Jones has recently posted his SNMP walkthrough from a mediatrix 1104 - you might 
reference that, configuring 1204 should be very similar to that of 
1104.

Regards,
Dave

  -Original Message-From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED]On Behalf Of Wojciech 
  TrycSent: Thursday, May 06, 2004 5:27 AMTo: 
  [EMAIL PROTECTED]Subject: [Asterisk-Users] Mediatrix 
  1204 (4x FXO)
  
  
  I have successfully implemented 1204 in semi 
  production environment. Just want to share that it works very well, through 
  the firewall (NATed). 
  Unfortunately, it can not register with the 
  server (and authenticate) but otherwise everything is fine. The audio quality 
  is very good.
  Regards,
  Wojtek

Re: [Asterisk-Users] WI FI IP phones??

2004-05-07 Thread John Fraizer

James Moran wrote:

We need to have about 30 phones on one floor

And you really think that WiFi phones are suited for this application? 
Not an RF engineer, are ya?

John
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RE: [Asterisk-Users] mpg123 versions ?

My thoughts also. :P

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Rich Adamson
 Sent: Friday, May 07, 2004 7:50 AM
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] mpg123 versions ?

  We find that mpg123 0.59r works best.  mpg123 0.59s-mh4 = the devil.
 
  What versions does everyone use without problems.
 
  0.59r is PERFECT

 Been using mpg123-0.59q-1.i386.rpm since late September with no problems.

 Not had a need to change; ain't broke, don't fix it.



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RE: [Asterisk-Users] Missing digits on TDM400P incomplete dial string - Email found in subject

Run /usr/src/zaptel/ztmonitor 32 -v 
And adjust your gains in /etc/asterisk/zapata.conf accordingly.

Gregory P. Scasny

Golden Technologies Inc.

http://www.golden-tech.com

219-462-7200

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of bam
Sent: Friday, May 07, 2004 3:35 AM
To: [EMAIL PROTECTED]
Subject: [SPAM] - [Asterisk-Users] Missing digits on TDM400P incomplete
dial string - Email found in subject

We are experiencing problems on a FXS interface where the client is
dialing 
numbers, but digits are being dropped somewhere from the dial string. 
Typically one or two digits are not being presented. We've tried
different 
handsets to no avail, and I am assuming that it is some sort of timing
problem.

Are there any parameters I can tweak to try and rectify this?


zapata.conf

context=hardwire
group=3
signalling=fxo_ks
mailbox=8765
callerid=Acme 8765
channel=32




extensions.conf

[hardwire]
;
exten = _NXX,1,SetCallerID(0141411${CALLERIDNUM})
exten = _NXX,2,CallingPres(3)
exten = _NXX,3,Dial(Zap/g1/0141${EXTEN})

exten = _0.,1,SetCallerID(0141411${CALLERIDNUM})
exten = _0.,2,CallingPres(3)
exten = _0.,3,Dial(Zap/g1/${EXTEN})

exten = t,1,Hangup ; If they take too long, give
up.
exten = i,1,Hangup ; If they get it wrong, give up 


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RE: [Asterisk-Users] meetme conf-background.agi









Only works on zap interfaces. What
are you using?



bkw





-Original Message-
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Atif Rasheed
Sent: Friday, May 07, 2004 7:57 PM
To:
[EMAIL PROTECTED]
Subject: [Asterisk-Users] meetme
conf-background.agi



Hello there!



Somebody tried the meetme|b which initiates the
conf-background AGI

Actually I want to originate another call from a
conferencemy AGI originates the call and connects it to the conference,
but the call is nowhere



My extension

exten = 21,1,meetme(21|pb)



and my AGI



#!/usr/bin/perl -w



$aginame=conf-background.agi;

use File::Copy cp;

use Asterisk::AGI;

$AGI = new Asterisk::AGI;

my %input = $AGI-ReadParse();





print STDERR Dialing your number\n;



$srcfile=/tmp/mycall;

$dstfile=/var/spool/asterisk/outgoing/mycall;

open(MYCALL,$srcfile) || die Cant't
open file :$srcfile $!\n;

print MYCALL Channel:Zap/1/13\n;

print MYCALL MaxRetries:2\n;

print MYCALL RetryTime:60\n;

print MYCALL WaitTime:30\n;

print MYCALL Context:default\n;

print MYCALL Extension:22\n;

print MYCALL Priority:1\n;

close MYCALL;

cp($srcfile,$dstfile);



#used to hold the AGI, otherwise it quits

$AGI-get_data('ccs-getnumber','1','2');



print STDERR dialing complete...\n;





Some one can sort out, where things are going wrong

Thank you

Atif


35,1 Top

Re: [Asterisk-Users] WI FI IP phones??

2004-05-07 Thread Michael Koehler

i prefer zyxel p200w

for a picture see 
http://www.voipbox.de/images/private/protzundco/equip.jpg at the upper 
left corner

James Moran wrote:

Are there any other wireless IP phones out there other then the Cisco
7920??
 



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RE: [Asterisk-Users] * ISDN-BRI-PTP DID ISDN4Linux does not show incoming number

2004-05-07 Thread ePyron Felix Deierlein

Hi Andreas,

I guess it is better to buy a B1 or C2 :-). They are not very expensive at
ebay. Or you buy digium hardware, it surely runs with *...
Or have a look at www.junghanns.net (author of chan_capi)

He sells a 4 Port BRI ...

Bye

Felix 

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Andreas Frackowiak
 Sent: Friday, May 07, 2004 11:05 AM
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] *  ISDN-BRI-PTP  DID  
 ISDN4Linux does not show incoming number
 
 Hallo Felix,
 
  it seems that the FAQ only describes windows  co. 
  Just try to use the capi driver, I guess you would get much more 
  support for capi here...
 
 Well now I am sure: The AVM-Fritz-CAPI does not work with PTP.
 
 o I have tried it and it doesn't work
 o I asked AVM and they answered that the Fritz
   CAPI-Software (Windows + Linux) does not support
   DDI/PTP-Mode.
 o I found a lot of messages in old archives of this list
   and the i4l-list which also say that PTP with
   Fritz CAPI does not work.
 
 Also mISDN (ISDN4Linux successor with CAPI20) maybe will 
 support P2P with Fritz Card sometime, but not today.
 
 And so it seems that my problem between ISDN4Linux and the 
 chan_modem_i4l driver remains an unsolved mystery.
 
 So maybe I have to buy an AVM B1 or C2 card to circumvent 
 this problem or use something else than asterisk.
 
 thanks and regards
 Andreas
 
 
   -Original Message-
   From: [EMAIL PROTECTED]
   [mailto:[EMAIL PROTECTED] On Behalf 
 Of Andreas 
   Frackowiak
   Sent: Wednesday, May 05, 2004 8:08 PM
   To: [EMAIL PROTECTED]
   Subject: Re: [Asterisk-Users] *  ISDN-BRI-PTP  DID  ISDN4Linux 
   does not show incoming number
   
   Hi Felix,
   
 I am using Asterisk on a DSS1 ISDN-BRI with ISDN4Linux
   (and a Fritz
 Card PnP).
 The ISDN-BRI is in PTP-Mode (Point to Point german: 
 Anlagenanschluss) which is enabled within I4L with 
 hisaxctrl 
 fcpcipnp0 7 1.
are you shure, that the capi does not support PTP?
I have an AVM C4 card, but it should be the same with 
 the fritz..
   
   Well, I am not sure, but AVM says in:
   http://www.avm.de/de/Service/FAQs/FAQ_Sammlung/2671.php3
   that only the B1-family of cards and the C2 and C4 Controllers 
   support PTP.
   
