Re: [Asterisk-Users] Fax detection

[Tilghman Lesher]

On Wednesday 17 December 2003 00:34, Alexander Romanov wrote:
> Is there any way to turn off fax detection?

Don't create a fax extension.

-Tilghman

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[Asterisk-Users] Fax detection

[Alexander Romanov]

Is there any way to turn off fax detection?


Alex.

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Re: [Asterisk-Users] IAX2 using non standard port

[Dan]

Hi,

>- Original Message - 
>From: "Walker Haddock" <[EMAIL PROTECTED]>
>Subject: Re: [Asterisk-Users] IAX2 using non standard port
> Solution:
>
> edit iax2.h file and change line 73 as follows:
> #define IAX_DEFAULT_PORTNO  80/* 4569 */
>
> in /usr/src/asterisk
>
> make clean ; make install
>
> in iax.conf, in the register record, append the port number of the peer:
> register => user:[EMAIL PROTECTED]:4569
>
> Walker

Now I think to something more...
What about making Asterisk to listen on multiple ports for IAX2?
Let's say for example the default 4569 and 80?
If IAX2 is to replace IAX, why the port number cannot be configured from
iax.conf file?
There is any technical reason? It is not a good ideea to make changes in the
source file for that.

BR,
Dan


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[Asterisk-Users] Unable to Receive Fax -- RxFAX Application

[Kannaiyan Natesan]

Hi,

  Below if the error message which I got from asterisk.
  I was trying to fax to asterisk from my fax machine. I really dunno what
is the problem. I use alaw & ulaw codec only through my ATA 186. Can anyone
help me what could be the problem.


-- Executing Goto("SIP/-080ef9a0", "13732|s|1") in new stack
-- Goto (13732,s,1)
-- Executing Wait("SIP/-080ef9a0", "2") in new stack
-- Executing Answer("SIP/-080ef9a0", "") in new stack
-- Executing RxFAX("SIP/-080ef9a0", "/root/kans.tif") in new stack
Changed from phase 0 to 1
NOTICE[278543]: File rtp.c, Line 264 (process_rfc3389): RFC3389 support
incomplete.  Turn off on client if possible
NOTICE[278543]: File rtp.c, Line 264 (process_rfc3389): RFC3389 support
incomplete.  Turn off on client if possible
Start receiving document
Changed from phase 1 to 4
Sending ident
>>> CSI: 40 38 37 36 35 34 33 32 31 20 20 20 20 20 20 20 20 20 20 20 20
DIS:
Store and forward Internet fax: no
Real-time Internet fax: no
Preferred octets: 256
Can receive fax
Data signalling rate: V.29
R8x7.7lines/mm and/or 200x200pels/25.4mm OK
2D coding OK
Scan line length: 215mm
Recording length: A4 (297mm)
Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85
R8x15.4lines/mm OK
Inch-based resolution preferred: no
Metric-based resolution preferred: no
Minimum scan line time for higher resolutions: T15.4 = T7.7
>>> DIS: 80 00 c6 f0 80 80 01
NOTICE[278543]: File rtp.c, Line 418 (ast_rtp_read): Unknown RTP codec 100
received

Thanks in advance,
Kannaiyan




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Re: [Asterisk-Users] more questions

[Tilghman Lesher]

On Tuesday 16 December 2003 20:27, Isianto Istiadi wrote:
> > Have you placed any mp3 files in /var/lib/asterisk/mohmp3 ?
>
> There is one mp3 (default/testing) that come with * installations
> right? It has that one. The problem is when I do ps -ax, there's no
> reference to mpg123 at all, and in the debug log in *, I see that *
> tries to call the mpg123, but it dies

So what is the current output of the following command?

echo | strings `which mpg123` | grep -c mpg123

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Re: [Asterisk-Users] AT&T access code entry by Asterisk

[Tilghman Lesher]

On Tuesday 16 December 2003 23:04, Walker Haddock wrote:
> On Tue, Dec 16, 2003 at 07:04:47PM -0600, Brian West wrote:
> > Its an upppercase W

Actually, it's both.  The 'w' is case-insensitive.

> OK, still no joy!  Here's the way I tried it:
>
> exten => _91NX,1,Dial(ZAP/g1/${EXTEN}WWW555,70)

You may want to time yourself as to how long the system waits before
prompting for the code.  Each 'w' waits exactly 0.5 seconds.  So if you
need to wait 5 seconds before inputting the code, then you'll need 10
'w's.

-Tilghman

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Re: [Asterisk-Users] IAX2 using non standard port

[Walker Haddock]

On Wed, Dec 17, 2003 at 12:21:28AM -0500, William Waites wrote:
> On Tue, Dec 16, 2003 at 10:59:50PM -0600, Walker Haddock wrote:
> > 
> > edit iax2.h file and change line 73 as follows:
> > #define IAX_DEFAULT_PORTNO  80/* 4569 */
> 
> this is *really* the *wrong* way to fix it.
> the correct way is to set port = 80 in iax.conf

I agree.  The only problem is that IAX grabs this.  Even when I put noload => 
chan_iax.so, it still would not bind to port 80.  that's when I went looking in the 
source.

> 
> BUT...
> 
> you will notice near the beginning of the load_module()
> routine in chan_iax2.c, it does
> 
> sin.sin_port = ntohs(IAX_DEFAULT_PORTNO);
> sin.sin_addr.s_addr = INADDR_ANY;

I saw that.

> 
> in other words, the local address the iax2 process 
> binds to, as well as the port, are hardcoded in
> the source.
> 
> not good. these should come from the config file
> with INADDR_ANY and IAX_DEFAULT_PORTNO as defaults.

I was thinking it should work this way. 

> 
> can you create a bug for this on bugs.digium.com?

I'd be glad to, I haven't opened a bug before.  Do I just log in and state these facts?

Walker

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Re: [Asterisk-Users] IAX2 using non standard port

[William Waites]

On Tue, Dec 16, 2003 at 10:59:50PM -0600, Walker Haddock wrote:
> 
> edit iax2.h file and change line 73 as follows:
> #define IAX_DEFAULT_PORTNO  80/* 4569 */

this is *really* the *wrong* way to fix it.
the correct way is to set port = 80 in iax.conf

BUT...

you will notice near the beginning of the load_module()
routine in chan_iax2.c, it does

sin.sin_port = ntohs(IAX_DEFAULT_PORTNO);
sin.sin_addr.s_addr = INADDR_ANY;

in other words, the local address the iax2 process 
binds to, as well as the port, are hardcoded in
the source.

not good. these should come from the config file
with INADDR_ANY and IAX_DEFAULT_PORTNO as defaults.

can you create a bug for this on bugs.digium.com?

-w
-- 
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\ /No HTML/RTF in email
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/ \  Respect for open standards
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Re: AW: [Asterisk-Users] CLIP in Germany

[Richard Scobie]

Johannes von Drachenfels wrote:
Hi,

i have a Siemens Gigaset phone with clip-capabilities on the tdm400 with
extension 610 installed. Extension 600 is another phone. But i still cannot
see the number of the caller at the display of the Siemens phone... here is
my config:
; Main extension
exten => 600,1,SetCallerID,${CALLERIDNUM}
exten => 600,2,Dial,Zap/33|30
; Johannes von Drachenfels's Extension
exten => 610,1,SetCallerID,${CALLERIDNUM}
exten => 610,2,Dial,Zap/32|30
exten => 610,3,VoiceMail,u610
exten => 610,103,VoiceMail,b610
... and the debug of a call from 600 to 610:

cm*CLI>
-- Starting simple switch on 'Zap/33-1'
-- Executing SetCallerID("Zap/33-1", "600") in new stack
-- Executing Dial("Zap/33-1", "Zap/32|30") in new stack
-- Called 32
-- Zap/32-1 is ringing
-- Zap/32-1 is ringing
-- Zap/32-1 is ringing
So what i would like to see is 600 in the display of 610 ...

Hi,

I have CID working fine on TDM400s with Gigasets. I do not have the 
SetCallerID lines in my extensions.conf.

You need to add a line in zapata.conf

usecallerid=yes

above your TDM400 channel declarations which also have the caller ID info.

Here is part of mine:

usecallerid=yes
busydetect=no
group=2
context=default
signalling=fxo_ks
threewaycalling=yes
transfer=yes
cancallforward=yes
callerid="Avid" <500>
channel => 3
callerid="Terry" <501>
channel => 4
Hope this helps.

Richard



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Re: [Asterisk-Users] Stupid Newbie Questions

[Steven Critchfield]

On Tue, 2003-12-16 at 22:32, TkVoice Author wrote:
> Thanks TC
> 
> I will check this out.
> 
> I did notice 1 error if anyone is concerned:
> 
> I patched against asterisk-v0.5.0 and I got this result:

You probably need to be at a more current version than 0.5.0, and you
should be concerned that the offsets are so far off since this means
there was major changes since the diff was made.

> patch -p0 < distinctiveringpatch.diff
> 
> patching file asterisk/channels/chan_zap.c
> Hunk #1 succeeded at 134 (offset -1 lines).
> Hunk #2 succeeded at 317 (offset -6 lines).
> Hunk #3 succeeded at 381 (offset -1 lines).
> Hunk #4 succeeded at 504 (offset -14 lines).
> Hunk #5 succeeded at 3869 (offset -96 lines).
> Hunk #6 succeeded at 4387 (offset -18 lines).
> Hunk #7 FAILED at 4416.
> Hunk #8 succeeded at 4368 (offset -98 lines).
> Hunk #9 succeeded at 5256 (offset -55 lines).
> Hunk #10 succeeded at 5240 (offset -100 lines).
> Hunk #11 succeeded at 6786 (offset -210 lines).
> Hunk #12 succeeded at 6976 (offset -100 lines).
> Hunk #13 succeeded at 7254 (offset -221 lines).
> Hunk #14 succeeded at 7405 (offset -100 lines).
> Hunk #15 succeeded at 7377 (offset -221 lines).
> 1 out of 15 hunks FAILED -- saving rejects to file 
> asterisk/channels/chan_zap.c.rej
> 
> TC wrote:
> 
> >You can try this hacking this patch
> >http://bugs.digium.com/bug_view_page.php?bug_id=504
> >  
> >
-- 
Steven Critchfield <[EMAIL PROTECTED]>

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[Asterisk-Users] Asterisk Crash

[Kevin]

Asterisk Crash

I have an application that using the System() command.  When ever I
invoke the command my asterisk crashes.

I have updated to the latest CVS and it crashes.  Can someone who has
the most recent CVS help me to verify if this is a code issue or my
setup?

Thanks
 
Extensions.conf


exten => 2810,1,System(date)
exten => 2810,2,Goodbye




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Re: [Asterisk-Users] AT&T access code entry by Asterisk

[Walker Haddock]

On Tue, Dec 16, 2003 at 07:04:47PM -0600, Brian West wrote:
> Its an upppercase W
> 
> bkw

OK, still no joy!  Here's the way I tried it:

exten => _91NX,1,Dial(ZAP/g1/${EXTEN}WWW555,70)

asterisk*CLI> show version
Asterisk CVS-12/11/03-17:07:52 built by [EMAIL PROTECTED] on a i686 running Linux

Thanks, must be something I'm doing wrong!

> 
> On Tue, 16 Dec 2003, Walker Haddock wrote:
> 
> > I have a dialplan that requires that we use * to send the long distance access 
> > code to AT&T.  I have found in the list that the `w` command can be used to inject 
> > a pause, I have tried the following:
> >
> > exten => _91NX,1,Dial(ZAP/g1/${EXTEN}www555,70)
> >
> > There `555` is the ld access code.  I tried various quantities of `w`s but I 
> > never got * to dial the ld access code.  Allof the cases I saw the `w` token 
> > discussed was at the beginning of the dialed digits.  I tried putting a `w` at 
> > before the ${EXTEN} and it did appear to wait a little before dialing the digits.  
> > Maybe it doesn't work inside the dial string?
> >
> > I also tried the recommendation to use the Flash, SendDTMF, Wait, SendDTMF, but 
> > that didn't work at all.
> >
> > I need to implement this functionality in the dialplan.  Does anyone have any 
> > suggestions?
-- 
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Walker Haddock   http://www.datacrest.com
DataCrest, Inc.e-mail:  [EMAIL PROTECTED]
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Re: [Asterisk-Users] IAX2 using non standard port

[Walker Haddock]

Thanks Mr. Tilghman!  Here's a summary incase anyone else is looking for the way to do 
this:

Problem:  Need to use non standard port for IAX2 server listening behind a firewall.

In this case, we were required to use UDP port 80 to listen on.

Solution:

edit iax2.h file and change line 73 as follows:
#define IAX_DEFAULT_PORTNO  80/* 4569 */

in /usr/src/asterisk

make clean ; make install

in iax.conf, in the register record, append the port number of the peer:
register => user:[EMAIL PROTECTED]:4569

Walker
On Tue, Dec 16, 2003 at 07:55:13PM -0600, Tilghman Lesher wrote:
> On Tuesday 16 December 2003 18:56, Walker Haddock wrote:
> > I am working on a project where we must connect to the * server
> > behind a firewall.  They can only open up port 80 udp.
> >
> > I have managed to get IAX working fine by specifying the port=80 in
> > the iax.conf file.  However I would like to get IAX2 working.
> >
> > There was a recommendation in the list 13 SEP 2003:
> > http://lists.digium.com/pipermail/asterisk-users/2003-September/02056
> >0.html
> >
> > I tried to change the IAX_DEFAULT_PORTNO in iax2.h to 80, but then
> > the internal server also tried to talk to the external server on port
> > 80.  Looks like IAX_DEFAULT_PORTNO sets both the origin and
> > destination port numbers.
> >
> > So, if I want the internal IAX2 service to listen on port 80 and I
> > want to register with the external IAX2 server using the standard
> > port, is there another change I need to make to the IAX2 code?
> 
> You should be able to specify the port number in the register line:
> 
> register => user:[EMAIL PROTECTED]:4569
> 
> -Tilghman
-- 
   DataCrest, Inc. -- Technically Superior   **
Walker Haddock   http://www.datacrest.com
DataCrest, Inc.e-mail:  [EMAIL PROTECTED]
1634A Montgomery Hwy.phone:  1-888-941-3282, 1-205-335-8589
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[Asterisk-Users] RE: Help! VoiceTronix Multi FXO/FXS Problem (Jacky)

[Paul Bagyenda]

Hi,

 Firstly it appears you are not passing the extension to be dialed to 
the Dial command, so the driver is dialing badly. It should be 
something like dial(vpb/1-2/102) since this is a line port. Admittedly 
the driver needs to check better that the number it is trying to dial 
is legit but it doesn't :)

P.

--- You Wrote ---
From: "Jacky" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Date: Tue, 16 Dec 2003 19:05:27 +0800
Subject: [Asterisk-Users] Help! VoiceTronix Multi FXO/FXS Problem
Reply-To: [EMAIL PROTECTED]
Hello, Hacker

I install VoiceTronix OpenSwitch 12 port PCI Telephone Card,
and setting vpb.conf, extensions.conf following
My problem is:

When i dial to fxo(channel 9-12), it is ok,
but when i continue press exten 102, the channel crach with error 
messages
following
exception caught: VPBAPI_DIAL_INVALID_LEVEL, file: vpbdial.cpp line:872

Do i ignore some setting for VoiceTronix OpenSwitch12 Card, Or other 
issues?

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Re: [Asterisk-Users] command Authenticate

[Juan J. Sierralta P.]

On Tue, 2003-12-16 at 18:24, SW wrote:
> Hi folks,
> 
> Sorry to post bunch of messages today, I am deep into this and hope any help
> from the experts.
> 
> I am using the command Authenticate as explained in wi-ki:
> 
> 
> ===
> Authenticate(password[|options])
> 
> Requires a user to enter a given password in order to continue execution. If
> the password begins with the '/' character, it is interpreted as a file
> which contains a list of valid passwords (1 per line).
> 
> An optional set of options may be provided by concatenating any of the
> following letters:
> 
> 
> a - Set account code to the password that is entered

It means Authenticate(2342,a) will use "2342" as the account code, so
you can do something like

exten => 4,1,Authenticate(2342)
exten => 4,2,SetAccount(abc)


-- 
Juanjo sin .sig

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Re: [Asterisk-Users] broken pipe - * does not respond

[Juan J. Sierralta P.]

