Re: [Asterisk-Users] Asterisk-Users List Etiquette

[Brian Capouch]

Andrew, you are right on with your final point about absurdity.
Hopefully this vile top-posting will illustrate exactly why.
Sorry, I couldn't resist.
B.
Andrew Kohlsmith wrote:
On Thursday 17 June 2004 09:21, Troy Settle wrote:
However, my preference is for top posting.  The reason, is that in order to
read my message here, you had to scroll through ~70 lines of previous
discussion.  Stuff that you've /already/ read since you've been following
this thread.

That's because you didn't trim anything.  To see what I wrote to you You had 
less than 10 lines to look at.  Please don't use absurdity to try and prove 
your point.


Oh!  Wait, you found this in an archive, so you /want/ to have the thread
fully quoted so you don't have to go hunting down the references.  Good,
that's why I didn't trim this post.

Um no, that's why the archives are threaded themselves.  Attempt at reductio 
ad dbsurdum #2 failed.


Oh, wait, the guys that are following this thread as it's being discussed
would prefer that I trim out the stuff up there, in which case, I would be
neither top posting, nor bottom posting.  This message would be a post unto
itself that wouldn't have any quoted material at all.  Afterall, you've
already read the referenced material.

I consider trimming the quoted text and replying to the bits you keep as they 
occur bottom posting -- your text is FOLLOWING the relevant bits of the 
conversation.

Inline posting is something completely different and it's even more heinous:

So, the bottom line is that top-posters are lazy?  [ yes, they are 
absolutely.  Inline posters are even worse! ] I say yes, we are.  We
don't want to have to scroll through pages of quoted material just to get
to the new stuff.  [ so trim your damned posts ]

That above is an example of inline posting.  Some managers have a penchant for 
that.


I say that the bottom posters are lazy.  They want a bottom post so that
they enter into a thread 12 messages later, and not have to read the thread
'backwards.'  Read your mail to begin with, and you wouldn't have this
problem, and you would actually start to appreciate the top posters,
because they're making it so you don't have to scroll through ~70 lines of
quoted material to get to the new stuff.

That's not laziness, that is following natural language laws.  I have over 25k 
messages in my local copy of asterisk-users.  My MUA understands message 
threading so if people posted the One True Way (editing quoted content and 
replying underneath, as I am doing to you here) then there is no problem 
following the flow of the thread, and if I need more information I move up to 
the message parent and see the entire message.

It's not a difficult thing to understand, and this absurdity you're spewing to 
try and prove your point only goes to show that your argument doesn't hold 
much logic.

Regards,
Andrew
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Re: [Asterisk-Users] Need guides on setting up PDA on asterisk server

[Holger Schurig]

 I should follow this up to accurately state that audio was not
 operational in my test calls from the PDA.  I have patched the
 iaxclient library with the changes available from ZiaxPhone that word
 align the IAX2 library on the ARM platform.  I haven't finished
 compiling a new binary to test with.

If you want to even patch ZiaxPhone, you can't: there's no source.

There is something similar at my homepage, 
http://www.holgerschurig.de/qtiax.html. It doesn't yet run on my PDA) and 
lacks a config file support, but it's all source code.

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Re: [Asterisk-Users] Maximum retries exceeded on call

[Holger Schurig]

 i'm new to asterisk and am having trouble placing outbound calls.  i

Bug Grandstream so that they finally fix their buggy software.

The GS phone sends occassional SIP packets to port 0, not to port 5060, as 
tcpdump or (better) ethereal will show you.

There's a page on this at voip-info.org.


I'd love to see that we e-mail in MASSES to Grandstream, so that they fix 
their software. The problem is that it doesn't happen always. Try 
[EMAIL PROTECTED] :-)

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[Asterisk-Users] IAX Jitter Buffer

[George Pajari]

We have a customer who is connected to our PSTN gateway using IAX and
noticing that even when the traffic from their site is modest their outbound
audio has short dropouts. Inbound audio is fine. (They have ADSL so it is
expected that outbound audio would be the first to experience problems.)