   I would be very happy if someone has a Fritz with CAPI 
 working with 
   a PTP und could proove that I am wrong.
   
   I also would be very happy if someone could help me with the 
   original question, why I4L does not give the called 
 number / MSN to 
   Asterisk (and help me fix it, of course :)
   
   Thanks
   Andreas
   
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Re: [Asterisk-Users] WI FI IP phones??

2004-05-07 Thread Andrew Kohlsmith

  We need to have about 30 phones on one floor

 And you really think that WiFi phones are suited for this application?
 Not an RF engineer, are ya?

at ~80kbps per phone and (guessing) 5.5mbps average connect I would be curious 
to see how bad 30 simultaneous conversations would be with a CSMA/CA network 
like 802.11b.  :-)  

I think you're right though, I think that that would be a bad thing, overall.  
Where's token ring 802.11 when you need it?  :-)

-A.
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Re: [Asterisk-Users] WI FI IP phones??

2004-05-07 Thread Andrew Kohlsmith

 i prefer zyxel p200w

Looks just like the Pulver OEM'd WiSIP.

Regards,
Andrew
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RE: [Asterisk-Users] Mediatrix 1204 (4x FXO)

The 1204 does not have the software routines implemented for register.
Their approach is the 1104 registers with the 1204.


 And what problem do you have with registering ?
 Jeremy Jones has recently posted his SNMP walkthrough from a mediatrix 1104 - you 
 might 
reference that, configuring 1204 should be very
 similar to that of 1104.
  
 Regards,
 Dave
 
 -Original Message-
 I have successfully implemented 1204 in semi production environment. Just want 
 to share 
that it works very well, through the firewall
 (NATed).
 Unfortunately, it can not register with the server (and authenticate) but 
 otherwise 
everything is fine. The audio quality is very good.
 Regards,
 Wojtek
---End of Original Message-


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[Asterisk-Users] caller id detection

2004-05-07 Thread listas iPfone




Hi!

I know that is a very posted matter but i have a 
question:

Some one can translate that messages for me? what 
is the mean of that messages? can i do something to correct this and get 
the caller id to work?

May 7 11:26:19 ERROR[1288925632]: 
callerid.c:192 callerid_feed: fsk_serie made mylen  0 (-22)May 7 
11:26:19 WARNING[1288925632]: chan_zap.c:4609 ss_thread: CallerID feed failed: 
SuccessMay 7 11:26:19 WARNING[1288925632]: chan_zap.c:4651 ss_thread: 
CallerID returned with error on channel 'Zap/1-1'May 7 11:26:19 
NOTICE[1288925632]: chan_zap.c:3640 zt_read: Fax detected, but no fax 
extension

Thanks fpr any help

Miklos

Re: [Asterisk-Users] SIP: Trouble with Moved temporarily (302)

Hi!

 302 Moved is not fully supported by chan_sip.  Personally I like this
 because the way Asterisk currently supports 302 Moved will prevent
 calls from being forwarded outside of Asterisk's dialplan.  I would
 just create an exten = joesmith,1,GoTo(xxx,n) where xxx is the
 extension you want the call to do and n is the priority.  See show
 application goto. 

Hm... I dunno... 

The way it works now I am not able to call an *unkown* Nikotel SIP user 
(i.e. I am not aware of which username is mapped to the 99xx 
number), and that's not really so nice. Of course the other option is to 
tell Nikotel to re-consider their user setup, but I don't think that'll 
work. ;-

I can see the redirection danger though: If Nikotel decided to send me 
through PSTN instead then I'd suddenly be charged for the call even 
though I still think this is a free VoIP call. So my option here would be 
to operate two Nikotel accounts, of which only one has a pre-paid budget, 
and the other one without budget is used for redirected/VoIP calls. This 
need results from the fact that only authenticated Nikotel users can call 
Nikotel users.

Which, by the way, makes me long again for some smarter extension 
matching rules than just [1-9], X and N: I'd love to have [alpha] or 
[alphanumeric] or [k-m], and possibly also [length:7-9] for variable 
numbering plans.

Cheers, Philipp

P.S.: For further discussion see bug #730


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Re: [Asterisk-Users] Trouble compiling latest CVS

Hi!

 Upgrade bison...i had the same problems until i upgraded bison.

Which means upgrading glibc ... :-(( 

In other words: Asterisk won't work with RH 7.2 (and the like) anymore, 
basically. Still I wonder why I was once able to compile the March 5 CVS, 
but can't do so anymore. Might be because I have today's zaptel, libpri 
and addons...?

Philipp

   since a couple of days I can't seem to be able to compile CVS HEAD on 
   RH7.2. On a RH7.3 machine with bison-1.35-1 it appears to be fine 
   though... any advice?
  
  Actually this doesn't seem to be related to bison - I can't even compile 
  my old CVS-HEAD-05/03/04-19:58:33 anymore, getting the same error. 
  Except for a CURL upgrade there was not major change on the system, at 
  least not that I know of... I did do a clean checkout, but no 
  improvement... am a bit puzzled...


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Re: [Asterisk-Users] Asterisk and Cisco 7960 problems persist (for me, anyway)

2004-05-07 Thread Iain Stevenson

I've had this too, reported it as a bug last week and got my butt kicked 
for not being responsive enough in providing support to sort it out.  You 
could file another bug report but be sure to have a thick book ready to 
stuff down your trousers.

 Iain

--On Friday, May 7, 2004 10:43 am -0400 Brian Cuthie [EMAIL PROTECTED] 
wrote:

It seems that each time I get a new checkout of * from CVS my Cisco 7960
works worse than before. I know this stuff's in flux, so I mention this
in case it's news.  Anyone else having trouble?  What I'm seeing (er,
hearing) is really choppy audio. The previous version I had installed had
fairly frequent audio dropouts (not present when I make the same calls
through the same * box using a TDM400P interface).
Cheers,

Brian
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Re: [Asterisk-Users] SIP: Trouble with Moved temporarily (302)

2004-05-07 Thread Eric Wieling

On Fri, 2004-05-07 at 09:58, Philipp von Klitzing wrote:
 The way it works now I am not able to call an *unkown* Nikotel SIP user 
 (i.e. I am not aware of which username is mapped to the 99xx 
 number), and that's not really so nice. Of course the other option is to 
 tell Nikotel to re-consider their user setup, but I don't think that'll 
 work. ;-

I feel that Nikotel should be using reinvites, not 302 Moved.

What about corporate users that NEVER want their users to go outside of
the corporate dial plan?

-- 
  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] WI FI IP phones??

2004-05-07 Thread Jonathan Moore

You are assuming that they all have to go to the same Access Point and be on the
same channel. For a high density setup you can get APs that allow you to turn
down the signal strength so they can be more densly placed. With the Wisip or
the Zytel you really need to go with g729 anyway for them to work properly at
which point I think the footprint is only about 30kbps. I am not saying this
will work, but I think it could be made to work with the correctly designed Wifi
network. Point of fact we really don't know much about his setup since he just
says on one floor. The real question here is what is the necessary density per AP?


-- 
Jonathan Moore
Director of Technology
Winfield Public Schools
Office 620.221.5100
Fax 620.221.0508


Quoting Andrew Kohlsmith [EMAIL PROTECTED]:

   We need to have about 30 phones on one floor
 
  And you really think that WiFi phones are suited for this application?
  Not an RF engineer, are ya?
 
 at ~80kbps per phone and (guessing) 5.5mbps average connect I would be
 curious 
 to see how bad 30 simultaneous conversations would be with a CSMA/CA network
 
 like 802.11b.  :-)  
 
 I think you're right though, I think that that would be a bad thing, overall.
  