On Wed, 2003-12-17 at 00:57, Nicolas wrote:
> After a little investigation, I noticed that after each call on the CAPI 
> server, two file Descriptor stays but the strange thing is what they are 
> linked to:
> 
> /proc/15118/fd> ls -l
> lr-x--1 root root   64 dÃc 17 04:50 25 -> pipe:[8022904]
> l-wx--1 root root   64 dÃc 17 04:50 26 -> pipe:[8022904]
> lr-x--1 root root   64 dÃc 17 04:50 35 -> pipe:[8025060]
> l-wx--1 root root   64 dÃc 17 04:50 36 -> pipe:[8025060]
> 
> I left only two examples, I can notice that this is a pipe within 
> asterisk that is not cleared but I don't have any knowlegde to look into 
> the code for this type of error. Could someone give me an hint?

Maybe to trace the bug, decrease the number of file descriptors with
ulimit or something and do a tail -f to the logs, probably * will start
to complain when open(3) returns an error.
Anyway there must be something suspicious at the logs.
If that particular open(3) wasnÂt surrounded by the respective
if/fprintf to log an error strace is your friend.

-- 
Juanjo sin .sig

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Re: [Asterisk-Users] Stupid Newbie Questions

[TkVoice Author]

Thanks TC

I will check this out.

I did notice 1 error if anyone is concerned:

I patched against asterisk-v0.5.0 and I got this result:

patch -p0 < distinctiveringpatch.diff

patching file asterisk/channels/chan_zap.c
Hunk #1 succeeded at 134 (offset -1 lines).
Hunk #2 succeeded at 317 (offset -6 lines).
Hunk #3 succeeded at 381 (offset -1 lines).
Hunk #4 succeeded at 504 (offset -14 lines).
Hunk #5 succeeded at 3869 (offset -96 lines).
Hunk #6 succeeded at 4387 (offset -18 lines).
Hunk #7 FAILED at 4416.
Hunk #8 succeeded at 4368 (offset -98 lines).
Hunk #9 succeeded at 5256 (offset -55 lines).
Hunk #10 succeeded at 5240 (offset -100 lines).
Hunk #11 succeeded at 6786 (offset -210 lines).
Hunk #12 succeeded at 6976 (offset -100 lines).
Hunk #13 succeeded at 7254 (offset -221 lines).
Hunk #14 succeeded at 7405 (offset -100 lines).
Hunk #15 succeeded at 7377 (offset -221 lines).
1 out of 15 hunks FAILED -- saving rejects to file 
asterisk/channels/chan_zap.c.rej

TC wrote:

You can try this hacking this patch
http://bugs.digium.com/bug_view_page.php?bug_id=504
 

--
Peter
http://tkvoice.netfirms.com/
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Re: [Asterisk-Users] broken pipe - * does not respond

[Nicolas]

After a little investigation, I noticed that after each call on the CAPI 
server, two file Descriptor stays but the strange thing is what they are 
linked to:

/proc/15118/fd> ls -l
lr-x--1 root root   64 déc 17 04:50 25 -> pipe:[8022904]
l-wx--1 root root   64 déc 17 04:50 26 -> pipe:[8022904]
lr-x--1 root root   64 déc 17 04:50 35 -> pipe:[8025060]
l-wx--1 root root   64 déc 17 04:50 36 -> pipe:[8025060]
I left only two examples, I can notice that this is a pipe within 
asterisk that is not cleared but I don't have any knowlegde to look into 
the code for this type of error. Could someone give me an hint?

Thank you for your help
Nicolas Bruxer
Nicolas wrote:

Hi,
My release is the last one from the CVS, I get the same problem but with 
a 4 BRI CAPI card (EICON DIVA SERVER 4BRI).
The number of file descriptor keeps growing with each call until the 
system hangs.
Currently in my office I have two asterisk servers, one for a gateway 
between 7960/SIP<->IAX and the other one IAX<->CAPI/BRI.
So: SIP<->IAX<->IAX<->CAPI

The strange thing is that only the second one(CAPI) crashes.
I tried this configuration of 2 servers to isolate this problem but with 
no luck although it has lowered the frequency of crashes!

For instance in the log file:
Dec  8 10:46:52 WARNING[524306]: File channel.c, Line 289 
(ast_channel_alloc): Alert pipe creation failed!

Dec  8 10:46:52 ERROR[524306]: File chan_capi.c, Line 1051 (capi_new): 
Unable to allocate channel!

Dec  8 10:46:52 NOTICE[524306]: File app_dial.c, Line 502 (dial_exec): 
Unable to create channel of type 'CAPI'

and it repeats many times until the caller stops trying or I restart the 
server.

Yours sincerly
Nicolas
Paulo H. Mannheimer wrote:

My release is about 2 month old. I don´t want to mess too much with my 
client´s site - it is a 100 seat call center with dozens of softphones 
and some app_queue.c patches we have programmed to register a couple 
of agents´ events.
 

How old is your release.. and why not update?  I'm using the latest 
cvs on
a few projects and its fine.

bkw

On Tue, 16 Dec 2003, Paulo H. Mannheimer wrote:


Hi, I´m having a serious problem at one customer. After 6 hours 
answering a


PRI

line, * stops responding in a very similar situation as described 
here ...

http://lists.digium.com/pipermail/asterisk-users/2003-July/015391.html

I took a look at "/proc/first * PID/fd" and there are hundreds of file
descriptors;
If I try to connect using asterisk -r I get the "broken pipe"  error.

I tried to find anything related at bugs.digium.com but couldn´t 
find any
mention to this specific situation. My CVS version is quite old, but I


would

rather update it by a specific patch than to replace it entirely by 
a new


one.

Any hint would be greatly appreciated.

Best regards,

PauloHM



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Re: [Asterisk-Users] broken pipe - * does not respond

[Nicolas]

Hi,
My release is the last one from the CVS, I get the same problem but with 
a 4 BRI CAPI card (EICON DIVA SERVER 4BRI).
The number of file descriptor keeps growing with each call until the 
system hangs.
Currently in my office I have two asterisk servers, one for a gateway 
between 7960/SIP<->IAX and the other one IAX<->CAPI/BRI.
So: SIP<->IAX<->IAX<->CAPI

The strange thing is that only the second one(CAPI) crashes.
I tried this configuration of 2 servers to isolate this problem but with 
no luck although it has lowered the frequency of crashes!

For instance in the log file:
Dec  8 10:46:52 WARNING[524306]: File channel.c, Line 289 
(ast_channel_alloc): Alert pipe creation failed!

Dec  8 10:46:52 ERROR[524306]: File chan_capi.c, Line 1051 (capi_new): 
Unable to allocate channel!

Dec  8 10:46:52 NOTICE[524306]: File app_dial.c, Line 502 (dial_exec): 
Unable to create channel of type 'CAPI'

and it repeats many times until the caller stops trying or I restart the 
server.

Yours sincerly
Nicolas
Paulo H. Mannheimer wrote:

My release is about 2 month old. I don´t want to mess too much with my client´s 
site - it is a 100 seat call center with dozens of softphones and some 
app_queue.c patches we have programmed to register a couple of agents´ events.
  


How old is your release.. and why not update?  I'm using the latest cvs on
a few projects and its fine.
bkw

On Tue, 16 Dec 2003, Paulo H. Mannheimer wrote:


Hi, I´m having a serious problem at one customer. After 6 hours answering a
PRI

line, * stops responding in a very similar situation as described here ...

http://lists.digium.com/pipermail/asterisk-users/2003-July/015391.html

I took a look at "/proc/first * PID/fd" and there are hundreds of file
descriptors;
If I try to connect using asterisk -r I get the "broken pipe"  error.

I tried to find anything related at bugs.digium.com but couldn´t find any
mention to this specific situation. My CVS version is quite old, but I
would

rather update it by a specific patch than to replace it entirely by a new
one.

Any hint would be greatly appreciated.

Best regards,

PauloHM



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[Asterisk-Users] Flash Transfer/Voicemail Bug

[Andy Hester]

Hi All,
Is anyone working on bug id 617?
http://bugs.digium.com/bug_view_page.php?bug_id=617
I am not sure where to start and I really need to get this fixed.  Any
ideas?

Sincerely,
Andy Hester
Consero

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Re: [Asterisk-Users] more questions

[Isianto Istiadi]

Have you placed any mp3 files in /var/lib/asterisk/mohmp3 ?
-Tilghman

There is one mp3 (default/testing) that come with * installations right? It 
has that one. The problem is when I do ps -ax, there's no reference to mpg123 
at all, and in the debug log in *, I see that * tries to call the mpg123, but 
it dies. Also I don't have a sound card in * box, is it related?
I'm sorry if it sound confusing, I lack the English proficient.
Thanks


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RE: [Asterisk-Users] DISA - Zap/DTMF Problem

[Alexander Romanov]

Please Point me to the right direction.

Regards,
Alex.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alexander
Romanov
Sent: Wednesday, 17 December 2003 12:02 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] DISA - Zap/DTMF Problem


Hi guys,

I am trying to use DISA. The scenario is - I call my home number (where
X100P seats) from mobile phone, enter the password, enter international
number and get connected via voiptel. It works perfectly when I call
extension setup with DISA from X-PRO SIP phone, but when I dial into
Zap, It seems that it does not detect DTMF tones. Here is a log and
config files Please help

[[Zaptel.conf]]

fxsks=1
loadzone=au
defaultzone=au

[[Zapata.conf]]

[channels]
busydetect=yes
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
usecallerid=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
rxgain=3.0
txgain=3.0
immediate=no
context=inbound-analog
signalling=fxs_ks
callerid=asreceived
channel=>1


[[extensions.conf]]

[inbound-analog]
exten => s,1,DISA,1234|voiptel
exten => s,2,Hangup


And finaly the log ...

*CLI> -- Starting simple switch on 'Zap/1-1'
DEBUG[1255338944]: File chan_zap.c, Line 1069 (zt_enable_ec): Enabled
echo cancellation on channel 1
-- Executing DISA("Zap/1-1", "1234|voiptel") in new stack
DEBUG[1255338944]: File chan_zap.c, Line 1961 (zt_answer): Took Zap/1-1
off hook
DEBUG[1255338944]: File chan_zap.c, Line 3511 (zt_read): DTMF digit: f
on Zap/1-1
NOTICE[1255338944]: File chan_zap.c, Line 3545 (zt_read): Fax detected,
but no fax extension
DEBUG[1255338944]: File chan_zap.c, Line 3511 (zt_read): DTMF digit: f
on Zap/1-1
DEBUG[1255338944]: File chan_zap.c, Line 3549 (zt_read): Fax already
handled
DEBUG[1255338944]: File chan_zap.c, Line 3511 (zt_read): DTMF digit: f
on Zap/1-1
DEBUG[1255338944]: File chan_zap.c, Line 3549 (zt_read): Fax already
handled
DEBUG[1255338944]: File chan_zap.c, Line 3511 (zt_read): DTMF digit: f
on Zap/1-1
DEBUG[1255338944]: File chan_zap.c, Line 3549 (zt_read): Fax already
handled
DEBUG[1255338944]: File chan_zap.c, Line 3511 (zt_read): DTMF digit: f
on Zap/1-1
DEBUG[1255338944]: File chan_zap.c, Line 3549 (zt_read): Fax already
handled
DEBUG[1255338944]: File chan_zap.c, Line 3511 (zt_read): DTMF digit: f
on Zap/1-1
DEBUG[1255338944]: File chan_zap.c, Line 3549 (zt_read): Fax already
handled
DEBUG[1255338944]: File chan_zap.c, Line 3511 (zt_read): DTMF digit: f
on Zap/1-1
DEBUG[1255338944]: File chan_zap.c, Line 3549 (zt_read): Fax already
handled
DEBUG[1255338944]: File chan_zap.c, Line 3511 (zt_read): DTMF digit: f
on Zap/1-1
DEBUG[1255338944]: File chan_zap.c, Line 3549 (zt_read): Fax already
handled
DEBUG[1255338944]: File chan_zap.c, Line 3511 (zt_read): DTMF digit: f
on Zap/1-1
DEBUG[1255338944]: File chan_zap.c, Line 3549 (zt_read): Fax already
handled
DEBUG[1255338944]: File chan_zap.c, Line 3511 (zt_read): DTMF digit: f
on Zap/1-1
DEBUG[1255338944]: File chan_zap.c, Line 3549 (zt_read): Fax already
handled
DEBUG[1255338944]: File chan_zap.c, Line 3511 (zt_read): DTMF digit: f
on Zap/1-1
DEBUG[1255338944]: File chan_zap.c, Line 3549 (zt_read): Fax already
handled
DEBUG[1255338944]: File chan_zap.c, Line 3511 (zt_read): DTMF digit: f
on Zap/1-1
DEBUG[1255338944]: File chan_zap.c, Line 3549 (zt_read): Fax already
handled
DEBUG[1255338944]: File chan_zap.c, Line 3511 (zt_read): DTMF digit: f
on Zap/1-1
DEBUG[1255338944]: File chan_zap.c, Line 3549 (zt_read): Fax already
handled
DEBUG[1255338944]: File chan_zap.c, Line 3511 (zt_read): DTMF digit: f
on Zap/1-1
DEBUG[1255338944]: File chan_zap.c, Line 3549 (zt_read): Fax already
handled
DEBUG[1255338944]: File chan_zap.c, Line 3511 (zt_read): DTMF digit: f
on Zap/1-1
DEBUG[1255338944]: File chan_zap.c, Line 3549 (zt_read): Fax already
handled
DEBUG[1255338944]: File chan_zap.c, Line 3511 (zt_read): DTMF digit: f
on Zap/1-1
DEBUG[1255338944]: File chan_zap.c, Line 3549 (zt_read): Fax already
handled
DEBUG[1255338944]: File app_disa.c, Line 185 (disa_exec): DISA password
entry timeout on chan Zap/1-1
DEBUG[1255338944]: File chan_zap.c, Line 3511 (zt_read): DTMF digit: f
on Zap/1-1
DEBUG[1255338944]: File chan_zap.c, Line 3549 (zt_read): Fax already
handled


-
Alex.

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Re: [Asterisk-Users] more questions

[Tilghman Lesher]

On Tuesday 16 December 2003 19:47, Isianto Istiadi wrote:
> I tried to run my music on hold configuration but
> to no avail. Here's my musiconhold.conf:
>
> [classes]
> default => quietmp3:/var/lib/asterisk/mohmp3
> loud => mp3:/var/lib/asterisk/mohmp3
> random => quietmp3:/var/lib/asterisk/mohmp3,-z

Have you placed any mp3 files in /var/lib/asterisk/mohmp3 ?

-Tilghman

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Re: [Asterisk-Users] IAX2 using non standard port

[Tilghman Lesher]

On Tuesday 16 December 2003 18:56, Walker Haddock wrote:
> I am working on a project where we must connect to the * server
> behind a firewall.  They can only open up port 80 udp.
>
> I have managed to get IAX working fine by specifying the port=80 in
> the iax.conf file.  However I would like to get IAX2 working.
>
> There was a recommendation in the list 13 SEP 2003:
> http://lists.digium.com/pipermail/asterisk-users/2003-September/02056
>0.html
>
> I tried to change the IAX_DEFAULT_PORTNO in iax2.h to 80, but then
> the internal server also tried to talk to the external server on port
> 80.  Looks like IAX_DEFAULT_PORTNO sets both the origin and
> destination port numbers.
>
> So, if I want the internal IAX2 service to listen on port 80 and I
> want to register with the external IAX2 server using the standard
> port, is there another change I need to make to the IAX2 code?