We have several questions to pose to the collective wisdom of this list.

Q1: Are there any statistics collected/available or diagnostics tools to
tell us how much of this can be attributed to packet loss and how much to
packet jitter and to measure quantitatively how bad this is?

The use of the jitterbuffer in iax.conf seems to have problems. Extensive
searching turns up comments such as:

1. jitterbuffer, unfortunately, is buggy and don't work as expected.
[asterisk-users/2003-July/016029.html.]

2.  This supports my thinking that there is some sort of broken logic in
the IAX jitter buffer - HRH Mark Spencer
[http://www.marko.net/asterisk/archives/0302/0077.html]

When we enabled jitterbuffer the sound quality seemed to improve but we
noticed some problems:

(a) sometime we would get only one-way audio;
(b) other times we would experience no audio in one direction for between 1
and 4 seconds and then things would seem to work fine;
(c) some times users reported a clipped and almost half duplex sound
quality as the flow of the conversation shifted back and forth.

We also noticed some wingnut values for Lag and Jitter such as:
Lag: -65476ms
Jitter: 12897799ms

PSTN gateway is CVS-04/20/04-01:11:29 
Client machine is CVS-HEAD-06/02/04-07:56:41

Searching the Asterisk bug lists shows some significant fixes (1696, 1643).

Q2: Is jitterbuf working well enough to try again?

Q3: Any other suggestions for improving voice quality with IAX links?


Thanks.

g.

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RE: [Asterisk-Users] trying to set an internal ivr

[Jay Milk]

You're basically looking for hotline functionality.  I'm using Sipuras
for my FXS ports, and they can be configured to dial a phone number upon
pickup.  I played with that before, and the call was established so
quickly that I had to add a Wait instruction in there so the receiver
could make it to the ear :)

If you're using zap channels for FXS, you could do something line (in
zapata.conf):
context = instantpickup
immediate = yes
channel = 10-60

And then in extensions.conf:
[instantpickup]
exten = s,1,Answer
exten = s,1,Wait(1)
exten = VoiceMailMain()

(or whereever you want to go from here)

 -Original Message-
 From: Greg Hill [mailto:[EMAIL PROTECTED] 
 Sent: Thursday, June 17, 2004 6:53 PM
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] trying to set an internal ivr
 
 
 On Thu, 17 Jun 2004, PAZ wrote:
  I'm trying to implement an IVR for internal use for the 
 enterprise I 
  work for, but the goal I'm trying to reach is that the main menu of 
  this IVR present itself to the user after 5 seconds he picks up his 
  extension (and only if the user doesn't press any key, off 
 course). I 
  imagine the solution (if exists) maybe relies in timeout 
 properties, 
  but I can't see it. Any suggestions for my extension.conf file ?.
 
 once a connection is established to the server, you could 
 have exten = t,1,Goto(yourIVR) or similar. But that depends 
 on the phone making a connection to * as soon as the handset 
 is lifted. The xten softphone (currently my only SIP device 
 :( ) doesn't actually connect to the SIP server until you 
 push the call button. I guess that if a hardware phone 
 actually connects immediately, then you could probably make 
 the timeout extension work. Maybe you can adjust the timeout 
 length with exten = s,1,DigitTimeout(5) or something 
 similar. ResponseTimeout might work for that too.. I'm just 
 guessing, though.. I had an idea but no hardware to test on.
 
 Greg
 
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[Asterisk-Users] Draytek Vigor 2600Vi as SIP client on Asterisk

[Michael Hamann]

Hi Everybody,

as a relative newby I´m just trying to get a Draytek Vigor Router (2600Vi)
connected to my Asterisk System (CVS-05/31/04). With X-Lite and a Cisco
Phone it is no problem, but the Vigor seems to have some problems with
Asterisk.