 Where's token ring 802.11 when you need it?  :-)
 
 -A.
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Re: [Asterisk-Users] Asterisk and Cisco 7960 problems persist (for me, anyway)

2004-05-07 Thread Brian Cuthie

It's not the switch. It's lightly loaded 100Mb.

-brian

Bisker, Scott (7805) wrote:

What kind of switch do you have your phones plugged into?  If your switch is highly loaded, or you are doing lots of multicast or broadcast, your SIP streams are going to suffer unless you are filtering that traffic at the port level or have separate VOIP VLANS.



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Brian Cuthie
Sent: Friday, May 07, 2004 10:43 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Asterisk and Cisco 7960 problems persist (for
me, anyway)


It seems that each time I get a new checkout of * from CVS my Cisco 7960 
works worse than before. I know this stuff's in flux, so I mention this 
in case it's news.  Anyone else having trouble?  What I'm seeing (er, 
hearing) is really choppy audio. The previous version I had installed 
had fairly frequent audio dropouts (not present when I make the same 
calls through the same * box using a TDM400P interface).

Cheers,

Brian
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Re: [Asterisk-Users] Asterisk and Cisco 7960 problems persist (for me, anyway)

2004-05-07 Thread Tom

At 09:43 AM 5/7/2004, you wrote:

It seems that each time I get a new checkout of * from CVS my Cisco 7960 
works worse than before. I know this stuff's in flux, so I mention this in 
case it's news.  Anyone else having trouble?  What I'm seeing (er, 
hearing) is really choppy audio. The previous version I had installed had 
fairly frequent audio dropouts (not present when I make the same calls 
through the same * box using a TDM400P interface).
No dropout problems or choppy audio running Asterisk CVS-04/19/04-14:31:03 
with 4 Cisco 7940/60 SIP 6.3 phones on a 2.4GHz P4 Supermicro 
server.  Analog phones through our TDM400P do sound much better but the 
audio problems on our Cisco SIP phones are echo problems.  People are 
working on solutions.

Tom

Cheers,

Brian
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Re: [Asterisk-Users] HOW TO PROGRAM NEW MODULES

Hi!

  | exten = 999,1,SetGroup(moh) 
  | exten = 999,2,CheckGroup(1) 
  | exten = 999,3,Answer 
  | exten = 999,4,MusicOnHold(default)
  |
  | See?
  | You can limit that to just 1 user at a time or what ever you wish :
  |
  | bkw

Great! So this is a means that can be used as an outgoing limit feature 
to restrict the number of active IAX2 calls (or SIP calls etc) through a 
thin DSL uplink?

Cheers, Philipp


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Re: [Asterisk-Users] Trunk with CIRPAK

2004-05-07 Thread Arnaud Pignard

Hello,

I have fix the problem, i haven't notice that's in general i have
videosupport=yes
with this in sip.conf, it's doesn't disable videosupport :

[provider]
host=x.x.x.x
type=peer
videosupport=no
silenceSuppression=no
Now working with videosupport=no in general

At 17:08 07/05/2004, you wrote:
Hello,

I have trouble to enable a sip trunk with a CIRPAK.
CIRPAK support answer that's there parameter are unvalid :
a=silenceSupp:off - - - -
is not standard and not working with cirpak - to be remove
m=video 13072 RTP/AVP
no video, how to remove it ?
my extension.conf :
exten = _6X.,1,Dial,SIP/${EXTEN:[EMAIL PROTECTED]
Regards,

--
Arnaud Pignard ([EMAIL PROTECTED])
Frontier Online - Opérateur Internet
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Re: [Asterisk-Users] Asterisk and Cisco 7960 problems persist (for me, anyway)

 It seems that each time I get a new checkout of * from CVS my Cisco 7960 
 works worse than before. I know this stuff's in flux, so I mention this 
 in case it's news.  Anyone else having trouble?  What I'm seeing (er, 
 hearing) is really choppy audio. The previous version I had installed 
 had fairly frequent audio dropouts (not present when I make the same 
 calls through the same * box using a TDM400P interface).

Brian,

Are you having the choppy audio only on iax2 links or on other calls as
well?

There was an issue with erratic iax2 timestamps which caused the Cisco
phones to effectively drop any sip packet that had uneven timestamps
causing extremely choppy audio. The choppy audio (as I seen it) was
only in one direction (from the iax2 source with the erratic timestamps
towards to 7960 phone).

If your issue is not associated with iax2, then be aware the Cisco v6.x
code changed DSP firmware internally, and any sip/rtp packets arriving with
uneven timestamps (within the rtp pkts) will be dropped and cause the
choppy audio. You should be able to see the timestamps with ethereal.
The timestamp difference between successive pkts should be exactly
160 milliseconds; if its anything else, the phone will drop the pkt.
(Not sure if that is a real Cisco bug or a planned change, but it
certainly has a hugh negative impact on voice quality.)

I'm running CVS-HEAD-05/02/04 with no problems today.

Rich


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RE: [Asterisk-Users] Asterisk and Cisco 7960 problems persist (for me, anyway)

2004-05-07 Thread Bisker, Scott (7805)

I've got Asterisk STABLE-CVS-4/19/04 with 12 Cisco 7960 phones 6.0 Firmware using 
ulaw, 6 Polycom IP500 ulaw phones, and 192 Zap channels.  I have Gig-E Copper to my 
server and 100Mbit-Full to all my phones.  I haven't had any choppy audio at all.  My 
switch is a Cisco 4500.  

-sb



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Tom
Sent: Friday, May 07, 2004 11:30 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk and Cisco 7960 problems persist
(for me, anyway)


At 09:43 AM 5/7/2004, you wrote:

It seems that each time I get a new checkout of * from CVS my Cisco 7960 
works worse than before. I know this stuff's in flux, so I mention this in 
case it's news.  Anyone else having trouble?  What I'm seeing (er, 
hearing) is really choppy audio. The previous version I had installed had 
fairly frequent audio dropouts (not present when I make the same calls 
through the same * box using a TDM400P interface).

No dropout problems or choppy audio running Asterisk CVS-04/19/04-14:31:03 
with 4 Cisco 7940/60 SIP 6.3 phones on a 2.4GHz P4 Supermicro 
server.  Analog phones through our TDM400P do sound much better but the 
audio problems on our Cisco SIP phones are echo problems.  People are 
working on solutions.

Tom

Cheers,

Brian
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Re: [Asterisk-Users] Asterisk and Cisco 7960 problems persist (for me, anyway)

2004-05-07 Thread Brian Cuthie

Ah, this reminds me that I forgot to mention that our network looks like 
this:

   Cisco --- SIP   Asterisk  IAX   Aterisk  IAX 
 Asterisk  PRI  PSTN

-brian

Tom wrote:

At 09:43 AM 5/7/2004, you wrote:

It seems that each time I get a new checkout of * from CVS my Cisco 
7960 works worse than before. I know this stuff's in flux, so I 
mention this in case it's news.  Anyone else having trouble?  What 
I'm seeing (er, hearing) is really choppy audio. The previous version 
I had installed had fairly frequent audio dropouts (not present when 
I make the same calls through the same * box using a TDM400P interface).