You should be able to specify the port number in the register line:

register => user:[EMAIL PROTECTED]:4569

-Tilghman

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Re: [Asterisk-Users] IAX2 using non standard port

[Adam Hart]

I find reading what people have said actually improves my ability to help
them :p

-Adam


> Just use iax2 since iax2 can do everything iax can and then some.
>
> bkw
>
> On Tue, 16 Dec 2003, Walker Haddock wrote:
>
> > I am working on a project where we must connect to the * server behind a
firewall.  They can only open up port 80 udp.
> >
> > I have managed to get IAX working fine by specifying the port=80 in the
iax.conf file.  However I would like to get IAX2 working.
> >
> > There was a recommendation in the list 13 SEP 2003:
> >
http://lists.digium.com/pipermail/asterisk-users/2003-September/020560.html
> >
> > I tried to change the IAX_DEFAULT_PORTNO in iax2.h to 80, but then the
internal server also tried to talk to the external server on port 80.  Looks
like IAX_DEFAULT_PORTNO sets both the origin and destination port numbers.
> >
> > So, if I want the internal IAX2 service to listen on port 80 and I want
to register with the external IAX2 server using the standard port, is there
another change I need to make to the IAX2 code?
> >
> > Thanks
> > --
> >    DataCrest, Inc. -- Technically Superior   **
> > Walker Haddock   http://www.datacrest.com
> > DataCrest, Inc.e-mail:  [EMAIL PROTECTED]
> > 1634A Montgomery Hwy.phone:  1-888-941-3282, 1-205-335-8589
> > Birmingham, AL 35216  fax:  1-205-823-7838
> > ***
> > ___
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> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
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Re: [Asterisk-Users] more questions

[Isianto Istiadi]

Dear all,
first of all, I want to say thanks to Tilghman, I don't mean to disrespect/
doubt your answer, I just want to understand the logic more. (I'm sorry)
I tried to run my music on hold configuration but to no avail.
Here's my musiconhold.conf:

[classes]
default => quietmp3:/var/lib/asterisk/mohmp3
loud => mp3:/var/lib/asterisk/mohmp3
random => quietmp3:/var/lib/asterisk/mohmp3,-z

and here's my related zapata.conf:

musiconhold=default

I'm using slackware 9.1, and did uninstall mpg321 from slackware and install 
the mpg123, I checked that mpg123 is in /usr/bin, but when I put the call on 
hold, nothing happend. I look at the * log, and it says that mpg123 is died.
The questions:
1. Do musiconhold required me to have a soundcard in * ( I don't have a sound 
card in my * box)
2. If it doesn't need one, what did I do wrong?
3. I want to ask you guys, is it normal, when I press *8# to pick up call, the 
* says transfering, and then followed by invalid extension, but it does let 
me pick up the call afterward. And in * console, it says that it's playing 
transfering voice, and then something about extension ' '. If it's not normal 
how do I fix that?

Thanks again

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[Asterisk-Users] Transcoding CPU usage: surveys?

[John Todd]

Before I put myself to the task (next month, maybe) of surveying the 
CPU costs of transcoding, perhaps someone else has already done this 
work and would be willing to share it or refer me to a link of 
previously published data.  My reviews of the mailing list with 
various keywords were unsuccessful in finding adequate references, 
though I admit I only spent 20 minutes looking.

What I seek is this:

  Using a "standard unit of processing power", such as a 2ghz P4 
processor with a bottleneck-free 100baseT connection between 
endpoints, at what point does voice quality noticeably degrade in the 
following circumstances:

   1) SIP G.711 to IAX2 GSM
   2) SIP G.711 to IAX2 iLBC
   3) SIP G.711 to IAX2 g.729
   4) Zap to IAX2 GSM
   5) Zap to IAX2 iLBC
   6) Zap to IAX2 g.729
   7) Zap to IAX2 g.711
   8) Zap to SIP GSM
   9) Zap to SIP iLBC
  10) Zap to SIP g.729
  11) Zap to SIP g.711
  12) TDMoE to SIP G.711
  13) TDMoE to SIP GSM
  14) TDMoE to SIP iLBC
  15) TDMoE to SIP g.729
  16) TDMoE to IAX2 GSM
  17) TDMoE to IAX2 iLBC
  18) TDMoE to IAX2 g.729
  19) TDMoE to IAX2 g.711
  20) Reverse tests 1-20 to see if there is a difference

I exclude IAX2 to IAX2 transcoding, since I think that SIP to IAX2 
should be close enough to compare.  I also exclude passthrough 
traffic (no transcoding) on all protocols though that would be 
interesting if anyone has it.

I realize that some of these tests may be redundant, and some have 
such minimal differences as to make testing somewhat silly.  However, 
any subset of the data would be great to have in a "known" testing 
environment instead of rumor, so I exhaustively list them for my 
future test construction.

The term "noticeably degrade" is of course a matter of opinion, but 
your opinions are still better than nothing.  :-)  I am very 
interested in anyone who can quote me raw figures for packets-only 
network.  In other words, no Zap or TDMoE channels - just SIP-to-SIP, 
or SIP-to-IAX2, or IAX2-to-IAX2 type of circumstances, as I think 
these are more representative of what the processor (versus the 
backplane and IRQ peculiarities) represent in possible throughput.  I 
honestly don't care what processor or speed you tell me you've used, 
as long as you tell me exactly what the system looked like for your 
experiment(s).

JT
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Re: [Asterisk-Users] broken pipe - * does not respond

[Paulo H. Mannheimer]

My release is about 2 month old. I don´t want to mess too much with my client´s 
site - it is a 100 seat call center with dozens of softphones and some 
app_queue.c patches we have programmed to register a couple of agents´ events.
  

> How old is your release.. and why not update?  I'm using the latest cvs on
> a few projects and its fine.
> 
> bkw
> 
> On Tue, 16 Dec 2003, Paulo H. Mannheimer wrote:
> 
> > Hi, I´m having a serious problem at one customer. After 6 hours answering a
> PRI
> > line, * stops responding in a very similar situation as described here ...
> >
> > http://lists.digium.com/pipermail/asterisk-users/2003-July/015391.html
> >
> > I took a look at "/proc/first * PID/fd" and there are hundreds of file
> > descriptors;
> >
> > If I try to connect using asterisk -r I get the "broken pipe"  error.
> >
> > I tried to find anything related at bugs.digium.com but couldn´t find any
> > mention to this specific situation. My CVS version is quite old, but I
> would
> > rather update it by a specific patch than to replace it entirely by a new
> one.
> >
> > Any hint would be greatly appreciated.
> >
> > Best regards,
> >
> > PauloHM
> >
> >
> >
> >
> > ___
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> >
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Re: [Asterisk-Users] IAX2 using non standard port

[Brian West]

Just use iax2 since iax2 can do everything iax can and then some.

bkw

On Tue, 16 Dec 2003, Walker Haddock wrote:

> I am working on a project where we must connect to the * server behind a firewall.  
> They can only open up port 80 udp.
>
> I have managed to get IAX working fine by specifying the port=80 in the iax.conf 
> file.  However I would like to get IAX2 working.
>
> There was a recommendation in the list 13 SEP 2003:
> http://lists.digium.com/pipermail/asterisk-users/2003-September/020560.html
>
> I tried to change the IAX_DEFAULT_PORTNO in iax2.h to 80, but then the internal 
> server also tried to talk to the external server on port 80.  Looks like 
> IAX_DEFAULT_PORTNO sets both the origin and destination port numbers.
>
> So, if I want the internal IAX2 service to listen on port 80 and I want to register 
> with the external IAX2 server using the standard port, is there another change I 
> need to make to the IAX2 code?
>
> Thanks
> --
>    DataCrest, Inc. -- Technically Superior   **
> Walker Haddock   http://www.datacrest.com
> DataCrest, Inc.e-mail:  [EMAIL PROTECTED]
> 1634A Montgomery Hwy.phone:  1-888-941-3282, 1-205-335-8589
> Birmingham, AL 35216  fax:  1-205-823-7838
> ***
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Re: [Asterisk-Users] Flash Problem

[Brian West]

Do any other phones work fine?  Or just this one?

bkw

On Tue, 16 Dec 2003, Alvaro Parres wrote:

> Hi, i have the next problem:
>
>  I have a new Motorola cordless analog phone plug at FXS ports at my
> *. But the * does not detect when i press flash?...
>
>  Any idea how can i solve this?
>
>  Is any way i can set another key to work as flash? maybe # or * ??
>
> Thanks
>
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Re: [Asterisk-Users] AT&T access code entry by Asterisk

[Brian West]

Its an upppercase W

bkw

On Tue, 16 Dec 2003, Walker Haddock wrote:

> I have a dialplan that requires that we use * to send the long distance access code 
> to AT&T.  I have found in the list that the `w` command can be used to inject a 
> pause, I have tried the following:
>
> exten => _91NX,1,Dial(ZAP/g1/${EXTEN}www555,70)
>
> There `555` is the ld access code.  I tried various quantities of `w`s but I 
> never got * to dial the ld access code.  Allof the cases I saw the `w` token 
> discussed was at the beginning of the dialed digits.  I tried putting a `w` at 
> before the ${EXTEN} and it did appear to wait a little before dialing the digits.  
> Maybe it doesn't work inside the dial string?
>
> I also tried the recommendation to use the Flash, SendDTMF, Wait, SendDTMF, but that 
> didn't work at all.
>
> I need to implement this functionality in the dialplan.  Does anyone have any 
> suggestions?
>
> Thanks for your help!
> --
>    DataCrest, Inc. -- Technically Superior   **
> Walker Haddock   http://www.datacrest.com
> DataCrest, Inc.e-mail:  [EMAIL PROTECTED]
> 1634A Montgomery Hwy.phone:  1-888-941-3282, 1-205-335-8589
> Birmingham, AL 35216  fax:  1-205-823-7838
> ***
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[Asterisk-Users] DISA - Zap/DTMF Problem

[Alexander Romanov]

Hi guys,

I am trying to use DISA. The scenario is - I call my home number (where
X100P seats) from mobile phone, enter the password, enter international
number and get connected via voiptel. It works perfectly when I call
extension setup with DISA from X-PRO SIP phone, but when I dial into
Zap, It seems that it does not detect DTMF tones. Here is a log and
config files
Please help

[[Zaptel.conf]]

fxsks=1
loadzone=au
defaultzone=au

[[Zapata.conf]]

[channels]
busydetect=yes
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
usecallerid=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
rxgain=3.0
txgain=3.0
immediate=no
context=inbound-analog
signalling=fxs_ks
callerid=asreceived
channel=>1


[[extensions.conf]]

[inbound-analog]
exten => s,1,DISA,1234|voiptel
exten => s,2,Hangup


And finaly the log ...

*CLI> -- Starting simple switch on 'Zap/1-1'
DEBUG[1255338944]: File chan_zap.c, Line 1069 (zt_enable_ec): Enabled
echo cancellation on channel 1
-- Executing DISA("Zap/1-1", "1234|voiptel") in new stack
DEBUG[1255338944]: File chan_zap.c, Line 1961 (zt_answer): Took Zap/1-1
off hook
DEBUG[1255338944]: File chan_zap.c, Line 3511 (zt_read): DTMF digit: f
on Zap/1-1
NOTICE[1255338944]: File chan_zap.c, Line 3545 (zt_read): Fax detected,
but no fax extension
DEBUG[1255338944]: File chan_zap.c, Line 3511 (zt_read): DTMF digit: f
on Zap/1-1
DEBUG[1255338944]: File chan_zap.c, Line 3549 (zt_read): Fax already
handled
DEBUG[1255338944]: File chan_zap.c, Line 3511 (zt_read): DTMF digit: f
on Zap/1-1
DEBUG[1255338944]: File chan_zap.c, Line 3549 (zt_read): Fax already
handled
DEBUG[1255338944]: File chan_zap.c, Line 3511 (zt_read): DTMF digit: f
on Zap/1-1
DEBUG[1255338944]: File chan_zap.c, Line 3549 (zt_read): Fax already
handled
DEBUG[1255338944]: File chan_zap.c, Line 3511 (zt_read): DTMF digit: f
on Zap/1-1
DEBUG[1255338944]: File chan_zap.c, Line 3549 (zt_read): Fax already
handled
DEBUG[1255338944]: File chan_zap.c, Line 3511 (zt_read): DTMF digit: f
on Zap/1-1
DEBUG[1255338944]: File chan_zap.c, Line 3549 (zt_read): Fax already
handled
DEBUG[1255338944]: File chan_zap.c, Line 3511 (zt_read): DTMF digit: f
on Zap/1-1
DEBUG[1255338944]: File chan_zap.c, Line 3549 (zt_read): Fax already
handled
DEBUG[1255338944]: File chan_zap.c, Line 3511 (zt_read): DTMF digit: f
on Zap/1-1
DEBUG[1255338944]: File chan_zap.c, Line 3549 (zt_read): Fax already
handled
DEBUG[1255338944]: File chan_zap.c, Line 3511 (zt_read): DTMF digit: f
on Zap/1-1
DEBUG[1255338944]: File chan_zap.c, Line 3549 (zt_read): Fax already
handled
DEBUG[1255338944]: File chan_zap.c, Line 3511 (zt_read): DTMF digit: f
on Zap/1-1
DEBUG[1255338944]: File chan_zap.c, Line 3549 (zt_read): Fax already
handled
DEBUG[1255338944]: File chan_zap.c, Line 3511 (zt_read): DTMF digit: f
on Zap/1-1
DEBUG[1255338944]: File chan_zap.c, Line 3549 (zt_read): Fax already
handled
DEBUG[1255338944]: File chan_zap.c, Line 3511 (zt_read): DTMF digit: f
on Zap/1-1
DEBUG[1255338944]: File chan_zap.c, Line 3549 (zt_read): Fax already
handled
DEBUG[1255338944]: File chan_zap.c, Line 3511 (zt_read): DTMF digit: f
on Zap/1-1
DEBUG[1255338944]: File chan_zap.c, Line 3549 (zt_read): Fax already
handled
DEBUG[1255338944]: File chan_zap.c, Line 3511 (zt_read): DTMF digit: f
on Zap/1-1
DEBUG[1255338944]: File chan_zap.c, Line 3549 (zt_read): Fax already
handled
DEBUG[1255338944]: File chan_zap.c, Line 3511 (zt_read): DTMF digit: f
on Zap/1-1
DEBUG[1255338944]: File chan_zap.c, Line 3549 (zt_read): Fax already
handled
DEBUG[1255338944]: File chan_zap.c, Line 3511 (zt_read): DTMF digit: f
on Zap/1-1
DEBUG[1255338944]: File chan_zap.c, Line 3549 (zt_read): Fax already
handled
DEBUG[1255338944]: File app_disa.c, Line 185 (disa_exec): DISA password
entry timeout on chan Zap/1-1
DEBUG[1255338944]: File chan_zap.c, Line 3511 (zt_read): DTMF digit: f
on Zap/1-1
DEBUG[1255338944]: File chan_zap.c, Line 3549 (zt_read): Fax already
handled


-
Alex.

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[Asterisk-Users] IAX2 using non standard port

[Walker Haddock]

I am working on a project where we must connect to the * server behind a firewall.  
They can only open up port 80 udp.

I have managed to get IAX working fine by specifying the port=80 in the iax.conf file. 
 However I would like to get IAX2 working.

There was a recommendation in the list 13 SEP 2003:
http://lists.digium.com/pipermail/asterisk-users/2003-September/020560.html

I tried to change the IAX_DEFAULT_PORTNO in iax2.h to 80, but then the internal server 
also tried to talk to the external server on port 80.  Looks like IAX_DEFAULT_PORTNO 
sets both the origin and destination port numbers.

So, if I want the internal IAX2 service to listen on port 80 and I want to register 
with the external IAX2 server using the standard port, is there another change I need 
to make to the IAX2 code?

Thanks
-- 
   DataCrest, Inc. -- Technically Superior   **
Walker Haddock   http://www.datacrest.com
DataCrest, Inc.e-mail:  [EMAIL PROTECTED]
1634A Montgomery Hwy.phone:  1-888-941-3282, 1-205-335-8589
Birmingham, AL 35216  fax:  1-205-823-7838
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[Asterisk-Users] AT&T access code entry by Asterisk

[Walker Haddock]

I have a dialplan that requires that we use * to send the long distance access code to 
AT&T.  I have found in the list that the `w` command can be used to inject a pause, I 
have tried the following:

exten => _91NX,1,Dial(ZAP/g1/${EXTEN}www555,70)

There `555` is the ld access code.  I tried various quantities of `w`s but I never 
got * to dial the ld access code.  Allof the cases I saw the `w` token discussed was 
at the beginning of the dialed digits.  I tried putting a `w` at before the ${EXTEN} 
and it did appear to wait a little before dialing the digits.  Maybe it doesn't work 
inside the dial string?