The first thing ist when I do a sip show peers on the console I get:

4002/4002172.16.183.37   (D)  255.255.255.255  5060 Unmonitored
4001/4001172.16.183.37   (D)  255.255.255.255  5060 Unmonitored

What does this status unmonitored mean? With my softphone the entry looks
like:

6275/6275172.16.181.49   (D)  255.255.255.255  5060 OK (8 ms)

The next thing is that when I try to call one of the vigors SIP Ports via
X-Lite I see the following message in the debug console:

Jun 18 10:09:54 NOTICE[131081]: chan_sip.c:5150 handle_response: Dunno
anything about a 0 Unkown status code response from SIP/4001-b2fc

No call is signalled to the phone. The other way, my X-Lite rings but the
connection is hung up the moment I accept the call.

The Draytek support says that the Vigor does not support SIP Reinvite and
that I should try to disable it in my PBX system.

So I changed my sip.conf to:

[4001]
type=friend
username=4001
secret=4001
mailbox=2000
canreinvite=no
context=default
host=dynamic

But it still does not work. Does anybody has this combination working and
could send me his config files? Or any other ideas?

best regards from germany

Michael
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Re: [Asterisk-Users] IAX Jitter Buffer

[steve]

On Thu, 17 Jun 2004, George Pajari wrote:

 Q1: Are there any statistics collected/available or diagnostics tools to
 tell us how much of this can be attributed to packet loss and how much to
 packet jitter and to measure quantitatively how bad this is?

 Q2: Is jitterbuf working well enough to try again?
 
 Q3: Any other suggestions for improving voice quality with IAX links?
 

Hi George,

I'm looking at the jitter buffer and will persevere until it works right
for me.  (Here in South Africa Internet quality is not of US standard!)

I did find one small problem and have a fix which hopefully will go into 
CVS.  But I think further tweaking is also desirable.

I see on bugs.digium.com stevek has also submitted some adjustments which 
have stimulated discussion.

So check asterisk-dev, check bugs.digium.com and I think we'll get the 
jitter buffering right.

Steve

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RE: [Asterisk-Users] oh323

[T. Chan]

Jeremy

I speak for myself, I have been testing with oh323 driver as well, because
in my case, your h323 driver is not working, it was working before, but then
when I started to upgrade to 0.7.0 version of asterisk and from that point
onwards (beginning of January), calls have had no audio. I tried making
calls and I was getting no audio at all when the call was connected. Since
then, I have not been able to upgrade the asterisk version, because if so, I
would not be able to run h323. That is why in my case, I have been trying to
explore the other alternative. If you have some idea to it, please let me
know, thanks alot

TC

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jeremy
McNamara
Sent: Thursday, June 17, 2004 10:56 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] oh323


Michael M. Saunders wrote:

 Can I just pay you to fix it for me.

 I cant see anywhere where I use the debug


Why do you see a need to run a 3rd party channel driver?  Asterisk has
native H.323 support.



Jeremy McNamara

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Re: [Asterisk-Users] oh323

[Jeremy McNamara]

T. Chan wrote:
Jeremy
I speak for myself, I have been testing with oh323 driver as well, because
in my case, your h323 driver is not working, it was working before, but then
when I started to upgrade to 0.7.0 version of asterisk and from that point
onwards (beginning of January), calls have had no audio. I tried making
calls and I was getting no audio at all when the call was connected. Since
then, I have not been able to upgrade the asterisk version, because if so, I
would not be able to run h323. That is why in my case, I have been trying to
explore the other alternative. If you have some idea to it, please let me
know, thanks alot

So you didn't feel it was important to report your trouble anywhere?
I have tested the cvs -head of asterisk with many different types of 
H.323 gateways and cannot make it fail.

Jeremy McNamara
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[Asterisk-Users] problem number analize

[Petr Grussmann]

call from  PBX with analog FXS line to ISDN PRI T100P
if I use number analize exten = 452., dial call not working becouse 
Asterisk get connect to analog line and analog line not proclaim all 
number for call
if I useexten = 452XXX, Dial call working after pres on 
analog phone all number define in XXX

this problems not if I use digital Phone (this phone sending all number)
If you know how resolve this problem send mail thankx
Best regards,
Petr Grussmann
technical director
Opavanet a.s.
Czech republic
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[Asterisk-Users] Asterisk command

[GIBERT Frédéric]

Hello,

I would like to know if someone gets a doc which resumes what changes need
a reload and what changes need a restart of asterisk.