No dropout problems or choppy audio running Asterisk 
CVS-04/19/04-14:31:03 with 4 Cisco 7940/60 SIP 6.3 phones on a 2.4GHz 
P4 Supermicro server.  Analog phones through our TDM400P do sound much 
better but the audio problems on our Cisco SIP phones are echo 
problems.  People are working on solutions.

Tom

Cheers,

Brian
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RE: [Asterisk-Users] Asterisk and Cisco 7960 problems persist (for me, anyway)

Just an FYI if you can run tethereal -n udp port 4569

And watch the timestamps(should be even 20ms increments per call leg).  Both
ends will need to be updated also.  If not you will get some very strange
timestamp issues and jitter and timestamps might not be right.  If you
have one end on cvs-stable and one on cvs-head you might see this problem
also.  I don't see any issues with IAX2 from my 7960 out nufone.

bkw

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Rich Adamson
 Sent: Friday, May 07, 2004 11:36 AM
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] Asterisk and Cisco 7960 problems persist
 (for me, anyway)

  It seems that each time I get a new checkout of * from CVS my Cisco 7960
  works worse than before. I know this stuff's in flux, so I mention this
  in case it's news.  Anyone else having trouble?  What I'm seeing (er,
  hearing) is really choppy audio. The previous version I had installed
  had fairly frequent audio dropouts (not present when I make the same
  calls through the same * box using a TDM400P interface).

 Brian,

 Are you having the choppy audio only on iax2 links or on other calls as
 well?

 There was an issue with erratic iax2 timestamps which caused the Cisco
 phones to effectively drop any sip packet that had uneven timestamps
 causing extremely choppy audio. The choppy audio (as I seen it) was
 only in one direction (from the iax2 source with the erratic timestamps
 towards to 7960 phone).

 If your issue is not associated with iax2, then be aware the Cisco v6.x
 code changed DSP firmware internally, and any sip/rtp packets arriving
 with
 uneven timestamps (within the rtp pkts) will be dropped and cause the
 choppy audio. You should be able to see the timestamps with ethereal.
 The timestamp difference between successive pkts should be exactly
 160 milliseconds; if its anything else, the phone will drop the pkt.
 (Not sure if that is a real Cisco bug or a planned change, but it
 certainly has a hugh negative impact on voice quality.)

 I'm running CVS-HEAD-05/02/04 with no problems today.

 Rich


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RE: [Asterisk-Users] HOW TO PROGRAM NEW MODULES

Or any channel for that matter.

Bkw :)

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Philipp von Klitzing
 Sent: Friday, May 07, 2004 10:35 AM
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] HOW TO PROGRAM NEW MODULES

 Hi!

   | exten = 999,1,SetGroup(moh)
   | exten = 999,2,CheckGroup(1)
   | exten = 999,3,Answer
   | exten = 999,4,MusicOnHold(default)
   |
   | See?
   | You can limit that to just 1 user at a time or what ever you wish :
   |
   | bkw

 Great! So this is a means that can be used as an outgoing limit feature
 to restrict the number of active IAX2 calls (or SIP calls etc) through a
 thin DSL uplink?

 Cheers, Philipp


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[Asterisk-Users] Cisco 7940 Phones as paging system?

2004-05-07 Thread Chris Travers

Hi all;

I have been searching for an answer to a question that a customer asked 
me and I have only found a few older answers.  So, wanting to find out 
if anyone has any experience with this issue and can help provide me 
with some advice.

I have a customer which is strongly interested in using Asterisk as a 
PBX.  One of the core requirements, however, is that the system MUST be 
able to support intercom/paging.  Having searched the archives, it 
appears that this question was asked about 6 months ago, and the answer 
was that the Cisco phones support this using SCCP and having one line 
set to auto-answer, but at the time this was not supported in the SIP 
image.  Is this still the case?

Although I know that SIP is the preferred protocol for connecting these 
phones with Asterisk, how stable/reliable are the skinny channels?  Is 
there any reason I should be rethinking this solution?

Best Wishes,
Chris Travers
Metatron Technology Consulting
begin:vcard
fn:Chris Travers
n:Travers;Chris
email;internet:[EMAIL PROTECTED]
x-mozilla-html:FALSE
version:2.1
end:vcard

Re: [Asterisk-Users] asterisk-oh323, new version 0.6.1

2004-05-07 Thread Jeremy McNamara

Paul Berger wrote:

Le jeu 06/05/2004 à 18:52, Michael Manousos a écrit :
 

This new version (0.6.1) of asterisk-oh323 fixes the one-way audio
problem of the previous release.
   

Hi, what is the difference between chan_h323 and asterisk-oh323? Are
they mutually exclusive? Is one better than the other?
chan_h323 came directly with my .deb package, and I am currently
compiling the CVS version of *, to test ast-oh323, so I may get some
answers then :-)
Thanks,
Paul
 

fact  I created chan_h323 because the author of asterisk-oh323 
wouldn't listen to the rest of the community on proper implementation of 
an Asterisk channel driver.  After many complaints from others about 
asterisk-oh323 on the mailing list and the IRC channel, I took it upon 
myself and created chan_h323 and disclaimed my code to Digium so that 
Asterisk could have H.323 support 'out of the box' (for better or worse) 
/fact

Any debate on which one is better is beyond me, I am certainly biased.

Jeremy McNamara





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Re: [Asterisk-Users] One voicemail - multiple boxes?

Hi!

 I don't want to re-invent the wheel if someone has already hacked a way 
 to do this.
 
 One of my customers has a number of stores, and he wants to leave one 
 voicemail that would be delivered to all the managers at once.  Each has 
 a voicemail account on his server.
 
 I have googled around and looked on the WIKI.  Maybe I'm missing it?

Check bugs.digium.com - there is right now smth in the works that 
addresses exactly this. Look for the term broadcast in connection with 
voicemail (or vm).

Cheers, Philipp


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Re: [Asterisk-Users] Asterisk and Cisco 7960 problems persist (for me, anyway)

Upgrade each asterisk (iax2) and the problem will go away. As bkw
mentioned, the problem sources from the location with the older
iax2 code (which probably includes the Stable cvs I believe).

NuFone had the problem in mid/late April as well, but they apparently
updated their code when the issue was discovered/corrected. Other
iax2 providers are likely to source the problem as well. Will take
awhile for everyone to get the code into their production machines.


 Ah, this reminds me that I forgot to mention that our network looks like 
 this:
 
 Cisco --- SIP   Asterisk  IAX   Aterisk  IAX 
  Asterisk  PRI  PSTN
 
 -brian
 
 
 Tom wrote:
 
  At 09:43 AM 5/7/2004, you wrote:
 
  It seems that each time I get a new checkout of * from CVS my Cisco 
  7960 works worse than before. I know this stuff's in flux, so I 
  mention this in case it's news.  Anyone else having trouble?  What 
  I'm seeing (er, hearing) is really choppy audio. The previous version 
  I had installed had fairly frequent audio dropouts (not present when 
  I make the same calls through the same * box using a TDM400P interface).
 
 
  No dropout problems or choppy audio running Asterisk 
  CVS-04/19/04-14:31:03 with 4 Cisco 7940/60 SIP 6.3 phones on a 2.4GHz 
  P4 Supermicro server.  Analog phones through our TDM400P do sound much 
  better but the audio problems on our Cisco SIP phones are echo 
  problems.  People are working on solutions.
 
  Tom
 
  Cheers,
 
  Brian
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Re: [Asterisk-Users] No Audio from Hard Phone to SIP

2004-05-07 Thread Jason A. Pattie

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Eric Wieling wrote:
| Allow ULAW or ALAW, not both, at least for trying to solve a problem.
What is the difference between these codecs?  Which is better?