I also tried the recommendation to use the Flash, SendDTMF, Wait, SendDTMF, but that 
didn't work at all.

I need to implement this functionality in the dialplan.  Does anyone have any 
suggestions?

Thanks for your help!
-- 
   DataCrest, Inc. -- Technically Superior   **
Walker Haddock   http://www.datacrest.com
DataCrest, Inc.e-mail:  [EMAIL PROTECTED]
1634A Montgomery Hwy.phone:  1-888-941-3282, 1-205-335-8589
Birmingham, AL 35216  fax:  1-205-823-7838
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[Asterisk-Users] Flash Problem

[Alvaro Parres]

Hi, i have the next problem:

I have a new Motorola cordless analog phone plug at FXS ports at my 
*. But the * does not detect when i press flash?...

Any idea how can i solve this?

Is any way i can set another key to work as flash? maybe # or * ??

Thanks

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Re: [Asterisk-Users] broken pipe - * does not respond

[Brian West]

How old is your release.. and why not update?  I'm using the latest cvs on
a few projects and its fine.

bkw

On Tue, 16 Dec 2003, Paulo H. Mannheimer wrote:

> Hi, I´m having a serious problem at one customer. After 6 hours answering a PRI
> line, * stops responding in a very similar situation as described here ...
>
> http://lists.digium.com/pipermail/asterisk-users/2003-July/015391.html
>
> I took a look at "/proc/first * PID/fd" and there are hundreds of file
> descriptors;
>
> If I try to connect using asterisk -r I get the "broken pipe"  error.
>
> I tried to find anything related at bugs.digium.com but couldn´t find any
> mention to this specific situation. My CVS version is quite old, but I would
> rather update it by a specific patch than to replace it entirely by a new one.
>
> Any hint would be greatly appreciated.
>
> Best regards,
>
> PauloHM
>
>
>
>
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[Asterisk-Users] broken pipe - * does not respond

[Paulo H. Mannheimer]

Hi, I´m having a serious problem at one customer. After 6 hours answering a PRI 
line, * stops responding in a very similar situation as described here ...

http://lists.digium.com/pipermail/asterisk-users/2003-July/015391.html

I took a look at "/proc/first * PID/fd" and there are hundreds of file 
descriptors; 

If I try to connect using asterisk -r I get the "broken pipe"  error.

I tried to find anything related at bugs.digium.com but couldn´t find any 
mention to this specific situation. My CVS version is quite old, but I would 
rather update it by a specific patch than to replace it entirely by a new one. 

Any hint would be greatly appreciated.

Best regards,

PauloHM




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RE: [Asterisk-Users] Free Software/Open Source-Telephony-Summit 2004

[Klaus-Peter Junghanns]

Hi,

since there will be people from around the globe it will all
be done in English.

regards

kapejod

> Hi Philipp-
>
> Just out of curiosity, are these types of workshops generally conducted
> in German, or in English?
>
> Cheers
> Scott
>
> London
>
> Scott M. Stingel
> Emerging Voice Technology Inc.
>
> Email:  [EMAIL PROTECTED] 
> URL:www.evtmedia.com 
>
>
>
>> -Original Message-
>> From: [EMAIL PROTECTED]
>> [mailto:[EMAIL PROTECTED] On Behalf Of
>> Philipp von Klitzing
>> Sent: Tuesday, December 16, 2003 3:35 PM
>> To: [EMAIL PROTECTED]
>> Subject: [Asterisk-Users] Free Software/Open
>> Source-Telephony-Summit 2004
>>
>>
>> Hi,
>>
>> I just came across this annoncement, which is particularly
>> interesting as
>> it is only 25 min away from my place... :-) Anyway, I guess
>> the core of
>> this is targeted at developers mainly.
>>
>> Cheers, Philipp
>>
>>
>> Free Software/Open Source-Telephony-Summit 2004
>>
> http://www.guug.de/veranstaltungen/telephony-summit-2004/
> http://www.heise.de/newsticker/data/avr-16.12.03-000/
>
> We are happy to announce that the first summit on Free Software/Open
> Source-telephony solutions is going to take place from January 16th till
>  20th in Geilenkirchen, Germany.
>
> The event will be divided into three parts:
>
> a developer workshop from January 16th to January 18th
> a conference day (January 19th)
> a tutorial day (January 20th)
> There will be an exhibition during the conference and the tutorial day.
>
> The developer workshop is free of charge and only-open for active
> developers in Free Software/Open Source telephony projects. If you are
> interested in participating, please contact Martin Schulte telephony-
> [EMAIL PROTECTED]
>
> During the one-day conference, the participating projects will give an
> overview about their current status and their future goals.
>
> In the morning of the tutorial day, people interested in using a
> particular software will get an in depth-introduction to installation,
> configuration and usage by their developers. In the afternoon, there's a
>  big "How everything works together"-tutorial.
>
> Online-Registration is opened!
>
>
> ___
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>
>
> ___
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[Asterisk-Users] Gateway in proxy mode

[Aaron Martin]

[For Google to find for others:]
I have a Micronet SP5052 FXO gateway which I 
currently use in Peer-to-peer SIP mode with my asterisk box.
 
I have been having an intermittant problem with the 
Micronet cutting off the incoming audio stream (i.e. PSTN to VoIP) so that the 
user on the PSTN side can hear me, but I cant hear them.
 
I have talked to Micronet about this, and they have 
said it is a known bug in the SIP Peer-to-peer in their firmware and that they 
are currently working on a fix.
 
[Help for me]
They have said that in the meantime I shoudl use 
the gateway either in proxy mode or use H323.  I would prefer to stick with 
SIP, so I was wondering, what do I need to do to my asterisk box to allow the 
micronet to use it as a proxy?
 

[Asterisk-Users] John Brown from Chagres

[David Gomillion]

Some people have been airing dirty laundry on this person, so I thought
I'd air some clean laundry!

John,

Thank you so much for answering my email so quickly, and I also got your
voicemail.  And thank you for shipping on the same day I ordered.
Without these phones, there's no way I would be able to get this demo
done!

Thanks again,

David Gomillion

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Re: [Asterisk-Users] undefined symbol: ast_moh_stop

[Eric Wieling]

I was not aware that chan_sip required MOH.  This is the EXACT
modules.conf copied from a different Asterisk system running SIP and I
did not have to put res_musiconhold.so in modules.conf.

On Tue, 2003-12-16 at 15:54, Tilghman Lesher wrote:
> On Tuesday 16 December 2003 15:28, Eric Wieling wrote:
> > This is a new install so it's a fresh checkout of everything.  Does
> > anyone know what might cause this error?
> >
> >  [chan_sip.so]WARNING[16384]: File loader.c, Line 239
> > (ast_load_resource): /usr/lib/asterisk/modules/chan_sip.so:
> > undefined symbol: ast_moh_stop
> 
> You probably have noload => res_musiconhold.so or no explicit
> load of res_musiconhold.so, so that it gets loaded before anything
> else.
> 
> -Tilghman
> 
> ___
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-- 
Go to http://www.digium.com/index.php?menu=documentation and look at
the "Unofficial Links" section.  This section has links to a wide
variety of 3rd party Asterisk related pages.  My page is the
"Asterisk Resource Pages".

BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643

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RE: [Asterisk-Users] asterisk - scalable ?

[David Gomillion]

Dawid Mielnik <> wrote:
 
> How scalable is asterisk ?
> 

This has been hashed a number of times.  Check out
http://www.google.com/search?hl=en&lr=&ie=UTF-8&oe=UTF-8&q=+site:lists.d
igium.com+asterisk+scale

There are literally hundreds of hits, and people have approached the
topic from a number of ways.  One of them are sure to be the angle
you're looking for!

> Asterisk offers conferencing (meetme). The conference rooms as such
> dont have any user verification, setting up passwords etc. Has anyone
> implemented anything like that ?  
> 

Asterisk has an authenticate method, which you could use just before you
put the person into the meetme room (in extensions.conf).  I'm sure
there are better ways, but just rest assured, yes, there is a way to
password protect a conference room.

> regards,
> 
> Dave

Hope this helps,
David Gomillion

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Re: [Asterisk-Users] undefined symbol: ast_moh_stop

[Tilghman Lesher]

On Tuesday 16 December 2003 15:28, Eric Wieling wrote:
> This is a new install so it's a fresh checkout of everything.  Does
> anyone know what might cause this error?
>
>  [chan_sip.so]WARNING[16384]: File loader.c, Line 239
> (ast_load_resource): /usr/lib/asterisk/modules/chan_sip.so:
> undefined symbol: ast_moh_stop

You probably have noload => res_musiconhold.so or no explicit
load of res_musiconhold.so, so that it gets loaded before anything
else.

-Tilghman

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Re: [Asterisk-Users] Stupid Newbie Questions

[TC]

You can try this hacking this patch
http://bugs.digium.com/bug_view_page.php?bug_id=504
- Original Message - 
From: "TkVoice Author" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, December 16, 2003 11:34 AM
Subject: Re: [Asterisk-Users] Stupid Newbie Questions


> 
> Okay I don't have ISDN but I want to still be able to receive FAX's and 
> run a PPPD dial-in server.
> 
> I do have distinctive ring on my main line.
> 
> Can I configure Asterisks to ignore certain distinctive ring patterns 
> (i.e. don't pick up 3-short ) so that my modems controlled by mgetty can 
> pick up the FAX's & Data-Dial-In Calls?
> 
> Thanks
> 
> -- 
> Peter
> http://tkvoice.netfirms.com/
> 
> 
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[Asterisk-Users] asterisk - scalable ?

[Dawid Mielnik]

Hi all,

How scalable is asterisk ?

I am considering using asterisk as a VoIP platform/gateway between Internet
and PSTN (switches) to offer services to home customers. What goes along
with it is eventually a lot of users - upto thousands probably. Is load
balancing possible with multiple asterisk boxes ? Does anyone have any sort
of info/experience with such projects ? How would asterisk cope with such
load.

Asterisk offers conferencing (meetme). The conference rooms as such dont
have any user verification, setting up passwords etc. Has anyone implemented
anything like that ?

regards,

Dave

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[Asterisk-Users] undefined symbol: ast_moh_stop

[Eric Wieling]

This is a new install so it's a fresh checkout of everything.  Does
anyone know what might cause this error?

 [chan_sip.so]WARNING[16384]: File loader.c, Line 239
(ast_load_resource): /usr/lib/asterisk/modules/chan_sip.so: undefined
symbol: ast_moh_stop

-- 
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[Asterisk-Users] command Authenticate

[SW]

Hi folks,

Sorry to post bunch of messages today, I am deep into this and hope any help
from the experts.

I am using the command Authenticate as explained in wi-ki:


===
Authenticate(password[|options])

Requires a user to enter a given password in order to continue execution. If
the password begins with the '/' character, it is interpreted as a file
which contains a list of valid passwords (1 per line).

An optional set of options may be provided by concatenating any of the
following letters:


a - Set account code to the password that is entered

=

exact statement in my extensions.conf is

exten => s,1,Authenticate(1234|abc)

So I get the voice prompt for the password, I send the digits, *
authenticate, and call goes through => all are happy.

Then I go to the CDR and I find the accountcode is logged as 1234. I would
expect it to be abc ?

If I do this, exten => s,1,Authenticate(1234|10) nothing get logged?

Is this the way it supposed to work, or am I doing something wrong here ?

Cheers

SW


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Re: [Asterisk-Users] Cisco AT-18x SIP 3.0 Firmware

[Juan J. Sierralta P.]

On Tue, 2003-12-16 at 16:15, Brian West wrote:
> http://www.cisco.com/univercd/cc/td/doc/product/voice/ata/atarn/atarn3_0.htm

One of the nicest features is that the programs to upgrade and generate
TFTP confs are available for Linux and Solaris also.

-- 
Juanjo sin .sig

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Re: [Asterisk-Users] DIAX-SJPHONE REGISTRATION PROBLEM

[Softprofit Solutions]

From: "Softprofit Solutions" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] DIAX-SJPHONE REGISTRATION PROBLEM
Date: Tuesday, December 16, 2003 2:34 PM

Sorry - yes I was able to get DIAX to register, but not sjphone

added,  nat=yes to sip.conf, no luck

tried to drop secret but no luck...


- Original Message - 
From: "Dan" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, December 16, 2003 2:26 PM
Subject: Re: [Asterisk-Users] DIAX-SJPHONE REGISTRATION PROBLEM


> 
> >- Original Message - 
> >From: "Softprofit Solutions" <[EMAIL PROTECTED]>
> > Subject: Re: [Asterisk-Users] DIAX-SJPHONE REGISTRATION PROBLEM
> 
> > Thanks for making me think, this is what worked
> 
> Not clear for me it works now or not?
> 
> Best regards,
> Dan
> ___
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- Original Message - 
From: "Dan" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, December 16, 2003 2:26 PM
Subject: Re: [Asterisk-Users] DIAX-SJPHONE REGISTRATION PROBLEM


> 
> >- Original Message - 
> >From: "Softprofit Solutions" <[EMAIL PROTECTED]>
> > Subject: Re: [Asterisk-Users] DIAX-SJPHONE REGISTRATION PROBLEM
> 
> > Thanks for making me think, this is what worked
> 
> Not clear for me it works now or not?
> 
> Best regards,
> Dan
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Re: [Asterisk-Users] DIAX-SJPHONE REGISTRATION PROBLEM

[Andrew Thompson]

- Original Message -
From: "Softprofit Solutions" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, December 16, 2003 3:11 PM
Subject: Re: [Asterisk-Users] DIAX-SJPHONE REGISTRATION PROBLEM


> Thanks for making me think, this is what worked
>
>
> iax.conf
>
> [general]
> port=5036
> bandwidth=low
> disalow=lpc10

I believe that should be: disallow=lpc10

> [user]
> type=friend
> username=user
> secret=user
> host=dynamic
> context=extensions
> context=default
>

-
Andrew Thompson http://aktzero.com/
Your eyes are weary from staring at the CRT. You feel sleepy. Notice how
restful it is to watch the cursor blink. Close your eyes. The opinions
stated above are yours. You cannot imagine why you ever felt otherwise.



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Re: [Asterisk-Users] DIAX-SJPHONE REGISTRATION PROBLEM

[Dan]

>- Original Message - 
>From: "Softprofit Solutions" <[EMAIL PROTECTED]>
> Subject: Re: [Asterisk-Users] DIAX-SJPHONE REGISTRATION PROBLEM

> Thanks for making me think, this is what worked

Not clear for me it works now or not?

Best regards,
Dan
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Re: [Asterisk-Users] DIAX-SJPHONE REGISTRATION PROBLEM

[Softprofit Solutions]

Thanks for making me think, this is what worked


iax.conf

[general]
port=5036
bandwidth=low
disalow=lpc10

[user]
type=friend
username=user
secret=user
host=dynamic
context=extensions
context=default




- Original Message - 
From: "Dan" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, December 16, 2003 1:08 PM
Subject: Re: [Asterisk-Users] DIAX-SJPHONE REGISTRATION PROBLEM


> Hi,
> >- Original Message - 
> >From: Softprofit Solutions
> >Subject: [Asterisk-Users] DIAX-SJPHONE REGISTRATION PROBLEM
>
>
> > I am having a problem with softphone registration, having read the list
> and watched it for a while for similar problems I just
> > cant seem to figure out the problem.  Using SJPHONE or DIAX , I can make
> outgoing calls but I can't get them to register
> > with asterisk, I have other sip devices registering OK-7940's.  From the
> list and the digium web site this seems to be a
> > straight forward set up , I must be missing something, help would be
> appreciated.
>
> Please send the user and general sections in iax.conf to see where is the
> problem.
>
> Best regards,
> Dan
>
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Re: [Asterisk-Users] codec negotiation

[Andrew Thompson]

- Original Message -
From: "Eduardo Goncalves" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, December 16, 2003 1:08 PM
Subject: [Asterisk-Users] codec negotiation


> Hi list,
>
> I'm with a little problem on codec negotiation between a cisco827 and
> asterisk.
>
> My sip.conf is like that:
>
> [general]
> port = 5060
> bindaddr = 0.0.0.0
> context = default
> amaflags = default
> allow=g729
> allow=gsm
> allow=alaw
> allow=ulaw
> ;disallow=all
>
> and cisco like that:
>
> dial-peer voice 6 voip
>  destination-pattern 0T
>  session protocol sipv2
>  session target ipv4:
>  dtmf-relay rtp-nte
>  codec g711alaw
>  no vad
> !
>
> When I try to make a call, cisco shows codec g711alaw, but asterisk
> shows codec g729A (i have the licenses) and there is no audio. When I
> try disallow=g729, the same occurs, but this time asterisk shows codec
> gsm.
>
> The only way to make a call is allowing only alaw. But this is not
> convenience, since i need to use g279 with another endpoint (working
> ok).
>


You could try setting the codec before dialing that particular provider.
Except I don't see the command now that I'm trying to find it...