Thanks.



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[Asterisk-Users] Poopy errors on quad wcfxo

[Matteo Brancaleoni]

Hi all,

I'm experiencing problems with the TDM card
with 4 fxo modules. on all tests,
if the cards has 4 modules, I get
poopy kernel messages on the card.
The card works for sometime,then hangs
and a asterisk restart must be done,
along with kern modules unload/reload .

if I remove the first module, the card
works without problems at all on the
remaining 3 modules.

using latest zaptel cvs.

anyone is experiencing that or have
a workaround ?

thanks a lot,

Matteo

-- 
Matteo Brancaleoni [EMAIL PROTECTED]
Espia - Emmegi Srl

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RE: [Asterisk-Users] oh323

[T. Chan]

Jeremy,

Yes, I felt that it was important to report my trouble and I did it three
times, reporting to the asterisk community, but for some reasons, I was not
being responded to at all. I thought my messages were embedded among the
hundreds of them and were missed out or everyone was having the same problem
and was not able to help.

Jeremy, I have followed all instructions of yours by compiling the correct
verson of pwlib and openh323 (by doing make clean opt under each directory),
I have then gone into H323 and done a 'make' before going back to
/usr/src/asterisk to do a 'make install'. I tried using sjphone, I tried
using another asterisk, I tried using cisco to call into it, but I just was
not able to get any audio at all, when using the old version, I was able to
do so no problem with all the equipment above.

Jeremy, I don't know if there is any change on the h323.conf or any other
file that I need to do, please let me know, because I have not changed any
configuration files.

Thanks

TC

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jeremy
McNamara
Sent: Friday, June 18, 2004 3:57 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] oh323


T. Chan wrote:

 Jeremy

 I speak for myself, I have been testing with oh323 driver as well, because
 in my case, your h323 driver is not working, it was working before, but
then
 when I started to upgrade to 0.7.0 version of asterisk and from that point
 onwards (beginning of January), calls have had no audio. I tried making
 calls and I was getting no audio at all when the call was connected. Since
 then, I have not been able to upgrade the asterisk version, because if so,
I
 would not be able to run h323. That is why in my case, I have been trying
to
 explore the other alternative. If you have some idea to it, please let me
 know, thanks alot


So you didn't feel it was important to report your trouble anywhere?


I have tested the cvs -head of asterisk with many different types of
H.323 gateways and cannot make it fail.


Jeremy McNamara
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Re: [Asterisk-Users] Cost of IP Phones, or Isn't It Just Software?

[Chris Lee]

SNIP
On the other hand...  Go take a look at all of the ~$100 wireless 
router/firewall/print server/gateway boxes on the market, and you'll see 
one thing that almost all of them have in common: they all run Linux.  
Most of them are even based on the same small number of tools; things 
like busybox and uclibc.  If you want to see cheap, powerful VoIP 
phones, think about what they really need in terms of software, and then 
set out to write it and license it so the phone companies can 
incorporate it into their products.  I'm kind of amazed that FXS ports 
aren't standard on medium-end home routers right now; they'd probably 
only add $5-10 to the cost of the router, *IF* they had the software and 
felt like the demand was there.
My Draytek ADSL 2600v comes with two FXS ports
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RE: [Asterisk-Users] BT Caller ID - From Patch ? - Distinctive ring

[Kevin Walsh]

Kannaiyan Natesan [EMAIL PROTECTED] wrote:
 I got the dring value from the following call log.
 
 -- Detected ring pattern: 337,0,0
 
 Here is the configuration for my BT Line:
 
 usedistinctiveringdetection=yes
 dring1 = 367,0,0
 dring1context = default
 dring2 = 337,0,0
 dring2context = business  ; this matches the second phone number alloted by BT. 
 