- --
Jason A. Pattie
[EMAIL PROTECTED]
Xperience, Inc. (http://www.xperienceinc.com)
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RE: [Asterisk-Users] Missing digits on TDM400P incomplete dial string - Email found in subject

2004-05-07 Thread bam

I've  had a quick fiddle to little avail, the readings looked prey good to 
be honest before I started fiddling. Looking a little closer it appears 
that it is the digit 1 that is being lost more that any other.



At 15:25 07/05/04, you wrote:
Run /usr/src/zaptel/ztmonitor 32 -v
And adjust your gains in /etc/asterisk/zapata.conf accordingly.
Gregory P. Scasny

Golden Technologies Inc.

http://www.golden-tech.com

219-462-7200

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of bam
Sent: Friday, May 07, 2004 3:35 AM
To: [EMAIL PROTECTED]
Subject: [SPAM] - [Asterisk-Users] Missing digits on TDM400P incomplete
dial string - Email found in subject
We are experiencing problems on a FXS interface where the client is
dialing
numbers, but digits are being dropped somewhere from the dial string.
Typically one or two digits are not being presented. We've tried
different
handsets to no avail, and I am assuming that it is some sort of timing
problem.
Are there any parameters I can tweak to try and rectify this?

zapata.conf

context=hardwire
group=3
signalling=fxo_ks
mailbox=8765
callerid=Acme 8765
channel=32


extensions.conf

[hardwire]
;
exten = _NXX,1,SetCallerID(0141411${CALLERIDNUM})
exten = _NXX,2,CallingPres(3)
exten = _NXX,3,Dial(Zap/g1/0141${EXTEN})
exten = _0.,1,SetCallerID(0141411${CALLERIDNUM})
exten = _0.,2,CallingPres(3)
exten = _0.,3,Dial(Zap/g1/${EXTEN})
exten = t,1,Hangup ; If they take too long, give
up.
exten = i,1,Hangup ; If they get it wrong, give up
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RE: [Asterisk-Users] Cisco 7940 Phones as paging system?

2004-05-07 Thread Carlton J. O'Riley

The SIP 6.1 image has auto answer available, which would function the same
as the SCCP implementation. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Travers
Sent: Friday, May 07, 2004 12:02 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Cisco 7940 Phones as paging system?

Hi all;

I have been searching for an answer to a question that a customer asked me
and I have only found a few older answers.  So, wanting to find out if
anyone has any experience with this issue and can help provide me with some
advice.

I have a customer which is strongly interested in using Asterisk as a PBX.
One of the core requirements, however, is that the system MUST be able to
support intercom/paging.  Having searched the archives, it appears that this
question was asked about 6 months ago, and the answer was that the Cisco
phones support this using SCCP and having one line set to auto-answer, but
at the time this was not supported in the SIP image.  Is this still the
case?

Although I know that SIP is the preferred protocol for connecting these
phones with Asterisk, how stable/reliable are the skinny channels?  Is there
any reason I should be rethinking this solution?

Best Wishes,
Chris Travers
Metatron Technology Consulting
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Re: [Asterisk-Users] Cisco 7940 Phones as paging system?

 I have been searching for an answer to a question that a customer asked 
 me and I have only found a few older answers.  So, wanting to find out 
 if anyone has any experience with this issue and can help provide me 
 with some advice.
 
 I have a customer which is strongly interested in using Asterisk as a 
 PBX.  One of the core requirements, however, is that the system MUST be 
 able to support intercom/paging.  Having searched the archives, it 
 appears that this question was asked about 6 months ago, and the answer 
 was that the Cisco phones support this using SCCP and having one line 
 set to auto-answer, but at the time this was not supported in the SIP 
 image.  Is this still the case?
 
 Although I know that SIP is the preferred protocol for connecting these 
 phones with Asterisk, how stable/reliable are the skinny channels?  Is 
 there any reason I should be rethinking this solution?

Apparently your search didn't find several other postings on the subject.

The cisco v6.x sip releases also include the ability to auto-answer a
call (required for phone paging), however some folks tend to suggest that
is a security problem as anyone can call that autoanswer extn number
and listen in on whatever is going on around the phone. There is no
beep or other indication the phone/microphone is open.

If your customer is looking for overhead (as in PA) paging, there are lots
of postings relative to add-on hardware, etc, to do that. Use of the 
sound card within the * box has historically been hit/miss as not all
sound cards are supported, etc.

Rich


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Re: [Asterisk-Users] asterisk-oh323, new version 0.6.1

Le ven 07/05/2004 à 18:03, Jeremy McNamara a écrit :
 fact  I created chan_h323 because the author of asterisk-oh323 
 wouldn't listen to the rest of the community on proper implementation of 
 an Asterisk channel driver.  After many complaints from others about 
 asterisk-oh323 on the mailing list and the IRC channel, I took it upon 
 myself and created chan_h323 and disclaimed my code to Digium so that 
 Asterisk could have H.323 support 'out of the box' (for better or worse) 
 /fact

Thanks for the update Jeremy!

 Any debate on which one is better is beyond me, I am certainly biased.

I understand. My question was more regarding features, from an H323
point of view (what does asterisk-oh323 that chan_h323 doesn't, and
vice-versa?). Despite being biased, I think you - more than others - 
might have relevant info about this...

Again, thanks for your time,
Paul

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Re: [Asterisk-Users] WI FI IP phones??

No I'm not but it's a hospital that nurses are on call and need to have
a way to contact them.  On Fri, 2004-05-07 at 09:52, John Fraizer wrote:
 James Moran wrote:
 
  We need to have about 30 phones on one floor
  
 
 And you really think that WiFi phones are suited for this application? 
 Not an RF engineer, are ya?
 
 John
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James Moran [EMAIL PROTECTED]
Potential Technologies

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Re: [Asterisk-Users] No Audio from Hard Phone to SIP

2004-05-07 Thread Kyle Hagan

 The manufacturer think its a bad phone. So Im getting another one today.
They said thay have gotten the Zultys phones to work with asterisk with no
problems. Will let everyone know.

 Zultys sait ULAW was the most common but did not state why most use it.

Kyle


- Original Message - 
From: Jason A. Pattie [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, May 07, 2004 9:17 AM
Subject: Re: [Asterisk-Users] No Audio from Hard Phone to SIP


 -BEGIN PGP SIGNED MESSAGE-
 Hash: SHA1

 Eric Wieling wrote:
 | Allow ULAW or ALAW, not both, at least for trying to solve a problem.

 What is the difference between these codecs?  Which is better?

 - --
 Jason A. Pattie
 [EMAIL PROTECTED]
 Xperience, Inc. (http://www.xperienceinc.com)
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Re: [Asterisk-Users] Newbie x100p install question

 PROBLEM: My x100p only does bad things. When I plug the line from the 
 wall into the card, other phones in my house go nuts, no dialtone, crazy 
 clicking, random tones, etc. Following the steps below, the line is 
 screwed up any time I test after step 2. Just plugging the line from the 
 wall into the x100p takes my other phones off line. External calls to my 
 line hear busy. Obviously, can't do the first testing with asterisk+x100p.
 
 1. Installed asterisk
 2. Installed card in slot (no shared IRQs)
 3. Installed (modprobed) drivers
 4. Modified config files to just basics for my environment
 5. Ran ztcfg
 6. Started asterisk
 
 No errors, asterisk starts fine, modprobe -q shows drivers. Next I'm 
 going to experiment with context since I've seen several 
 recommendations including incoming and bell.
 
 This is kicking my butt, can someone help?