> Why this negotiation problem happens?

Can't help on that one...

>
> Thanks
> Eduardo

-
Andrew Thompson http://aktzero.com/
Your eyes are weary from staring at the CRT. You feel sleepy. Notice how
restful it is to watch the cursor blink. Close your eyes. The opinions
stated above are yours. You cannot imagine why you ever felt otherwise.



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Re: [Asterisk-Users] sip registration send out by asterisk

[Andrew Thompson]

- Original Message -
From: "SW" <[EMAIL PROTECTED]>
To: "[EMAIL PROTECTED] Digium. Com" <[EMAIL PROTECTED]>
Sent: Tuesday, December 16, 2003 1:47 PM
Subject: [Asterisk-Users] sip registration send out by asterisk


> Hi friends,
>
> I've noticed that first register message sent by * always get rejected by
> the destination sip server. Then * sends a second registration message (
> with Autherization section, and that get accepted by the destination
host).
>
> Why is this ?
>
> Isnt there a way to tell * to send with Autothorization message the first
> attempt ?
>
>
> Asterisk sends this first
>
> 9 headers, 0 lines
> 11 headers, 0 lines
> Reliably Transmitting:
> REGISTER sip:sipauth.deltathree.com SIP/2.0
> Via: SIP/2.0/UDP 66.xxx.xxx.xxx:5060;branch=z9hG4bK6b37ba4a
> From: ;tag=as3e96887d
> To: 
> Call-ID: [EMAIL PROTECTED]
> CSeq: 106 REGISTER
> User-Agent: Asterisk PBX
> Expires: 160
> Contact: 
> Event: registration
> Content-length: 0
>
>
> After 401 unautorized from iconnect asterisk sends this
>
> 8 headers, 0 lines
> 12 headers, 0 lines
> Reliably Transmitting:
> REGISTER sip:sipauth.deltathree.com SIP/2.0
> Via: SIP/2.0/UDP 66.xxx.xxx.xxx:5060;branch=z9hG4bK6b37ba4a
> From: ;tag=as3e96887d
> To: 
> Call-ID: [EMAIL PROTECTED]
> CSeq: 107 REGISTER
> User-Agent: Asterisk PBX
> Authorization: Digest username="1510xxx", realm="deltathree.com",
> algorithm="MD5", uri="sip:sipauth.deltathree.com", nonce="3fdecbbf",
> response="49558c95bc3383bcbf76a26376e1614a"
> Expires: 160
> Contact: 
> Event: registration
> Content-length: 0
>
> Cheers
>
> SW
>


I could be wrong, but I believe there is a challenge token sent back with
the Unauthorized message that is used to build the properly Authenticated
request.

-
Andrew Thompson http://aktzero.com/
Your eyes are weary from staring at the CRT. You feel sleepy. Notice how
restful it is to watch the cursor blink. Close your eyes. The opinions
stated above are yours. You cannot imagine why you ever felt otherwise.



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Re: [Asterisk-Users] Nagios/measurement with Asterisk - any plugins?

[Richard Lyman]

John Todd wrote:
  I have spent some time digging through the archives for comments 
concerning Asterisk and monitoring systems, and I have found few results.

check_asterisk.pl.gz (http://www.dynx.net/ASTERISK/misc-progs/) which 
gives an error on download, and has no further Google references
i made a copy of it called 
http://www.dynx.net/ASTERISK/misc-progs/chk_ast_thisisa_gz.txt
just save it and rename to check_asterisk.pl.gz


astping.tar (http://www.dynx.net/ASTERISK/misc-progs/ and also in the 
mailing list archives) supposedly sends a query to an Asterisk server, 
but I have been unable to get it to do anything other than reply with 
the IP address of the queried host (regardless of Asterisk status)

*snipped

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[Asterisk-Users] iconnect incoming works, so what ...

[SW]

Hi,

I would like to ask anyone, who use Iconnect incoming calls, receiving via a
context other than [general] to respond to this message. I challenge that
there is no way that we can do with current * code or the way SIP messages
are received from iconnect. No matter what you do, iconnect calls always
land in the general section. Best source of info out there, John Todds
example, sip.conf is also configured to receive iconnect calls through
general section. When I post here people would advise (appreciate), but I
would like to hear from any one who has their * sip.conf configured so that
iconnect calls find their context ?

Just to summarize, from Iconnect I get three parameters

1. Username (52671573)
2. Password (9876)
3. phone number (1411212)

so in sip.conf, we have the register string;

register=1411212:[EMAIL PROTECTED]/1411212


and the context for the peer/friend

(a)
[iconnect]
type=friend
secret=9876
username=52671573
host=sipauth.deltathree.com
dtmfmode=inband
context = iconnect-in

(b)
[sipauth.deltathree.com]
type=friend
secret=9876
username=52671573
host=sipauth.deltathree.com
dtmfmode=inband
context = iconnect-in

(a) or (b) above will function as the incoming context for iconnect call.

As a side note, method (b) works for incoming calls from fwd.

Cheers

SW


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Re: [Asterisk-Users] Stupid Newbie Questions

[TkVoice Author]

Okay I don't have ISDN but I want to still be able to receive FAX's and 
run a PPPD dial-in server.

I do have distinctive ring on my main line.

Can I configure Asterisks to ignore certain distinctive ring patterns 
(i.e. don't pick up 3-short ) so that my modems controlled by mgetty can 
pick up the FAX's & Data-Dial-In Calls?

Thanks

--
Peter
http://tkvoice.netfirms.com/
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[Asterisk-Users] Asterisk Support for QSIG

[Ahmed Tariq]

Hi,

The last post on this topic seems to be
http://lists.digium.com/pipermail/asterisk-users/2003-June/013324.html

EuroISDN seems to be supported by Asterisk
http://www.asterisk.org/index.php?menu=features

Regarding Asterisk support for QSIG, could someone please supply answers to
the following.

- What is the ESTI ISDN spec., for basic call control, that Asterisk
confirms to (ETS 300 102-1, EN 300 403-1, or some other).
- QSIG, basic call control, is specified in ECMA 143. Is there any
difference between the ESTI and ECMA messages and procedures as far as basic
call control is concerned?


Best Regards,
Ahmed N Tariq
RADIOFRAME NETWORKS
www.radioframenetworks.com


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[Asterisk-Users] SUBSCRIBE in channel_sip.c

[Olle E. Johansson]

Anyone that knows what support we have for SUBSCRIBE?
There's some code in chan_sip.c that do something. And there's a comment on SNOM 200.
Curious.

/Olle

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[Asterisk-Users] Cisco AT-18x SIP 3.0 Firmware

[Brian West]

http://www.cisco.com/univercd/cc/td/doc/product/voice/ata/atarn/atarn3_0.htm

LOVE IT.. call transfers work better but not totally as expected yet.  I'm
still tweaking the configs.  Also bitaid will help alot.

bkw
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Re: [Asterisk-Users] DIAX-SJPHONE REGISTRATION PROBLEM

[Dan]

Hi,
>- Original Message - 
>From: Softprofit Solutions
>Subject: [Asterisk-Users] DIAX-SJPHONE REGISTRATION PROBLEM


> I am having a problem with softphone registration, having read the list
and watched it for a while for similar problems I just
> cant seem to figure out the problem.  Using SJPHONE or DIAX , I can make
outgoing calls but I can't get them to register
> with asterisk, I have other sip devices registering OK-7940's.  From the
list and the digium web site this seems to be a
> straight forward set up , I must be missing something, help would be
appreciated.

Please send the user and general sections in iax.conf to see where is the
problem.

Best regards,
Dan

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[Asterisk-Users] asterisk and cisco call manager via h.323

[Pavel Zheltouhov]

Does asterisk work with CCM as gateway ?
When I trying call asterisk,I totally can't hear any sound.
When call ohphone - works  good.
10.0.1.219 is CCM, 10.0.1.207  asterisk.
Trace messages here :

	== New H.323 Connection created.
	-- Received SETUP message...
	== Setting up Call
	   -- Calling party name:  [5001,]
	   -- Calling party number:  [5001]
	   -- Called  party name:  [500]
	   -- Called  party number:  [500]
-- Executing Playback("H323/ip$10.0.1.219:2303/8", 
"demo-abouttotry") in new stack
-- Playing 'demo-abouttotry' (language 'en')
	=*= In CreateRealTimeLogicalChannel for call 8
		-- externalIpAddress: 10.0.1.207
		-- externalPort: 15210
		-- SessionID: 1
		-- Direction: IsTransmitter
	 -- Started logical channel: sending G.711-uLaw-64k{sw}
		-- channelsOpen = 1
		-- remoteIpAddress: 0.0.0.0
		-- remotePort: 0
		-- ExternalIpAddress: 10.0.1.207
		-- ExternalPort: 15210
	-- Connection Established with "5001, 5001 [10.0.1.219]"
	=*= In CreateRealTimeLogicalChannel for call 8
		-- externalIpAddress: 10.0.1.207
		-- externalPort: 15210
		-- SessionID: 1
		-- Direction: IsReceiver
	 -- Started logical channel: receiving G.711-uLaw-64k{sw}
		-- channelsOpen = 2
		-- remoteIpAddress: 10.0.1.219
		-- remotePort: 4000
		-- ExternalIpAddress: 10.0.1.207
		-- ExternalPort: 15210

--

> remoteIpAddress: 0.0.0.0
> remotePort: 0
Looks incorrectly !

Tested with latest cvs asterisk.
Maybe asterisk h.323 channel driver not correctly parse h.323 messages.
--
Pavel Zheltouhov
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[Asterisk-Users] DIAX-SJPHONE REGISTRATION PROBLEM

[Softprofit Solutions]

I am having a problem with softphone registration, 
having read the list and watched it for a while for similar problems I just cant 
seem to figure out the problem.  Using SJPHONE or DIAX , I can make 
outgoing calls but I can't get them to register with asterisk, I have other sip 
devices registering OK-7940's.  From the list and the digium web site this 
seems to be a straight forward set up , I must be missing something, help would 
be appreciated.
 
sjphone - registration denied
diax - registration rejected
 
Thanks.
 
Rob.
 
sip.conf..
 
[name]type=friendusername=name
secret=name
context=extensionscontext=defaulthost=dynamicqualify=2000disallow=g729mailbox=70666

[Asterisk-Users] DIAX not receiving calls

[Hector Q.-datafull]

I'm working with DIA096B on two remote computers that are behind NAT. They register ok.
The * has a static public IP address.
I saw other simliar posts btu this seems to be different.

The call is from test2 --> test3:

  -- Accepting AUTHENTICATED call from xx.xx.xx.xx , requested format = 2, actual 
format = 2
-- Executing Dial("[EMAIL PROTECTED]/3", "IAX/test2") in new stack
NOTICE[1200884528]: File app_dial.c, Line 506 (dial_exec): Unable to create channel of 
type 'IAX'
  == Everyone is busy at this time
-- Executing VoiceMail("[EMAIL PROTECTED]/3", "u2002") in new stack
xx.xx.xx.xx is the public ip of the NAT in front of DIAX096B (test2)

[general] 
port=5036
bindaddr=publicipaddress
disallow=all
allow=gsm
jitterbuffer=3
tos=reliability

[test2]
type=friend
username=test2
secret=
host=dynamic
context=test

[tito3]
type=friend
username=test3
secret=..
host=dynamic
context=test


Thanks.
HQ.

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[Asterisk-Users] sip registration send out by asterisk

[SW]

Hi friends,

I've noticed that first register message sent by * always get rejected by
the destination sip server. Then * sends a second registration message (
with Autherization section, and that get accepted by the destination host).

Why is this ?

Isnt there a way to tell * to send with Autothorization message the first
attempt ?


Asterisk sends this first

9 headers, 0 lines
11 headers, 0 lines
Reliably Transmitting:
REGISTER sip:sipauth.deltathree.com SIP/2.0
Via: SIP/2.0/UDP 66.xxx.xxx.xxx:5060;branch=z9hG4bK6b37ba4a
From: ;tag=as3e96887d
To: 
Call-ID: [EMAIL PROTECTED]
CSeq: 106 REGISTER
User-Agent: Asterisk PBX
Expires: 160
Contact: 
Event: registration
Content-length: 0


After 401 unautorized from iconnect asterisk sends this

8 headers, 0 lines
12 headers, 0 lines
Reliably Transmitting:
REGISTER sip:sipauth.deltathree.com SIP/2.0
Via: SIP/2.0/UDP 66.xxx.xxx.xxx:5060;branch=z9hG4bK6b37ba4a
From: ;tag=as3e96887d
To: 
Call-ID: [EMAIL PROTECTED]
CSeq: 107 REGISTER
User-Agent: Asterisk PBX
Authorization: Digest username="1510xxx", realm="deltathree.com",
algorithm="MD5", uri="sip:sipauth.deltathree.com", nonce="3fdecbbf",
response="49558c95bc3383bcbf76a26376e1614a"
Expires: 160
Contact: 
Event: registration
Content-length: 0

Cheers

SW


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RE: [Asterisk-Users] iaxclients missing calls

[firedude]

Speaking for me personally, I have experienced the problem when not using 
NAT.  The particular incidents I referenced dealt with an asterisk server 
and client, both being on the 192.168.22 network.
AJ


On Tue, 16 Dec 2003, Patrick Cantwell wrote:

> Just a thought:  Out of everyone having problems, how many are connecting
> through NAT to their asterisk server? (client -> nat -> ... -> asterisk).
> 
> NAT firewalls time ports out after a certain amount of inactivity.  I was
> having the same problem with my SIP analog adapter.  I could fire it up and
> receive calls, but after a little while, it could no longer receive calls
> (yet asterisk would show them coming in).  If I made a call out from the
> analog phone, I could again receive calls for a short while.  Then it
> occurred to me that I probably didn't change my firewall rules after
> renumbering my asterisk box and it was a NAT issue.
> 
> If anybody having this problem is running behind some sort of NAT device,
> try making your registration time rediculously low, such as 60s and see if
> it has an effect. (or try changing your rules, if possible)
> 
> Just my .02 =)
> 
> -Pat
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Michael Van
> Donselaar
> Sent: Monday, December 15, 2003 11:25 PM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] iaxclients missing calls
> 
> 
> On Mon, 15 Dec 2003 23:05:56 -0500 (EST), [EMAIL PROTECTED]
> wrote:
> 
> >Hello All
> >When I open up iaxcomm, it registers fine with the asterisk server.  If I
> >call into it, iaxcomm will ring; however if I leave iaxcomm sitting idle
> >for awhile (I haven't figured out exactly how long) it seems to miss
> >calls.  I can see the calls coming in on the asterisk server but they
> >never ring through on iaxcomm.  If I close it and reopen it, it takes
> >calls again fine.  I thought I saw someone else who had experienced this
> >problem too but I don't recall seeing a solution.  Any thoughts or
> >solutions?  I'm trying to use iaxcomm in my every daybusiness but it's kind
> of
> >hard when I miss calls.  It complicates things a bit.  Although I used
> >iaxcomm as an example here I've experienced the same problem with DIAX.
> >
> >Thanks a bunch.
> >AJ
> 
> Hi, AJ.
> 
> It sounds like this is related to the other incoming ring issue.  I'll check
> through the archives to verify, but I think a couple of other users were
> seeing
> this even soon after launching iaxComm or DIAX.
> 
> I'm wondering if the iaxclient lib might be having a problem with
> registration
> renewal, or if it's a problem with event dispatching.
> 
> Any other users seen this yet? (That is, seen incoming ring failures become
> more
> likely as uptime increases).
> 
> The next time you get an ignored incoming call, could you try doing an iax2
> debug on the cli?
> 
> Do you see the same problem with DIAX when using IAX instead of IAX2?
> 
> 
> >
> >
> >___
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> 
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[Asterisk-Users] codec negotiation

[Eduardo Goncalves]

Hi list,

I'm with a little problem on codec negotiation between a cisco827 and
asterisk.