My dring1/2 settings are different (BT too, by the way):

dring1 = 367,0,0
dring2 = 247,0,0

You got your settings from the right place (the log).  Perhaps BT vary
the distinctive ring cadence depending the exchange to which you're
connected, and the equipment used in that exchange.  It seems weird
that they don't (appear to) have a standard.

-- 
   _/   _/  _/_/_/_/  _/_/  _/_/_/  _/_/
  _/_/_/   _/_/  _/_/_/_/_/  _/   K e v i n   W a l s h
 _/ _/_/  _/ _/ _/_/  _/_/[EMAIL PROTECTED]
_/   _/  _/_/_/_/  _/_/_/_/  _/_/

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[Asterisk-Users] Problems reciving fax with Asterisk

[Michael Løjtnant]

Hi,

I am trying to recice a fax with * using SpanDSP - but it doesn't create the output 
file. (See the bottom of log file).
* Loads both app_rxfax.so and app_txfax.so fine.
Also I can't make * autodetect an incomming fax call (yes I have enabled 
faxdetect=both in zapata.conf - though it's not a Zap device)

Any ideas are welcome :-)

Best Regards
 Michael Løjtnant

System Details:

Lastest * CVS-HEAD
Libtiff-3.5.7
Linux-2.6.6 kernel
Fritz AVM ISDN Card

extension.conf:

[incomming]

exten = s,1,Answer
exten = s,2,rxfax(/tmp/minfax.tif)
exten = s,3,hangup

Console Log:

-- Executing Answer(Modem[i4l]/ttyI0, ) in new stack
-- Executing RxFAX(Modem[i4l]/ttyI0, /tmp/minfax.tif) in new stack
Changed from phase 0 to 1
Slow carrier up
Slow carrier down
Start receiving document
Changed from phase 1 to 4
Sending ident
 CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20
DIS:
Preferred octets: 256
Can receive fax
Supported data signalling rates: V.27ter and 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
Minimum scan line time for higher resolutions: T15.4 = T7.7
 DIS: 80 00 ce f0 80 80 01
HDLC underflow in state 9
Changed from phase 4 to 3
T4 timeout in state 9
Changed from phase 3 to 4
Sending ident
 CSI: 40 08 14 ce 18 f0 7a 15 af 14 c6 ef 10 e7 61 0b 44 1b 20 fb 59
DIS:
Preferred octets: 256
Can receive fax
Supported data signalling rates: V.27ter and 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
Minimum scan line time for higher resolutions: T15.4 = T7.7
 DIS: 80 00 ce f0 80 80 01
T2 timeout
Start receiving document
Sending ident
 CSI: 40 08 14 ce 18 1b 38 0e a4 ec af e6 e4 06 02 1c 21 09 a9 e9 0c
DIS:
Preferred octets: 256
Can receive fax
Supported data signalling rates: V.27ter and 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
Minimum scan line time for higher resolutions: T15.4 = T7.7
 DIS: 80 00 ce f0 80 80 01
HDLC underflow in state 9
Changed from phase 4 to 3
T4 timeout in state 9
Changed from phase 3 to 4
Sending ident
 CSI: 40 84 1c 0a 06 ae e5 65 f3 2b 17 ba 12 49 ec af e9 40 0e 0c 19
DIS:
Preferred octets: 256
Can receive fax
Supported data signalling rates: V.27ter and 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
Minimum scan line time for higher resolutions: T15.4 = T7.7
 DIS: 80 00 ce f0 80 80 01
T2 timeout
Start receiving document
Sending ident
 CSI: 40 08 14 ce 18 19 b4 12 49 ef e8 e5 2c 01 8e 1b 78 0d bf eb b3
DIS:
Preferred octets: 256
Can receive fax
Supported data signalling rates: V.27ter and 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
Minimum scan line time for higher resolutions: T15.4 = T7.7
 DIS: 80 00 ce f0 80 80 01
HDLC underflow in state 9
Changed from phase 4 to 3
T2 timeout
Start receiving document
Changed from phase 3 to 4
Sending ident
 CSI: 40 08 14 ce 18 01 8e 1b 78 0d bf eb b3 e7 61 07 04 1c 2e 08 ae
DIS:
Preferred octets: 256
Can receive fax
Supported data signalling rates: V.27ter and 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
Minimum scan line time for higher resolutions: T15.4 = T7.7
 DIS: 80 00 ce f0 80 80 01
T4 timeout in state 9
Sending ident
 CSI: 40 08 14 ce 18 ec f0 12 8c 17 58 f2 dc e5 83 07 04 1c 01 ff 7b
DIS:
Preferred octets: 256
Can receive fax
Supported data signalling rates: V.27ter and 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
Minimum scan line time for higher resolutions: T15.4 = T7.7
 DIS: 80 00 ce f0 80 80 01
HDLC underflow in state 9
Changed from phase 4 to 3
T2 timeout
Start receiving document
Changed from phase 3 to 4
Sending ident
 CSI: 40 08 14 ce 18 18 9e f8 fb e3 d1 f7 51 17 ba 15 76 f3 c9 e4 22
DIS:
Preferred octets: 256
Can receive fax
Supported data signalling rates: V.27ter and 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
Minimum scan line time for higher resolutions: T15.4 = T7.7
 DIS: 80 00 ce f0 80 