If you're using the x100p's as an interface to a pstn line, then:
1. ensure you are plugging the line into the line jack and not the
   phone jack on the x100p card.
2. check your /etc/zaptel.conf file to ensure it includes:
   # The following addresses x100p card #1 and card #2  
   fxsks=1-2   
   loadzone=us 
3. check your /etc/models.conf file to ensure it includes:
   post-install wcfxo /sbin/ztcfg 
4. If you're not familiar linux, reboot your system and watch for any
   errors loading modules during startup.



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Re: [Asterisk-Users] No Audio from Hard Phone to SIP

2004-05-07 Thread Eric Wieling

On Fri, 2004-05-07 at 11:17, Jason A. Pattie wrote:
 -BEGIN PGP SIGNED MESSAGE-
 Hash: SHA1
 
 Eric Wieling wrote:
 | Allow ULAW or ALAW, not both, at least for trying to solve a problem.
 
 What is the difference between these codecs?  Which is better?

Neither is better.  ulaw is used in T-1 land (mostly USA and Canada),
alaw is used in E-1 land (rest of the world).  It doesn't REALLY matter
which of the two you use, but there might be a tiny amount of extra
overhead if you have to convert from one to the other when hitting the
PSTN using a digital interface like PRI.

-- 
  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] Newbie x100p install question

2004-05-07 Thread Tim Sailer

On Fri, May 07, 2004 at 10:45:13AM -0400, Rick Beasley wrote:
 PROBLEM: My x100p only does bad things. When I plug the line from the 
 wall into the card, other phones in my house go nuts, no dialtone, crazy 
 clicking, random tones, etc. Following the steps below, the line is 
 screwed up any time I test after step 2. Just plugging the line from the 
 wall into the x100p takes my other phones off line. External calls to my 
 line hear busy. Obviously, can't do the first testing with asterisk+x100p.
 
 1. Installed asterisk
 2. Installed card in slot (no shared IRQs)
 3. Installed (modprobed) drivers
 4. Modified config files to just basics for my environment
 5. Ran ztcfg
 6. Started asterisk
 
 No errors, asterisk starts fine, modprobe -q shows drivers. Next I'm 
 going to experiment with context since I've seen several 
 recommendations including incoming and bell.

This sounds like a hardware error, not software. Change the line cord
from the FXO port to the wall. If that doesn't make a difference, change
jacks. Try that jack with another phone.

Tim

-- 

 Tim Sailer Coastal Internet, Inc.  
 Network and Systems Operations PO Box 726  
 http://www.buoy.comMoriches, NY 11955  
 [EMAIL PROTECTED]   (631) 399-2910 IAX 17003992910  

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Re: [Asterisk-Users] WI FI IP phones??

2004-05-07 Thread John Fraizer

James Moran wrote:

No I'm not but it's a hospital that nurses are on call and need to have
a way to contact them.  On Fri, 2004-05-07 at 09:52, John Fraizer wrote:
James Moran wrote:


We need to have about 30 phones on one floor

And you really think that WiFi phones are suited for this application? 
Not an RF engineer, are ya?

John
Um, I'm not so sure that you're going to be able to run WiFi at a 
hospital.  The life safety/support equipment is most likely not 
certified to be resistant to 2.4Ghz interference.  It's been a while 
since I looked up ISM allocations but, I can tell you that I've seen 
many good ideas shot down because of the potential to interfere with 
the medical equipment.

John
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Re: [Asterisk-Users] WI FI IP phones??

Hmm I'll look into it. Thanks.

On Fri, 2004-05-07 at 12:54, John Fraizer wrote:
 James Moran wrote:
 
  No I'm not but it's a hospital that nurses are on call and need to have
  a way to contact them.  On Fri, 2004-05-07 at 09:52, John Fraizer wrote:
  
 James Moran wrote:
 
 
 We need to have about 30 phones on one floor
 
 
 And you really think that WiFi phones are suited for this application? 
 Not an RF engineer, are ya?
 
 John
 
 Um, I'm not so sure that you're going to be able to run WiFi at a 
 hospital.  The life safety/support equipment is most likely not 
 certified to be resistant to 2.4Ghz interference.  It's been a while 
 since I looked up ISM allocations but, I can tell you that I've seen 
 many good ideas shot down because of the potential to interfere with 
 the medical equipment.
 
 John
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James Moran [EMAIL PROTECTED]
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Re: [Asterisk-Users] Trouble compiling latest CVS

Hi!

  Upgrade bison...i had the same problems until i upgraded bison.
 
 Which means upgrading glibc ... :-(( 

Ok ok, I got it - compiled bison from source and disregarded those good-
looking tail-shaking RPMs. ;- Works fine now.

Cheers, Philipp


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Re: [Asterisk-Users] Cisco 7940 Phones as paging system?

Hi!

 able to support intercom/paging.  Having searched the archives, it 
 appears that this question was asked about 6 months ago, and the answer 
 was that the Cisco phones support this using SCCP and having one line 
 set to auto-answer, but at the time this was not supported in the SIP 
 image.  Is this still the case?

Dunno about Cisco, but wanted to let you know that the recent Grandstream 
firmware (.55 and later) now also has an auto-answer option. Still I 
guess I should mention that the microphone of the GS phones in 
speakerphone mode is far from a brilliant implementation (- echo for the 
remote speaker talker, and too thin sound from the person in the room).

Cheers, Philipp


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[Asterisk-Users] PRI, multi D channels and conventional PBXs

2004-05-07 Thread Lee Redmayne

Hi all

OK this may sound like a good one but maybe someone can tell me.

Simple context is - I want to unplug my existing conventional PBX from the
Telco and place * with it's TE410P in between.

Now the difficult part, the existing connection is E1 PRI (Q.931) with 6
B-channels.  I need to be able to trigger a D-channel to the old PBX and a
D-Channel to the Telco (Not BT!).

Next I can put the PBX onto a span 2, it triggers the D-channel and all
seems hunky dory - until you try to acquire a line from * - this gives me:

 -- Extension '' in context 'blah' from '' does not exist.  Rejecting call
on channel 6, span 2

Any suggestions most welcome!
Lee

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RE: [Asterisk-Users] WI FI IP phones??

2004-05-07 Thread Mark Musone

Why not vocera?

http://www.vocera.com

they seem to have the exact product you are looking for and seem to
primarily server hospitals..

-Mark


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James Moran
Sent: Friday, May 07, 2004 1:06 PM
To: Asterisk
Subject: Re: [Asterisk-Users] WI FI IP phones??

Hmm I'll look into it. Thanks.

On Fri, 2004-05-07 at 12:54, John Fraizer wrote:
 James Moran wrote:
 
  No I'm not but it's a hospital that nurses are on call and need to
have
  a way to contact them.  On Fri, 2004-05-07 at 09:52, John Fraizer
wrote:
  
 James Moran wrote:
 
 
 We need to have about 30 phones on one floor
 
 
 And you really think that WiFi phones are suited for this
application? 
 Not an RF engineer, are ya?
 
 John
 
 Um, I'm not so sure that you're going to be able to run WiFi at a 
 hospital.  The life safety/support equipment is most likely not 
 certified to be resistant to 2.4Ghz interference.  It's been a while 
 since I looked up ISM allocations but, I can tell you that I've seen 
 many good ideas shot down because of the potential to interfere
with 
 the medical equipment.
 
 John
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James Moran [EMAIL PROTECTED]
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RE: [Asterisk-Users] Missing digits on TDM400P incomplete dial string - DTMF problem?