My sip.conf is like that: 

[general]
port = 5060
bindaddr = 0.0.0.0
context = default
amaflags = default
allow=g729
allow=gsm 
allow=alaw
allow=ulaw
;disallow=all

and cisco like that:

dial-peer voice 6 voip
 destination-pattern 0T
 session protocol sipv2
 session target ipv4:
 dtmf-relay rtp-nte
 codec g711alaw
 no vad   
! 

When I try to make a call, cisco shows codec g711alaw, but asterisk
shows codec g729A (i have the licenses) and there is no audio. When I
try disallow=g729, the same occurs, but this time asterisk shows codec
gsm.

The only way to make a call is allowing only alaw. But this is not
convenience, since i need to use g279 with another endpoint (working
ok). 

Why this negotiation problem happens?

Thanks
Eduardo
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[Asterisk-Users] Requesting advice from experienced * users/developers

[Steve Dolloff]

Greetings,

I have a couple of questions and figured I would put them all in one
message to not spam the list as much as possible.  I have searched
voip-info, google and the list archives for all of these questions.  If
I have missed the correct response, please accept my apologies.

I have been stuck on these for a long time and I am really hoping that
the other users out there will be able to help me out.

1)  VM attendant sounds scratchy?

I am using the G711ulaw codec via SIP.  The messages themselves sound
fine, recorded in WAV49 format (chosen for least hd space).  This is
especially true when reading the numbers from a mailbox or the number of
messages.  I would guess that it's due to the attendant messages being
recorded in gsm, but I have recorded a few messages using the Record App
and they sound fine too.  Is this something to do with the VM app?

2)  Privacy manager/Zapateller not working correctly.

I have posted the details of this problem in a previous post, but
basically, they don't seem to recognize anonymous callers vs callers
with no caller-id.

3)  Call-waiting caller-id doesn't work. (but either by themselves do) 

I am not using zapta interfaces, but I have enabled the appropriate
setting in that conf file in case it has some relevance.  I didn't see a
similar option in the sip.conf file.  I have tested with both the ATA186
and the SPA-2000.

4)  I am looking for a provider of caller-id name database services on a
U.S. national basis for incoming calls using an AGI script.


Willing and able to post all relevant config files.  Thanks for any help
you can provide.

Stephen
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Re: [Asterisk-Users] Stupid Newbie Questions

[Steven Critchfield]

On Tue, 2003-12-16 at 10:59, TkVoice Author wrote:
> I just learned about Asterisk yesterday.
> 
> Besides all the cool PBX stuff that asterisk can do can it also do these:

This is where you should be looking at any and all documentation too. 

> 1) Receive FAX's ?

Yes, very alpha at the moment judging by other peoples experiences.
Sending seems less problematic.

> 2) Control (serve) pppd dial-in connections ?

Only if it is on an ISDN B channel, no modems.

> 3) Is it possible to use voice modems as the FXO/FXS cards ?

ABSOLUTELY NOT. First the modem is only a FXO device and it does a poor
job at bidirectional voice. It is so bad that we do not support it here.
If you desire this functionality, you will spend 4 or more times the
difference in cost in of a real FXO device in the time it will take to
get a voice modem to function half way. Then you will be upset at the
results and time wasted. Save yourself the grief and us the next flame
way by not going this route.

> Thank You
-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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[Asterisk-Users] Cisco 7960 Firmware issues

[Lists]

I have an older 7960 and I have having this issue:

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_field_notice09186a008009451b.shtml

Can anyone with more experiance help me?

I do not have a CCO account.

Michael

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RE: [Asterisk-Users] IP 500/600 1.1.0 Firmware

[Greg Hill]

On Tue, 16 Dec 2003, mattf wrote:
> I've created a page on Wiki for the Polycom phones:
> http://www.voip-info.org/wiki-Polycom+Phones
>
> I don't have a problem with putting the FAQ and sample config for these
> on the voip-info site except for the fact that voip-info.org is really
> slow, does anyone else notice that? What kind of connection is it on?

I've noticed it slow sometimes.. might it be because of the processor load
of doing everything in php, rather than a limit on the internet pipe
coming out of that machine?

Greg


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RE: [Asterisk-Users] IP 500/600 1.1.0 Firmware

[mattf]

Hello,

Here's part of the email I sent to Scott at Polycom:

"We will also need a technical contact within Polycom that we can bring up
bugs and other issues with. You can limit this person's accessability to
only a few people if you like. We need this to deal with real issues with
the phone such as inconsistencies in the documentation and bugs with the
phone. John has already run into a few just in the last week."

I'll stress that when I talk to him today.

The bug process you stated is exactly how I want it to go as also.

MATT---


-Original Message-
From: John Baker [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 16, 2003 12:09 PM
To: [EMAIL PROTECTED]
Cc: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] IP 500/600 1.1.0 Firmware


We really need to push for an assigned tech from Polycom to talk to
asterisk-polycom-developers about problems with the SIP software and other
issues.  Any questions not covered by the wiki could be submitted to one of
us and if we can't answer it, then ask Polycom

John Baker
List Brokers, Inc.

- Original Message - 
From: "mattf" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, December 16, 2003 10:41 AM
Subject: RE: [Asterisk-Users] IP 500/600 1.1.0 Firmware


> Hello,
>
> I've created a page on Wiki for the Polycom phones:
> http://www.voip-info.org/wiki-Polycom+Phones
>
> I don't have a problem with putting the FAQ and sample config for these on
> the voip-info site except for the fact that voip-info.org is really slow,
> does anyone else notice that? What kind of connection is it on?
>
> I will still have the downloads available on a faster server and will keep
> copies of the FAQ and sample configs there when we have them finished.
>
> I am waiting to hear back from Scott from Polycom about the final
agreement,
> but what it looks like is that the Asterisk community will have to support
> their phones on Asterisk by itself. This means that a few people will have
> access to the full Polycom developer system and we will post the new
> firmware releases on a website as they are made available. Also any
> non-defective-product support queries made to Polycom about integrating
with
> Asterisk will be directed by Polycom to the Asterisk-Polycom support
> website(when it's up and running).
>
> MATT---
>
>
>
> -Original Message-
> From: Miguel Cavazos [mailto:[EMAIL PROTECTED]
> Sent: Monday, December 15, 2003 12:14 PM
> To: '[EMAIL PROTECTED]'
> Subject: RE: [Asterisk-Users] IP 500/600 1.1.0 Firmware
>
>
> good to hear theres going to be support for this phones, but why not put
> it on the wiki??? so we can have all the faq in one place.
>
> Miguel
> On Mon, 2003-12-15 at 22:23, mattf wrote:
> > I just got off of the phone with Scott Willard at Polycom and things
seem
> > promising. He's going to send me the latest stable boot firmware for the
> > Soundpoint 500/600 phones and I will make that available on a webiste
> > somewhere for people to download.
> >
> > It seems like their big stumbling block to giving us the firmware and
> admin
> > docs for their phones is that they don't want to have to support end
users
> > of their phones. They usually have their resellers do support for
specific
> > PBXs that their phones work with and because none of their resellers
> support
> > Asterisk or sell Asterisk that means no one supports Asterisk users of
> their
> > phones.
> >
> > I told him that we would be putting up a support website for their
phones
> > that they can point Asterisk end users to, to download the latest
firmware
> > and the admin docs as well as contact people for help with
configuration.
> He
> > seemed to accept that idea and is going to send me a copy of the latest
> > bootloader as a sign of goodwill for stalling for so long.
> >
> > I am to present him with a document outlining what kind of support we
will
> > provide and how they can point end users that call polycom to us so that
> we
> > can help. I plan on putting up a simple website with a few FAQs and
sample
> > configs as well as some people that can be emailed or called to ask
> > questions about Polycom on Asterisk. I'd love to have a few volunteers
for
> > that list so I don't get swamped :)
> >
> > I plan on hosting the website(donated by my company of course) at least
in
> > the near term just to get something up.
> >
> > Let me know if you're interested in helping out. I'll post the new
> > bootloader on a web site when I get it.
> >
> > Thanks,
> >
> > MATT---
> >
> >
> >
> > -Original Message-
> > From: mattf [mailto:[EMAIL PROTECTED]
> > Sent: Monday, December 15, 2003 4:01 PM
> > To: '[EMAIL PROTECTED]'
> > Subject: RE: [Asterisk-Users] IP 500/600 1.1.0 Firmware
> >
> >
> > Hello,
> >
> > I've talked with various people at Polycom about getting on their
> developer
> > program and they have been stalling me for weeks. The most information I
> > received was from Scott Willard, head of IP phone sales for Polycom. He
> said
> > that the company is

Re: [Asterisk-Users] IP 500/600 1.1.0 Firmware

[John Baker]

We really need to push for an assigned tech from Polycom to talk to
asterisk-polycom-developers about problems with the SIP software and other
issues.  Any questions not covered by the wiki could be submitted to one of
us and if we can't answer it, then ask Polycom

John Baker
List Brokers, Inc.

- Original Message - 
From: "mattf" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, December 16, 2003 10:41 AM
Subject: RE: [Asterisk-Users] IP 500/600 1.1.0 Firmware


> Hello,
>
> I've created a page on Wiki for the Polycom phones:
> http://www.voip-info.org/wiki-Polycom+Phones
>
> I don't have a problem with putting the FAQ and sample config for these on
> the voip-info site except for the fact that voip-info.org is really slow,
> does anyone else notice that? What kind of connection is it on?
>
> I will still have the downloads available on a faster server and will keep
> copies of the FAQ and sample configs there when we have them finished.
>
> I am waiting to hear back from Scott from Polycom about the final
agreement,
> but what it looks like is that the Asterisk community will have to support
> their phones on Asterisk by itself. This means that a few people will have
> access to the full Polycom developer system and we will post the new
> firmware releases on a website as they are made available. Also any
> non-defective-product support queries made to Polycom about integrating
with
> Asterisk will be directed by Polycom to the Asterisk-Polycom support
> website(when it's up and running).
>
> MATT---
>
>
>
> -Original Message-
> From: Miguel Cavazos [mailto:[EMAIL PROTECTED]
> Sent: Monday, December 15, 2003 12:14 PM
> To: '[EMAIL PROTECTED]'
> Subject: RE: [Asterisk-Users] IP 500/600 1.1.0 Firmware
>
>
> good to hear theres going to be support for this phones, but why not put
> it on the wiki??? so we can have all the faq in one place.
>
> Miguel
> On Mon, 2003-12-15 at 22:23, mattf wrote:
> > I just got off of the phone with Scott Willard at Polycom and things
seem
> > promising. He's going to send me the latest stable boot firmware for the
> > Soundpoint 500/600 phones and I will make that available on a webiste
> > somewhere for people to download.
> >
> > It seems like their big stumbling block to giving us the firmware and
> admin
> > docs for their phones is that they don't want to have to support end
users
> > of their phones. They usually have their resellers do support for
specific
> > PBXs that their phones work with and because none of their resellers
> support
> > Asterisk or sell Asterisk that means no one supports Asterisk users of
> their
> > phones.
> >
> > I told him that we would be putting up a support website for their
phones
> > that they can point Asterisk end users to, to download the latest
firmware
> > and the admin docs as well as contact people for help with
configuration.
> He
> > seemed to accept that idea and is going to send me a copy of the latest
> > bootloader as a sign of goodwill for stalling for so long.
> >
> > I am to present him with a document outlining what kind of support we
will
> > provide and how they can point end users that call polycom to us so that
> we
> > can help. I plan on putting up a simple website with a few FAQs and
sample
> > configs as well as some people that can be emailed or called to ask
> > questions about Polycom on Asterisk. I'd love to have a few volunteers
for
> > that list so I don't get swamped :)
> >
> > I plan on hosting the website(donated by my company of course) at least
in
> > the near term just to get something up.
> >
> > Let me know if you're interested in helping out. I'll post the new
> > bootloader on a web site when I get it.
> >
> > Thanks,
> >
> > MATT---
> >
> >
> >
> > -Original Message-
> > From: mattf [mailto:[EMAIL PROTECTED]
> > Sent: Monday, December 15, 2003 4:01 PM
> > To: '[EMAIL PROTECTED]'
> > Subject: RE: [Asterisk-Users] IP 500/600 1.1.0 Firmware
> >
> >
> > Hello,
> >
> > I've talked with various people at Polycom about getting on their
> developer
> > program and they have been stalling me for weeks. The most information I
> > received was from Scott Willard, head of IP phone sales for Polycom. He
> said
> > that the company is not sure of what direction they want to head with
> > selling phones to end users in the IP-phone area(and letting their
> firmware
> > and admin manuals out in the open). They are planning to continue
> > development of the IP phone line and not get into the PBX area. I sold
the
> > virtues of Asterisk to the best of my abilities and he seemed to be
> > interested in at least the possibility of working with the community,
but
> he
> > said the final decision would be made by the IP-phones division head of
> > development. When he did not say.
> >
> > Scott also shed a little light on Polycom relationship with other IP
phone
> > companies, he said that Polycom licenses a lot of their speakerphone and
> > other phone technologies to many companies

Re: [Asterisk-Users] more questions

[Tilghman Lesher]

On Monday 15 December 2003 21:21, Isianto Istiadi wrote:
> > > 3. Supposed I have 2 fxo cards (right now I have one already)
> > > and 3 fxs, and one of the fxo will have two phone (running
> > > pararell), is there any way for * to:
> > > a. It always dial the first fxo, if the fxo is busy or is being
> > > used (have other people conversation), will * be able to switch
> > > it to other fxo? Here's the approximiate the conditions of the
> > > phone.
> >
> >
> > Yes, define both FXO ports in the same group (e.g. 0) then dial
> > numbers using the group, (e.g. Dial(Zap/g0/2345678)).
>
> Thanks Tilghman, but the above command won't guaranteed the first
> FXO, will get dialed first, will it?

Why don't you try it first, before doubting my answer?

> More questions though
> 1. How to record a conversations? Can do it with asterisks? Can you
> give me the example of configurations?

show application monitor

> 2. How to forward call to other number automatically, say that my
> friend call me to *, but nobody pick it up, can it automatically
> forward the call to my cellphone number? If it can, can you show me
> how to do it?

Yes.

Dial(Zap/5,30)
Dial(Zap/2/2345678)

> 3. for music on hold do I need to put something in my
> extension.conf or it will work out of the box?

musiconhold.conf

-Tilghman

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[Asterisk-Users] Stupid Newbie Questions

[TkVoice Author]

I just learned about Asterisk yesterday.

Besides all the cool PBX stuff that asterisk can do can it also do these:

1) Receive FAX's ?

2) Control (serve) pppd dial-in connections ?

3) Is it possible to use voice modems as the FXO/FXS cards ?

Thank You
--
Peter
http://tkvoice.netfirms.com/
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[Asterisk-Users] Voice & Data Systems Engineer Career Opportunity

[Bob Murtagh]

Greetings,  

 

All,

 

Video Internet Broadcasting Corporation, a triple play services provider
(www.vib.tv)
, is looking to fill an open position for a Voice and Data Systems
Engineer.  I have attached a
summary of the qualifications as well as a job description.  Ideally the position will be based in
Seattle, WA and will entail 60+% travel. 
Compensation is based on experience.  Please send any inquiries and or resumes to me at [EMAIL PROTECTED].