Re: [Asterisk-Users] Draytek Vigor 2600Vi as SIP client on Asterisk

[Chris Lee]

Michael Hamann wrote:
Hi Everybody,
as a relative newby I´m just trying to get a Draytek Vigor Router (2600Vi)
connected to my Asterisk System (CVS-05/31/04). With X-Lite and a Cisco
Phone it is no problem, but the Vigor seems to have some problems with
Asterisk.
The first thing ist when I do a sip show peers on the console I get:
4002/4002172.16.183.37   (D)  255.255.255.255  5060 Unmonitored
4001/4001172.16.183.37   (D)  255.255.255.255  5060 Unmonitored
What does this status unmonitored mean? With my softphone the entry looks
like:
6275/6275172.16.181.49   (D)  255.255.255.255  5060 OK (8 ms)
The next thing is that when I try to call one of the vigors SIP Ports via
X-Lite I see the following message in the debug console:
Jun 18 10:09:54 NOTICE[131081]: chan_sip.c:5150 handle_response: Dunno
anything about a 0 Unkown status code response from SIP/4001-b2fc
No call is signalled to the phone. The other way, my X-Lite rings but the
connection is hung up the moment I accept the call.
The Draytek support says that the Vigor does not support SIP Reinvite and
that I should try to disable it in my PBX system.
So I changed my sip.conf to:
[4001]
type=friend
username=4001
secret=4001
mailbox=2000
canreinvite=no
context=default
host=dynamic
But it still does not work. Does anybody has this combination working and
could send me his config files? Or any other ideas?
best regards from germany
Michael
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I had this working once, now I have a grandstream so it is no longer needed.
It is vital that you get the latest version of the firmware for the 
vigor as previous versions do not work with the sip server on the lan 
ports only on the other side of the ADSL line.
The reason for this is the sip packets always originated from the ADSL 
address instead of the internal address which is the one you want to be 
using if you have an internal server.
Next I used a settup a bit like this:
Vigor:
	VOIP SETUP  SIP Related Functions
	SIP:
	SIP Port 5060
	Registrar asterisk.mydomain.com (or an IP address)
	Port1:
	Name: p1
	Password:  (I did not use one)
	Expiry Time: 10 mins
	
	VOIP Setuip  CODEC/RTP etc:
	Codecs:
	G.711MU
	Packet Size: 20ms
	DTMF:
	OutBand
	Payload Type 101
	RTP:
	Take the default ports

Asterisk:
Sip.conf:
[general]
port=5060   ; Port to bind to
bindaddr=0.0.0.0; Address to bind to
context=in-sip  ; Default for incoming calls
callerid=Call 909090
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm
maxexpirey=1800
defaultexpirey=600
tos=throughput
[p1]
type=friend
host=dynamic
user=p1
;secret=
dtmfmode=rfc2833
[EMAIL PROTECTED]
callerid=p1 3002
qualify=yes
context=home
hope this helps
Chris.
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RE: [Asterisk-Users] Compiling problem on Debian

[Lars Boegild Thomsen]

  I can't compile Asterisk on a Debian machine.
  What is wrong? :/
 debian... :-(
 I was only able to compile asterisk when I gave up on doing it by myself
 and decided to use the debian package (.deb).