2004-05-07 Thread bam

I turned down the rxgain and txgain to -22.0 and -16.0 respectively and 
things  started to look a whole lot more acceptable. Then the client sticks 
on his BT DECT phone and I start losing all the 1s from the dial string.

Does anyone know if BT DECT phones have dodgy DTMF tones?

At 17:19 07/05/04, you wrote:
I've  had a quick fiddle to little avail, the readings looked prey good to 
be honest before I started fiddling. Looking a little closer it appears 
that it is the digit 1 that is being lost more that any other.



At 15:25 07/05/04, you wrote:
Run /usr/src/zaptel/ztmonitor 32 -v
And adjust your gains in /etc/asterisk/zapata.conf accordingly.
Gregory P. Scasny

Golden Technologies Inc.

http://www.golden-tech.com

219-462-7200

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of bam
Sent: Friday, May 07, 2004 3:35 AM
To: [EMAIL PROTECTED]
Subject: [SPAM] - [Asterisk-Users] Missing digits on TDM400P incomplete
dial string - Email found in subject
We are experiencing problems on a FXS interface where the client is
dialing
numbers, but digits are being dropped somewhere from the dial string.
Typically one or two digits are not being presented. We've tried
different
handsets to no avail, and I am assuming that it is some sort of timing
problem.
Are there any parameters I can tweak to try and rectify this?

zapata.conf

context=hardwire
group=3
signalling=fxo_ks
mailbox=8765
callerid=Acme 8765
channel=32


extensions.conf

[hardwire]
;
exten = _NXX,1,SetCallerID(0141411${CALLERIDNUM})
exten = _NXX,2,CallingPres(3)
exten = _NXX,3,Dial(Zap/g1/0141${EXTEN})
exten = _0.,1,SetCallerID(0141411${CALLERIDNUM})
exten = _0.,2,CallingPres(3)
exten = _0.,3,Dial(Zap/g1/${EXTEN})
exten = t,1,Hangup ; If they take too long, give
up.
exten = i,1,Hangup ; If they get it wrong, give up
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Re: [Asterisk-Users] RFD: With echo and other distortion, can ulaw/alaw line quality ever be good enough for faxing?

2004-05-07 Thread Michael Graff

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

On Wednesday 05 May 2004 12:35 pm, Darren Nickerson wrote:
 Folks,

 The silence was deafening ... I had a few private replies but overall I'd
 have to conclude that most people on this list aren't interested in faxing
 thru Asterisk. You're all probably jazzed about VoIP and fax is forgotten
 for now ;-)

FWIW, we use faxes through ATA-186 devices, and they Just Worked.  Literally 
plug them in, they work first try.  One even has a TiVo on it as well as a 
standard compter modem.

- --Michael
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.4 (NetBSD)

iD8DBQFAm8gLuWDhEvjSJrsRAqA7AKCmTQ7agjlD2XZbZbA3zoQUM+w63QCgvJEF
w2OacduFfg62wq46RKGGZUs=
=Blfu
-END PGP SIGNATURE-
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[Asterisk-Users] Routing by called interface

2004-05-07 Thread Chris Wilson

Hey everyone,

I want to run different lines directly to different extensions on two
FXO analog interfaces. ie; Zap/1 goes to Ext. 101, Zap/2 goes to
extensions 102


Does anyone know of a way to do this?

Thanks!

Chris

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RE: [Asterisk-Users] WI FI IP phones??

Have you looked on how much they cost.

On Fri, 2004-05-07 at 13:20, Mark Musone wrote:
 Why not vocera?
 
 http://www.vocera.com
 
 they seem to have the exact product you are looking for and seem to
 primarily server hospitals..
 
 -Mark
 
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of James Moran
 Sent: Friday, May 07, 2004 1:06 PM
 To: Asterisk
 Subject: Re: [Asterisk-Users] WI FI IP phones??
 
 Hmm I'll look into it. Thanks.
 
 On Fri, 2004-05-07 at 12:54, John Fraizer wrote:
  James Moran wrote:
  
   No I'm not but it's a hospital that nurses are on call and need to
 have
   a way to contact them.  On Fri, 2004-05-07 at 09:52, John Fraizer
 wrote:
   
  James Moran wrote:
  
  
  We need to have about 30 phones on one floor
  
  
  And you really think that WiFi phones are suited for this
 application? 
  Not an RF engineer, are ya?
  
  John
  
  Um, I'm not so sure that you're going to be able to run WiFi at a 
  hospital.  The life safety/support equipment is most likely not 
  certified to be resistant to 2.4Ghz interference.  It's been a while 
  since I looked up ISM allocations but, I can tell you that I've seen 
  many good ideas shot down because of the potential to interfere
 with 
  the medical equipment.
  
  John
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James Moran [EMAIL PROTECTED]
Potential Technologies

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Re: [Asterisk-Users] Trouble compiling latest CVS

2004-05-07 Thread Denis E. Pilon

I just created my own rpms...or you could have downloaded the fedora
srpm and rebuilt it.

DP

On Fri, 2004-05-07 at 12:57, Philipp von Klitzing wrote:
 Hi!
 
   Upgrade bison...i had the same problems until i upgraded bison.
  
  Which means upgrading glibc ... :-(( 
 
 Ok ok, I got it - compiled bison from source and disregarded those good-
 looking tail-shaking RPMs. ;- Works fine now.
 
 Cheers, Philipp
 
 
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Re: [Asterisk-Users] Asterisk and Cisco 7960 problems persist (for me, anyway)

2004-05-07 Thread bdolljr

Do you have a jitterbuffer enabled on your inter-asterisk IAX trunks? If so, try disabling it cleared everything up for me. With jitter buffer enabled using the default settings my audio across the IAX trunk was terrible. BTW, my 7960's are running 5.3 firmware so I probably don't see the timestamp sensitive 6.x packet drops that have been discussed here.

Bill

Brian Cuthie [EMAIL PROTECTED]

Brian Cuthie [EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
05/07/2004 08:57 AM

Please respond to
[EMAIL PROTECTED]

To
[EMAIL PROTECTED]

Subject
Re: [Asterisk-Users] Asterisk and Cisco 7960 problems persist (for me, anyway)

Ah, this reminds me that I forgot to mention that our network looks like
this:

Cisco --- SIP Asterisk IAX Aterisk IAX
Asterisk PRI PSTN

-brian

Tom wrote:

At 09:43 AM 5/7/2004, you wrote:

It seems that each time I get a new checkout of * from CVS my Cisco
7960 works worse than before. I know this stuff's in flux, so I
mention this in case it's news. Anyone else having trouble? What
I'm seeing (er, hearing) is really choppy audio. The previous version
I had installed had fairly frequent audio dropouts (not present when
I make the same calls through the same * box using a TDM400P interface).

No dropout problems or choppy audio running Asterisk
CVS-04/19/04-14:31:03 with 4 Cisco 7940/60 SIP 6.3 phones on a 2.4GHz
P4 Supermicro server. Analog phones through our TDM400P do sound much
better but the audio problems on our Cisco SIP phones are echo
problems. People are working on solutions.

Tom

Cheers,

Brian
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inline: graycol.gifinline: pic16827.gifinline: ecblank.gif

RE: [Asterisk-Users] WI FI IP phones??

2004-05-07 Thread Joseph Finley


I've actually engineered some WiFi at come medical clinics and it does
depend on the gear you purchase.  Cisco addresses this in their marketing
and technical spec sheets.  The two major hospitals in my area use wireless
for their phones and mobile laptops for the nurses as they go room to room
on a push cart.