 

Voice
& Data Systems Engineer (SE)

 

Candidate Qualifications:

 


 5+ years experience providing engineering support for Telecom
 systems, IP telephony systems, IP networks 
 Must have customer-facing experience with an ability for both
 technical and business problem solving 
 Specific experience designing, installing, and supporting
 Class 5 Switches, IP Softswitches, and IP data networks with a preference
 for carrier access network experience 
 Engineering degree in
 Telecommunications/Electronics – BS in Eng. a plus 
 This position requires excellent
 interpersonal and communications skills 
 Organize and track documentation 
 Available to travel and work after
 business hours, anticipate 60% travel 


 

Job Description
Systems Engineer (SE):

 

This position will
work closely with Sales, Marketing, and the General Managers of our Service
Provider Field Operations.  The
role will be to provide technical engineering, design, implementation, and
technical support in all manners of field operations.  

 

Specifically the
Systems Engineer will be driving the technical aspects of our emerging Voice
Services and providing engineering supporting our ongoing IP data
services.  Aspects of our Voice
Services include creating solutions for our Municipal, Military and IOC clients
for Voice over IP services in their given network environment.  It will be expected that the SE
rigorously and proactively develop their own required knowledge base in order
to provide technical leadership through the sales and implementation process of
our voice offerings.  

 

The successful
Systems Engineer in fulfilling this role will aid in the creation of increased
revenue, increased customer satisfaction, and critical feedback to our central
Marketing and Software Engineering teams.

 

 

 

Bob Murtagh

VP Sales

VIB Corporation

p 925-485-0880

c 925-998-0727

www.vib.tv

 

RE: [Asterisk-Users] Free Software/Open Source-Telephony-Summit 2004

[Philipp von Klitzing]

Hi!

> Just out of curiosity, are these types of workshops generally conducted in
> German, or in English?

No clue, but looking at their web site and the list of participants I am 
pretty sure it'll be English.

Cheers, Philipp


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RE: [Asterisk-Users] IP 500/600 1.1.0 Firmware

[mattf]

Hello,

I've created a page on Wiki for the Polycom phones:
http://www.voip-info.org/wiki-Polycom+Phones

I don't have a problem with putting the FAQ and sample config for these on
the voip-info site except for the fact that voip-info.org is really slow,
does anyone else notice that? What kind of connection is it on?

I will still have the downloads available on a faster server and will keep
copies of the FAQ and sample configs there when we have them finished.

I am waiting to hear back from Scott from Polycom about the final agreement,
but what it looks like is that the Asterisk community will have to support
their phones on Asterisk by itself. This means that a few people will have
access to the full Polycom developer system and we will post the new
firmware releases on a website as they are made available. Also any
non-defective-product support queries made to Polycom about integrating with
Asterisk will be directed by Polycom to the Asterisk-Polycom support
website(when it's up and running).

MATT---



-Original Message-
From: Miguel Cavazos [mailto:[EMAIL PROTECTED]
Sent: Monday, December 15, 2003 12:14 PM
To: '[EMAIL PROTECTED]'
Subject: RE: [Asterisk-Users] IP 500/600 1.1.0 Firmware


good to hear theres going to be support for this phones, but why not put
it on the wiki??? so we can have all the faq in one place.

Miguel
On Mon, 2003-12-15 at 22:23, mattf wrote:
> I just got off of the phone with Scott Willard at Polycom and things seem
> promising. He's going to send me the latest stable boot firmware for the
> Soundpoint 500/600 phones and I will make that available on a webiste
> somewhere for people to download. 
> 
> It seems like their big stumbling block to giving us the firmware and
admin
> docs for their phones is that they don't want to have to support end users
> of their phones. They usually have their resellers do support for specific
> PBXs that their phones work with and because none of their resellers
support
> Asterisk or sell Asterisk that means no one supports Asterisk users of
their
> phones. 
> 
> I told him that we would be putting up a support website for their phones
> that they can point Asterisk end users to, to download the latest firmware
> and the admin docs as well as contact people for help with configuration.
He
> seemed to accept that idea and is going to send me a copy of the latest
> bootloader as a sign of goodwill for stalling for so long. 
> 
> I am to present him with a document outlining what kind of support we will
> provide and how they can point end users that call polycom to us so that
we
> can help. I plan on putting up a simple website with a few FAQs and sample
> configs as well as some people that can be emailed or called to ask
> questions about Polycom on Asterisk. I'd love to have a few volunteers for
> that list so I don't get swamped :)
> 
> I plan on hosting the website(donated by my company of course) at least in
> the near term just to get something up.
> 
> Let me know if you're interested in helping out. I'll post the new
> bootloader on a web site when I get it.
> 
> Thanks,
> 
> MATT---
> 
> 
> 
> -Original Message-
> From: mattf [mailto:[EMAIL PROTECTED]
> Sent: Monday, December 15, 2003 4:01 PM
> To: '[EMAIL PROTECTED]'
> Subject: RE: [Asterisk-Users] IP 500/600 1.1.0 Firmware
> 
> 
> Hello,
> 
> I've talked with various people at Polycom about getting on their
developer
> program and they have been stalling me for weeks. The most information I
> received was from Scott Willard, head of IP phone sales for Polycom. He
said
> that the company is not sure of what direction they want to head with
> selling phones to end users in the IP-phone area(and letting their
firmware
> and admin manuals out in the open). They are planning to continue
> development of the IP phone line and not get into the PBX area. I sold the
> virtues of Asterisk to the best of my abilities and he seemed to be
> interested in at least the possibility of working with the community, but
he
> said the final decision would be made by the IP-phones division head of
> development. When he did not say.
> 
> Scott also shed a little light on Polycom relationship with other IP phone
> companies, he said that Polycom licenses a lot of their speakerphone and
> other phone technologies to many companies including Cisco and Avaya but
> their hardware is made entirely differently than those companies, the
> resemblance to the Cisco phone line is purely coincidental.
> 
> 
> MATT---
> 
> 
> -Original Message-
> From: Bisker, Scott (7805) [mailto:[EMAIL PROTECTED]
> Sent: Monday, December 15, 2003 3:10 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] IP 500/600 1.1.0 Firmware
> 
> 
> Has anyone on the list been able to locate and try out the 1.1.0 firmware?
> It was released in November, but I have yet to get my hands on it.  The
> Polycom site has way more docs online, but the link to the firmware only
> brings up the release notes.
> 
> -sb
> ___

Re: [Asterisk-Users] iaxclients missing calls

[Dan]

Hi,

>- Original Message - 
>From: "Patrick Cantwell" <[EMAIL PROTECTED]>
> Just a thought:  Out of everyone having problems, how many are connecting
> through NAT to their asterisk server? (client -> nat -> ... -> asterisk).
>
> NAT firewalls time ports out after a certain amount of inactivity.  I was
> having the same problem with my SIP analog adapter.  I could fire it up
and
> receive calls, but after a little while, it could no longer receive calls
> (yet asterisk would show them coming in).  If I made a call out from the
> analog phone, I could again receive calls for a short while.  Then it
> occurred to me that I probably didn't change my firewall rules after
> renumbering my asterisk box and it was a NAT issue.
>
> If anybody having this problem is running behind some sort of NAT device,
> try making your registration time rediculously low, such as 60s and see if
> it has an effect. (or try changing your rules, if possible)
>
> Just my .02 =)

A debuging version of DIAX (just for IAX2) will be available in a couple of
days.
Then we will be able to better track this issue.

Thank you for your understanding and patience.

Best regards,
Dan


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RE: [Asterisk-Users] Cisco Gateway Integration

[Steve Dolloff]

I am using it with the AS5350 via SIP and it works great.  I was also
using the ATA186 with SIP but I am switching to the SPA-2000 for a
better feature set.

Stephen

> -Original Message-
> From: Bruce Hedreen [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, December 16, 2003 8:29 AM
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] Cisco Gateway Integration
> 
> Did you use the h323 module on asterisk?
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Anton
> Tinchev
> Sent: Tuesday, December 16, 2003 12:37 AM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] Cisco Gateway Integration
> 
> 
> Bruce Hedreen wrote:
> 
> > Has anyone succesfully integrated * with a cisco voice gateway ?
> >
> >
> Works well with AS5350 and ATA186.
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> 
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RE: [Asterisk-Users] Free Software/Open Source-Telephony-Summit 2004

[Scott Stingel]

Hi Philipp-

Just out of curiosity, are these types of workshops generally conducted in
German, or in English?

Cheers
Scott 

London

Scott M. Stingel 
Emerging Voice Technology Inc.

Email:  [EMAIL PROTECTED]    
URL:www.evtmedia.com    



> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Philipp von Klitzing
> Sent: Tuesday, December 16, 2003 3:35 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Free Software/Open 
> Source-Telephony-Summit 2004
> 
> 
> Hi,
> 
> I just came across this annoncement, which is particularly 
> interesting as 
> it is only 25 min away from my place... :-) Anyway, I guess 
> the core of 
> this is targeted at developers mainly.
> 
> Cheers, Philipp
> 
> 
> Free Software/Open Source-Telephony-Summit 2004
> 
http://www.guug.de/veranstaltungen/telephony-summit-2004/
http://www.heise.de/newsticker/data/avr-16.12.03-000/

We are happy to announce that the first summit on Free Software/Open 
Source-telephony solutions is going to take place from January 16th till 
20th in Geilenkirchen, Germany. 

The event will be divided into three parts:

a developer workshop from January 16th to January 18th 
a conference day (January 19th) 
a tutorial day (January 20th) 
There will be an exhibition during the conference and the tutorial day.

The developer workshop is free of charge and only-open for active 
developers in Free Software/Open Source telephony projects. If you are 
interested in participating, please contact Martin Schulte telephony-
[EMAIL PROTECTED]

During the one-day conference, the participating projects will give an 
overview about their current status and their future goals.

In the morning of the tutorial day, people interested in using a 
particular software will get an in depth-introduction to installation, 
configuration and usage by their developers. In the afternoon, there's a 
big "How everything works together"-tutorial.

Online-Registration is opened!


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Re: [Asterisk-Users] iaxclients missing calls

[Philipp von Klitzing]

Hi!

> I'm wondering if the iaxclient lib might be having a problem with registration
> renewal, or if it's a problem with event dispatching.

Maybe this can be of help:
http://bugs.digium.com/bug_view_page.php?bug_id=673
http://bugs.digium.com/bug_view_page.php?bug_id=558

Cheers, Philipp


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[Asterisk-Users] Free Software/Open Source-Telephony-Summit 2004

[Philipp von Klitzing]

Hi,

I just came across this annoncement, which is particularly interesting as 
it is only 25 min away from my place... :-) Anyway, I guess the core of 
this is targeted at developers mainly.

Cheers, Philipp


Free Software/Open Source-Telephony-Summit 2004

http://www.guug.de/veranstaltungen/telephony-summit-2004/
http://www.heise.de/newsticker/data/avr-16.12.03-000/

We are happy to announce that the first summit on Free Software/Open 
Source-telephony solutions is going to take place from January 16th till 
20th in Geilenkirchen, Germany. 

The event will be divided into three parts:

a developer workshop from January 16th to January 18th 
a conference day (January 19th) 
a tutorial day (January 20th) 
There will be an exhibition during the conference and the tutorial day.

The developer workshop is free of charge and only-open for active 
developers in Free Software/Open Source telephony projects. If you are 
interested in participating, please contact Martin Schulte telephony-
[EMAIL PROTECTED]

During the one-day conference, the participating projects will give an 
overview about their current status and their future goals.

In the morning of the tutorial day, people interested in using a 
particular software will get an in depth-introduction to installation, 
configuration and usage by their developers. In the afternoon, there's a 
big "How everything works together"-tutorial.

Online-Registration is opened!


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Re: [Asterisk-Users] AutoAttendent ON/OFF control by Attendent

[Sri]

Todd Lieberman wrote:

Using a Q is nice because you can have another employee or group of 
employees cover while the receptionist is on break.

TL

Olle E. Johansson wrote:

Tilghman Lesher wrote:

On Monday 15 December 2003 10:57, Sri wrote:
...
exten => *801,1,DBPut(auto/attendant=1)
exten => *802,1,DBPut(auto/attendant=0)
exten => s,1,DBGet(autoattendant=auto/attendant)
exten => s,2,GotoIf($[${autoattendant} = "1"]?auto|1)
exten => s,3,Dial(Zap/23,30,t)
exten => s,4,Goto(auto|1)


Thank you all!
http://www.voip-info.org/tiki-index.php?page=Asterisk+tips+autoattendant
created.

Yes. Thats true. But the requirement is to have it move to AutoAttendent 
which has options for different departments
and directory listing, when the attendent is out. Either the Attendent 
or Auto-Attendent.. no one else.
Othewise, Yes. Maybe a Q can come in handy.

Sri



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RE: [Asterisk-Users] Cisco Gateway Integration

[Bruce Hedreen]

Did you use the h323 module on asterisk?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton
Tinchev
Sent: Tuesday, December 16, 2003 12:37 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Cisco Gateway Integration


Bruce Hedreen wrote:

> Has anyone succesfully integrated * with a cisco voice gateway ?
>  
> 
Works well with AS5350 and ATA186.
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RE: [Asterisk-Users] iaxclients missing calls

[Patrick Cantwell]

Just a thought:  Out of everyone having problems, how many are connecting
through NAT to their asterisk server? (client -> nat -> ... -> asterisk).

NAT firewalls time ports out after a certain amount of inactivity.  I was
having the same problem with my SIP analog adapter.  I could fire it up and
receive calls, but after a little while, it could no longer receive calls
(yet asterisk would show them coming in).  If I made a call out from the
analog phone, I could again receive calls for a short while.  Then it
occurred to me that I probably didn't change my firewall rules after
renumbering my asterisk box and it was a NAT issue.

If anybody having this problem is running behind some sort of NAT device,
try making your registration time rediculously low, such as 60s and see if
it has an effect. (or try changing your rules, if possible)

Just my .02 =)

-Pat

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Michael Van
Donselaar
Sent: Monday, December 15, 2003 11:25 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] iaxclients missing calls


On Mon, 15 Dec 2003 23:05:56 -0500 (EST), [EMAIL PROTECTED]
wrote:

>Hello All
>When I open up iaxcomm, it registers fine with the asterisk server.  If I
>call into it, iaxcomm will ring; however if I leave iaxcomm sitting idle
>for awhile (I haven't figured out exactly how long) it seems to miss
>calls.  I can see the calls coming in on the asterisk server but they
>never ring through on iaxcomm.  If I close it and reopen it, it takes
>calls again fine.  I thought I saw someone else who had experienced this
>problem too but I don't recall seeing a solution.  Any thoughts or
>solutions?  I'm trying to use iaxcomm in my every daybusiness but it's kind
of
>hard when I miss calls.  It complicates things a bit.  Although I used
>iaxcomm as an example here I've experienced the same problem with DIAX.
>
>Thanks a bunch.
>AJ

Hi, AJ.

It sounds like this is related to the other incoming ring issue.  I'll check
through the archives to verify, but I think a couple of other users were
seeing
this even soon after launching iaxComm or DIAX.

I'm wondering if the iaxclient lib might be having a problem with
registration
renewal, or if it's a problem with event dispatching.

Any other users seen this yet? (That is, seen incoming ring failures become
more
likely as uptime increases).

The next time you get an ignored incoming call, could you try doing an iax2
debug on the cli?

Do you see the same problem with DIAX when using IAX instead of IAX2?


>
>
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Re: [Asterisk-Users] wiax

[Dan]

Hi,


>- Original Message - 
>From: Rattana BIV
>To: [EMAIL PROTECTED]
>Subject: [Asterisk-Users] wiax
>
>Hi,
>
>I try to use wiax.dll in a application.
>Is there some docs about this DLL ?
>Sample ?

First, please do not post this type of messages on this list.
Iax-devel on Sourceforge is a better place for that.

Wiax.dll is just a wrapper for the iaxclient library, nothing more.
It is usefull just if you want to develop further using MS platform (VC or
VB).
Take a look at the testcall or WInIAX sources (which are in the CVS) to see
the call flow.

Best regards,
Dan

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Re: [Asterisk-Users] AutoAttendent ON/OFF control by Attendent

[Brancaleoni Matteo]

just an advice...
what happens if the Q has no members?
ie they're all dynamic :)

prepare a voicemail after the Q prio, in order
not to get lost calls .

Matteo.