I've got Asterisk CVS running on at least 8 Debian machines - most current
at Testing level - a few current at Unstable level.  Asterisk compiled
cleanly on each system.  I remember there was a few package requirements
that was not default in Debian, but I am not entirely certain which ones it
was.  Check the error message and look for a package with that name + -dev
to get the headeres and libraries for compiling.  Anyway - with the right
packages Asterisk (and everything else) compiles cleanly on Debian.
Contrary to a few other distributions Debian only installs the stuff you
really need - which is the greatest benefit of that distribution.

Regards,

Lars...

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[Asterisk-Users] Asterisk and CISCO Gateway

[Martin Gebhard ( A+G connect GmbH )]

Hello

I have the following structure

  SIPH323 (chan_h323)
SIP Phone  Asterisk/H323 
---
   
  CISCO Gateway (CISCO 2610/NM2V-VIC-2BRI) - ISDN
SCCP Phone  CISCO CCM V3.3 
--
SCCPH323

I have the following problem:

Call from SIP to SCCP and from SCCP to SIP over H323 works fine. When I phone from SIP 
to an ISDN Phone (extern) the call is received but no voice is available after pickup. 
The Asterisk Server works as h323 Gateway. The Trace shows that packages are sendet 
from the SIP - Phone to Asterisk and from the CISCO Gateway to Asterisk. But the 
Asterisk doesn't pass the rtp - packages in both directions (not to the Phone and not 
to the Gateway).

Can anyone help ?

Thanks Martin Gebhard


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[Asterisk-Users] Asterisk does not start when cdr_odbc ist configured

[Thomas Frölich]

Hi,

i want to load the cdr into oracle using unixODBC.
I'm using RH 9 2.4.20-30.9smp, unixODBC 2.2.6, easysoft odbc driver for oracle 1.3.1.

My unixODBC is working well. 
With isql i can connect to the database, do selects, inserts and so on.
I created the table cdr as described on the asterisk wiki site.

When i configure the cdr_odbc.conf with the needed values, then * does not start any 
more.

My cdr_odbc.conf:
[global]
dsn=oracle
username=asterisk
password=asterisk
loguniqueid=on

odbc.ini
[ORACLE]
Driver  = ORACLE
Database= db9i
Servername  = 192.168.0.94
Port= 1523
User= asterisk
Password= asterisk
METADATA_ID = 0
ENABLE_USER_CATALOG = 1
ENABLE_SYNONYMS = 1

odbcinst.ini
[ORACLE]
Description = Easysoft ODBC Oracle Driver
Driver  = /usr/local/easysoft/oracle/libesoracle.so
Setup   = /usr/local/easysoft/oracle/libesoraclesetup.so
FileUsage   = 3

When i try to set autoload=no in modules.conf and then load the module cdr_odbc.so 
with cli this happens:
asterisk -vvr
  == Parsing '/etc/asterisk/asterisk.conf': Found
Asterisk CVS-NHEAD-06/09/04-16:25:34, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer [EMAIL PROTECTED]
==
Connected to Asterisk CVS-NHEAD-06/09/04-16:25:34 currently running on ospbx1 (pid = 
1033)
ospbx1*CLI load cdr_odbc.so
ospbx1*CLI
Disconnected from Asterisk server
Executing last minute cleanups



Any help would be appreciated.