Joe




-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Fraizer
Sent: Friday, May 07, 2004 12:54 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] WI FI IP phones??


James Moran wrote:

 No I'm not but it's a hospital that nurses are on call and need to 
 have a way to contact them.  On Fri, 2004-05-07 at 09:52, John Fraizer 
 wrote:
 
James Moran wrote:


We need to have about 30 phones on one floor


And you really think that WiFi phones are suited for this application?
Not an RF engineer, are ya?

John

Um, I'm not so sure that you're going to be able to run WiFi at a 
hospital.  The life safety/support equipment is most likely not 
certified to be resistant to 2.4Ghz interference.  It's been a while 
since I looked up ISM allocations but, I can tell you that I've seen 
many good ideas shot down because of the potential to interfere with 
the medical equipment.

John
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Re: [Asterisk-Users] SIP Wokflow diagram

2004-05-07 Thread Brancaleoni Matteo

I use callflow (callflow.sourceforge.net)

works under linux with ethereal dump, and produces
html+images pages, for viewing them via a web browser.

Matteo.

Il ven, 2004-05-07 alle 15:14, Ignace CARIA ha scritto:
 Hi everybody,
 
 I would like to create SIP call flow Diagram under Windows.  Is anybody 
 know a program to perform it?  I have already Ethereal and I would like 
 an explicit diagram just to show where something have problems...
 
 Thanks
 
 Ignace
 
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-- 
Brancaleoni Matteo [EMAIL PROTECTED]
Espia - Emmegi Srl

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[Asterisk-Users] CAPI Gain

2004-05-07 Thread Craig Waddington









I am using ISDN with CAPI and Eicon Diva card.



On ISDN calls in and out, some people are saying they find
it hard to hear us. Its only the odd few though, not everyone. We can hear them
no problem.



Do I just increase the txgain?



What is the limit for txgain, or are there any gotchas
for turning it up?



If you use the same what are your settings?



I have:



rxgain=0.4

txgain=1.5



Thanks.

Re: [Asterisk-Users] Routing by called interface

2004-05-07 Thread Jerimiah Cole

Chris Wilson wrote:
I want to run different lines directly to different extensions on two
FXO analog interfaces. ie; Zap/1 goes to Ext. 101, Zap/2 goes to
extensions 102
You should be able to stick different channels into different default 
contexts in zapata.conf.  Then just have the context for Zap/1 always 
dial extension 101, etc.  Makes sense, but I haven't tried this :)

Jerimiah
Tularosa Communications
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Re: [Asterisk-Users] WI FI IP phones??

2004-05-07 Thread listas iPfone

Symbol have the netvision line of  h.323 wireless phones used in hospitals
with multiple logins etc... , i have one here in my office and it works very
well with a  simple 3com officeconnect gateway, makes direct calls, have
integration with various pbx.. a good product.

www.symbol.com

Miklos



- Original Message - 
From: Mark Musone [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, May 07, 2004 2:20 PM
Subject: RE: [Asterisk-Users] WI FI IP phones??


 Why not vocera?

 http://www.vocera.com

 they seem to have the exact product you are looking for and seem to
 primarily server hospitals..

 -Mark


 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of James Moran
 Sent: Friday, May 07, 2004 1:06 PM
 To: Asterisk
 Subject: Re: [Asterisk-Users] WI FI IP phones??

 Hmm I'll look into it. Thanks.

 On Fri, 2004-05-07 at 12:54, John Fraizer wrote:
  James Moran wrote:
 
   No I'm not but it's a hospital that nurses are on call and need to
 have
   a way to contact them.  On Fri, 2004-05-07 at 09:52, John Fraizer
 wrote:
  
  James Moran wrote:
  
  
  We need to have about 30 phones on one floor
  
  
  And you really think that WiFi phones are suited for this
 application?
  Not an RF engineer, are ya?
  
  John
 
  Um, I'm not so sure that you're going to be able to run WiFi at a
  hospital.  The life safety/support equipment is most likely not
  certified to be resistant to 2.4Ghz interference.  It's been a while
  since I looked up ISM allocations but, I can tell you that I've seen
  many good ideas shot down because of the potential to interfere
 with
  the medical equipment.
 
  John
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 -- 
 James Moran [EMAIL PROTECTED]
 Potential Technologies

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RE: [Asterisk-Users] PRI, multi D channels and conventional PBXs

Looks like the pbx isn't sending any info such as called exten

bkw

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Lee Redmayne
 Sent: Friday, May 07, 2004 12:13 PM
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] PRI, multi D channels and conventional PBXs

 Hi all

 OK this may sound like a good one but maybe someone can tell me.

 Simple context is - I want to unplug my existing conventional PBX from the
 Telco and place * with it's TE410P in between.

 Now the difficult part, the existing connection is E1 PRI (Q.931) with 6
 B-channels.  I need to be able to trigger a D-channel to the old PBX and a
 D-Channel to the Telco (Not BT!).

 Next I can put the PBX onto a span 2, it triggers the D-channel and all
 seems hunky dory - until you try to acquire a line from * - this gives me:

  -- Extension '' in context 'blah' from '' does not exist.  Rejecting call
 on channel 6, span 2

 Any suggestions most welcome!
 Lee

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RE: [Asterisk-Users] WI FI IP phones??

2004-05-07 Thread Freddi Hansen

James Moran wrote:



We need to have about 30 phones on one floor

 



I have seen a couple of test where people claim that wi-fi phone network
should use max. 5 simultanoues calls per accesspoint or your audio will start to  
break up. I would take a look at www.kirk.com. They have a DECT basestation with
H323 interface. You can register all your hand set on a single station and then
use their DECT repeater to get the area coverage. Its expensive for a small system
but with 30 handsets it should be comparable to wifi phones and you get at least
5 times the standby/talktime.
my 2 cents.
Freddi 





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- Re: [Asterisk-Users] Routing by called interface - Email found in subject

That does work, I use that same approach to get analog extensions in a
norstar system to dial a specific sip phone in *. Works really well. We
then also tie the calleridname to which channel they dial out from as
well.

Gregory P. Scasny

Golden Technologies Inc.

http://www.golden-tech.com

219-462-7200


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jerimiah
Cole
Sent: Friday, May 07, 2004 1:01 PM
To: [EMAIL PROTECTED]
Subject: [SPAM] - Re: [Asterisk-Users] Routing by called interface -
Email found in subject


Chris Wilson wrote:
 I want to run different lines directly to different extensions on two
 FXO analog interfaces. ie; Zap/1 goes to Ext. 101, Zap/2 goes to
 extensions 102

You should be able to stick different channels into different default 
contexts in zapata.conf.  Then just have the context for Zap/1 always 
dial extension 101, etc.  Makes sense, but I haven't tried this :)

Jerimiah
Tularosa Communications

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[Asterisk-Users] Routing by called interface - Email found in subject


P.S. I can send examples of needed also.


Gregory P. Scasny

Golden Technologies Inc.

http://www.golden-tech.com

219-462-7200


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Wilson
Sent: Friday, May 07, 2004 12:34 PM
To: [EMAIL PROTECTED]
Subject: [SPAM] - [Asterisk-Users] Routing by called interface - Email
found in subject

Hey everyone,

I want to run different lines directly to different extensions on two
FXO analog interfaces. ie; Zap/1 goes to Ext. 101, Zap/2 goes to
extensions 102


Does anyone know of a way to do this?

Thanks!

Chris

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- RE: [Asterisk-Users] Missing digits on TDM400P incomplete dial string - DTMF problem? - Email found in subject