Il lun, 2003-12-15 alle 21:53, Todd Lieberman ha scritto:
> Using a Q is nice because you can have another employee or group of 
> employees cover while the receptionist is on break.
> >
> > Thank you all!
> > http://www.voip-info.org/tiki-index.php?page=Asterisk+tips+autoattendant
> > created.
> >
> > /O
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
Brancaleoni Matteo <[EMAIL PROTECTED]>
Espia - Emmegi Srl

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[Asterisk-Users] wiax

[Rattana BIV]

Hi,

 
I try to use wiax.dll in a 
application.
Is there some docs about this DLL ?
Sample ?
 
 
Regards
Rattana

Re: [Asterisk-Users] AutoAttendent ON/OFF control by Attendent

[Matteo Brancaleoni]

but also remind to add a voicemail after the Q.
What happens if all the members are logged out, 
is to say when the Q has no members? calls get lost,
if not voicemail :)

matteo.

Il lun, 2003-12-15 alle 21:53, Todd Lieberman ha scritto:
> Using a Q is nice because you can have another employee or group of 
> employees cover while the receptionist is on break.
> 
> TL

-- 
Matteo Brancaleoni
Espia System Administrator
Email : [EMAIL PROTECTED]
Web   : http://www.espia.it
Phone : +39 02 70633354  - ext 201
IAX(2): [EMAIL PROTECTED] - ext 201
Iaxtel: 1-700-56-62458   - ext 201

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[Asterisk-Users] voicemail

[Deepakumar JV]

Hello
 
Has anyone implemented a voicemail system 
for an existing cisco call manager express VOIP setup. I am looking for a open 
source based software which can integrate with my existing Cisco call manager 
express VOIP (based on 3640 router) with individual users having their own 
voicemail box.
 
Thanks, Deepak

[Asterisk-Users] Help! VoiceTronix Multi FXO/FXS Problem

[Jacky]

Hello, Hacker

I install VoiceTronix OpenSwitch 12 port PCI Telephone Card,
and setting vpb.conf, extensions.conf following

My problem is:

When i dial to fxo(channel 9-12), it is ok,
but when i continue press exten 102, the channel crach with error messages
following
exception caught: VPBAPI_DIAL_INVALID_LEVEL, file: vpbdial.cpp line:872

Do i ignore some setting for VoiceTronix OpenSwitch12 Card, Or other issues?



**
Call status

--
--  Event [0=>[08] Ring
] on vpb/1-9
-- Executing Wait("vpb/1-9", "1") in new stack
 Read_channel ##  vpb/1-9: Setting record mode, bridge = 0
--  Event [0=>[08] Ring
] on vpb/1-9
-- Executing Answer("vpb/1-9", "") in new stack
-- Executing DigitTimeout("vpb/1-9", "5") in new stack
-- Set Digit Timeout to 5
-- Executing ResponseTimeout("vpb/1-9", "10") in new stack
-- Set Response Timeout to 10
-- Executing BackGround("vpb/1-9", "demo-congrats") in new stack
-- Playing 'demo-congrats' (language 'en')
--  Event [8=>[08] DTMF digit: 1
] on vpb/1-9
--  Event [2=>[08] Tone Detect: Grunt
] on vpb/1-9
--  Event [8=>[08] DTMF digit: 0
] on vpb/1-9
--  Event [8=>[08] DTMF digit: 2
] on vpb/1-9
--  Event [2=>[08] Tone Detect: Grunt
] on vpb/1-9
  == CDR updated on vpb/1-9
-- Executing Dial("vpb/1-9", "vpb/1-2|20|tT") in new stack
 Read_channel ##  vpb/1-2: Setting record mode, bridge = 0
--  1-2 requested, got: [vpb/1-2]
--  VPB Calling 1-2 [t=0] on vpb/1-2 returned 0
-- Called 1-2
-- vpb/1-2 is ringing
exception caught: VPBAPI_DIAL_INVALID_LEVEL, file: vpbdial.cpp line:872



(1) vpb.conf

---
[interfaces]

echocancel = on
board = 1

context = vpbtest

; Note that V6PCI channel numbers start at 7!
mode = fxo
channel = 9

mode = dialtone
channel = 1
channel = 2
channel = 3
channel = 4
channel = 5
channel = 6
channel = 7
channel = 8


(2) extensions.conf

-
..original exten so ignore...

[vpbtest]
include => default
exten => 102,1,Dial(vpb/1-2,20,tT)
exten => 102,2,Hangup

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AW: [Asterisk-Users] CLIP in Germany

[Johannes von Drachenfels]

Hi,

i have a Siemens Gigaset phone with clip-capabilities on the tdm400 with
extension 610 installed. Extension 600 is another phone. But i still cannot
see the number of the caller at the display of the Siemens phone... here is
my config:


; Main extension
exten => 600,1,SetCallerID,${CALLERIDNUM}
exten => 600,2,Dial,Zap/33|30

; Johannes von Drachenfels's Extension
exten => 610,1,SetCallerID,${CALLERIDNUM}
exten => 610,2,Dial,Zap/32|30
exten => 610,3,VoiceMail,u610
exten => 610,103,VoiceMail,b610

... and the debug of a call from 600 to 610:

cm*CLI>
-- Starting simple switch on 'Zap/33-1'
-- Executing SetCallerID("Zap/33-1", "600") in new stack
-- Executing Dial("Zap/33-1", "Zap/32|30") in new stack
-- Called 32
-- Zap/32-1 is ringing
-- Zap/32-1 is ringing
-- Zap/32-1 is ringing

So what i would like to see is 600 in the display of 610 ...

Any help is great,

Johannes

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RE: [Asterisk-Users] voicemail as an attachement

[Dawid Mielnik]

Hi again,

sorry for the spam but, I have tried connecting * to a different internet
connection (isp) and the mail attachments work ok now - dont know why.

Best regards,

Dave

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Dawid Mielnik
Sent: Tuesday, December 16, 2003 10:30 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] voicemail as an attachement



It works with "attachment=no" !
[EMAIL PROTECTED] is my masqueraded (by sendmail) [EMAIL PROTECTED]

Any ideas ?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Andrew Thompson
Sent: Monday, December 15, 2003 11:55 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] voicemail as an attachement


- Original Message -
From: "Dawid Mielnik" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, December 15, 2003 4:16 PM
Subject: [Asterisk-Users] voicemail as an attachement


>
> Hi,
>
> I can not send voicemails as an attachement. When setting the "attach=yes"
> option in voicemail.conf the mails get rejected from the mail server:
>
>- Transcript of session follows -
> 451 4.4.1 timeout writing message to higgs.elka.pw.edu.pl.: Connection
timed
> out
>  with higgs.elka.pw.edu.pl.
> 451 4.4.1 timeout writing message to elektron.elka.pw.edu.pl.: Connection
> timed
> out with elektron.elka.pw.edu.pl.
> ... while talking to mion.elka.pw.edu.pl.:
> >>> MAIL From:<[EMAIL PROTECTED]> SIZE=21382
> <<< 553 5.7.1 For MAIL FROM address <[EMAIL PROTECTED]> access is denied by
the
> pol
> icy analysis functions.
> 501 5.6.0 Data format error
>
> Can anyone help me out with this ? Why is there a data format error.

You're not reading enough of the error message...

The mail is failing because it doesn't like your sending email address.

If it works with attach=no, I'd be suprised. Do the notifications come from
the same email address([EMAIL PROTECTED]) when the attachments are disabled?

-
Andrew Thompson http://aktzero.com/


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[Asterisk-Users] Re: Slightly OT and mildly insane: Modems through VoIP :-))

[Darren McIntosh]

> Message: 2
> Date: Mon, 15 Dec 2003 23:13:46 +0100
> From: Florian Overkamp <[EMAIL PROTECTED]>
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Slightly OT and mildly insane: Modems through
VoIP :-))
> Reply-To: [EMAIL PROTECTED]
>
> Hi,
>
> First off, let me state that _YES, I am fully aware that what I am doing
is
> insane, prone to major havoc and bad for general health_ :-))
>
> Scenario: My GF needs an analog modem to use with her banking software
> (sodding backs don't supply a decent web-application for company use). I
am
> experimenting to see if we can get it to work (albeit slow) trough our
ATA186
> talking g711 to Asterisk with chan_capi to the outside world. Should we
fail,
> there are more sane alternatives, but humour me :-))
>
> Now, with some modems on the other end I have received nice 26400bps
> handshakes, but it takes a long time and the successrate is about 25% :-)
My
> feeling is this should be better if we choose to slow it down more.
>
> But who can tell me what the best modem settings would be to try ? My
HAYES
> dialect is rather old :-))
>
> Any experiences or hints are appreciated.
>
> -- 
> Best regards,
> Florian Overkamp
>
I've made data calls over g.711 voip from australia to the uk and they
trained up to 31k2 fine. What will throw them is packet loss. Just one lost
20ms frame was normally enough to force a retrain. Lose a packet during
retraining and the call would probably drop. Might pay to check the duplex
settings on your ata and whatever its connected to.
echo cancellers could possibly interfere with things but they should be
disabled by the 2100hz tone at the front of the call.

cheers,
darren

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RE: [Asterisk-Users] voicemail as an attachement

[Dawid Mielnik]

It works with "attachment=no" !
[EMAIL PROTECTED] is my masqueraded (by sendmail) [EMAIL PROTECTED]

Any ideas ?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Andrew Thompson
Sent: Monday, December 15, 2003 11:55 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] voicemail as an attachement


- Original Message -
From: "Dawid Mielnik" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, December 15, 2003 4:16 PM
Subject: [Asterisk-Users] voicemail as an attachement


>
> Hi,
>
> I can not send voicemails as an attachement. When setting the "attach=yes"
> option in voicemail.conf the mails get rejected from the mail server:
>
>- Transcript of session follows -
> 451 4.4.1 timeout writing message to higgs.elka.pw.edu.pl.: Connection
timed
> out
>  with higgs.elka.pw.edu.pl.
> 451 4.4.1 timeout writing message to elektron.elka.pw.edu.pl.: Connection
> timed
> out with elektron.elka.pw.edu.pl.
> ... while talking to mion.elka.pw.edu.pl.:
> >>> MAIL From:<[EMAIL PROTECTED]> SIZE=21382
> <<< 553 5.7.1 For MAIL FROM address <[EMAIL PROTECTED]> access is denied by
the
> pol
> icy analysis functions.
> 501 5.6.0 Data format error
>
> Can anyone help me out with this ? Why is there a data format error.

You're not reading enough of the error message...

The mail is failing because it doesn't like your sending email address.

If it works with attach=no, I'd be suprised. Do the notifications come from
the same email address([EMAIL PROTECTED]) when the attachments are disabled?

-
Andrew Thompson http://aktzero.com/


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RE: [Asterisk-Users] voicemail as an attachement

[Dawid Mielnik]

I have tried sending mail notifications to different mail servers. Again,
each time I attache the voicemail - they fail.

regards,

Dave

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Tilghman Lesher
Sent: Monday, December 15, 2003 11:49 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] voicemail as an attachement


On Monday 15 December 2003 15:16, Dawid Mielnik wrote:
> I can not send voicemails as an attachement. When setting the
> "attach=yes" option in voicemail.conf the mails get rejected from
> the mail server:

This is not a problem with Asterisk.  Have a talk with your mail admin
about this destination mail server problem.

-Tilghman

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Re: [Asterisk-Users] E400 or TE410 (digium) vs PRI 30M (Eicon)

[Daniel ANDRE]

Roy Sigurd Karlsbakk a écrit:

  
I have used a test server with a passive card unde I4L and encountered
echo pb whe dialing out to pstn side from any IP Phone I have tested. So
I am considering to get some BRI HW with echo cancellation circuit. I
have found only card from Eicon. If my pbx extend to more than 15 lines,
I will use PRI Card so my question.

  
  
FWIK, running I4L may well result  in echo problems. Get some supported 
hardware :)

I will use Eicon BRI Cards as they have Echo cancel on board.

  

  
  

  Something you will also need to consider, what software will you be
using? Digium cards work with asterisk, and I doubt that the Eicon E1
cards do yet unless they use the capi driver.
  

The Eicon driver for PRI seems to be CAPI Compliant if I have understood
there doc.

  
  
The Eicon PRI card is quite possibly a more expensive, and not neccecarily a 
better solution IMHO.

Yes the Eicom PRI Card is very expensive and that was the reason of my
question. I am searching a No echo solution and if the E400 meats this
criteria I will undoubtly order this card when I will have to switch to
PRI Card.

Daniel


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

[Asterisk-Users] Re: transfer with threeway calling

[Cees de Groot]

Steven Critchfield  <[EMAIL PROTECTED]> said:
>Not exactly. The complaint here was that if a call was parked the user
>thought they had to wait for the timeout to have the call bounce back to
>their phone. My comment was that the user was just as capable of dialing
>the parked call as the intended transfer target, thus removing the wait
>for timeout.  
>
Which is exactly what I said, but let's not start an argument over who
said what - suffice it to conclude we're still talking about the same
topic ;-)

>> Which is a completely and utterly wrong answer to users complaining
>> about usability.
>
>Not always. Sometimes it is absolutely necessary.
>
Is there anything absolutely necessary, probably caused by *'s design,
that mandates that old-fashioned transfer mechanisms shouldn't work? 

>This is where it might come down to redesigning the way calls are dealt
>with in an organization. Sometimes new phone systems do this, and
>hopefully the company sees new efficiencies with dealing with the
>customer in general. 
>
Oh, I'm fully aware of the value of AA, IVR, and whatnot. In fact, we're
working hard to implement this in the various companies I'm involved
with. But it will not completely eliminate all transfers, and I must say
that it's not up to the 'technologists' to decide how this then should
be handled (barring some hard technological limits, budget limitations,
etcetera). If a customer wants to do old-fashioned, out-of-date, stupid
stuff like faxing, flash-transfers or wearing digital watches, we may
object, protest, and call the customer a [expletive deleted]; but then
we should shrug and implement the damn thing.

>All phones more complicated than a home phone start as a barrier to a
>new user. So does that thing usually on the right of the keyboard.
>Eventually they learn to use it and understand it, if it was implemented
>well to begin with.
>
So you throw Linux at people's desktops and say "stop whining, you'll
adapt eventually"? Or Macs, for that matter?

Steering users towards new behavior is ok. Ramming it down their throats
never works (and I should know - I've been in the 'convert people from
Word to StarOffice' market, it is always a matter of a lot of patience,
and the ONLY thing that finally gets them across is the promise of
compatibility).

>[...] Also on
>the occasions I have to park a call, I can usually walk over to the
>person about to answer the call and give them the quick 10 second
>briefing before they take the call.
>
We're completely virtual, we don't even have an office. I'd need to hop
in the car and drive for well over an hour to reach some colleagues ;-)

But the issue is not: 'how does the alternative feature work', the issue
is 'why is the original feature absent'. I haven't heard anyone giving
any reason whatsoever why * does not allow a user to retrieve an on-hold
call with old-fashioned flashing (or pressing #). I think that is what
the debate should focus on, not on whether the customer is right...


-- 
Cees de Groot   http://www.tric.nl <[EMAIL PROTECTED]>
tric, the new way   helpdesk/ticketing software, VoIP/CTI, 
web applications, custom development

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Re: [Asterisk-Users] AVM ISDN Fritz!Card USB works

[Michiel Betel]

Anthony Wood wrote:

On Tue, Dec 16, 2003 at 11:28:38AM +1100, Gonzalo Servat wrote:
 

On Tue, 2003-12-16 at 10:34, Michiel Betel wrote:
   

Is case anyone wants to know... The Fritz! USB ISDN box works fine with 
Asterisk!
I'm running CAPI 0.3.0 and love it, because the mini ITX server I have 
only takes one PCI slot which is now filled with a 4 port Digium card.
 

Is this the micro PABX model 4FXS + 1 ISDN FXO (USB 3.0) or the plain 1 ISDN FXO (USB 2.0)?

cheers,
Woody
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It's the little single BRI FXO that AVM calls the
FRITZ!CARD USB. Have not tested the other one, athough
there is a CAPI driver available for that box too.
No need for the mico PABX when running
Asterisk is there?
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