Tom

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[Asterisk-Users] bri-stuff with current CVS head

[Julian Pawlowski]

Hi everybody,

any hints when the next version of bri-stuff will be released so that it
will work with the current CVS head? (Klaus-Peter? ;-) )


Regards

Julian Pawlowski
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Re: [Asterisk-Users] IAXy and bandwidth requirements

[Michael George]

On Jun 17, 2004, at 10:18 PM, Brian K. West wrote:
g726 is 16,24,32 and 48k asterisk only does g726-32k.  The iaxy 
doesn't do
g726 it does ADPCM as g726 is too complex for the iaxy to do.

So in this case g711ulaw/alaw is all you have to choose from.
Okay, that's what it looked like.  So the IAXy is intended for an 
internal network installation, not to be used as a VOIP by itself.

Thanks for the informaion/confirmation!
- Original Message -
From: Michael George [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, June 17, 2004 7:43 PM
Subject: [Asterisk-Users] IAXy and bandwidth requirements

In the mailing list archives, I found a message that indicates that 
the
IAXy has the ulaw, alaw, and g726 codecs, but I cannot find anything
official on Digium's site about it.  The Installation Manual has an
example iax.conf file that indicates the ulaw codec, so I know that 
one
is good.

But we are thinking about using the IAXy over a VPN, to replace our
MultiVoip.  alaw and ulaw are 64kbps, which is too much for our 
tunnel.
  G726 would be good if is has decent sound quality at 16kbps.  We
currently use MultiVoip's Netcoder at 9.6kbps which works fine.

So, is there an official statement somewhere about which codecs the
IAXy supports?
Thanks!
-Michael
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-Michael
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Re: [Asterisk-Users] festival with asterisk problem

[Michael George]

On Jun 16, 2004, at 4:05 PM, Michael George wrote:
Following the installation directions on the wiki, I got festival 
built and installed.  However, when I hit it from my dialplan, I get:

Feature Token_Method not defined
I found only one reference to this error message in the archives and 
there was no solution...
Once again I find that I didn't follow the directions...
Since I found this question in the archives with no answer and since I 
asked it again, I thought I'd answer myself so that the next person who 
runs into it can have it solved very quickly...

The problem I had was that I FTP'd the festvox tarball, but I didn't 
unpack it.  So there was no default voice there for festival to use.  
Unpacking the festvox and the other non-source code files solved the 
problem right away.

-Michael
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Re: [Asterisk-Users] Maximum retries exceeded on call

[Eric C. Snowdeal III]

Holger Schurig wrote:
i'm new to asterisk and am having trouble placing outbound calls.  i
   

Bug Grandstream so that they finally fix their buggy software.
The GS phone sends occassional SIP packets to port 0, not to port 5060, as 
tcpdump or (better) ethereal will show you.

There's a page on this at voip-info.org.
 

thanks for the heads-up about grandstream, but as i stated in the 
original message, i'm using xten lite softphones.   hopefully this is 
the approproriate forum for this question; i believe this is not an xten 
configuration issue because i can connect to a ser/rtproxy/nathelper 
server without problems and i can connect directly to a voicepulse 
account, which leads me to believe that this is an * configuration 
problem on my part.  less likely, i suppose, is the chance that * isn't 
as robust in handling nat than ser or whatever voicepulse is running.

given the configuration files that i posted in the original message, are 
there any changes that i should make?  certainly the asterisk faq makes 
the solution seems straighforward [1]:

Most likely you have a SIP client behind NAT that is trying to 
communicate with Asterisk without having the nat=yes setting in place 
in sip.conf. Another cause for this could be related to a user device 
that has an sip entry but has been physically removed (switched off or 
LAN-disconnected).

but as my original message showed, i do have nat=yes in my sip.conf and 
i don't believe the latter scenario is true.

any help is greatly appreciated.
[1] http://www.voip-info.org/wiki-Asterisk+FAQ
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[Asterisk-Users] TE410P / Eicon PRI

[Michael Devenijn]

for today we only have experience with BRI applications together with asterisk.

is the following scenario possible and stable enough for production? 
FYI : We want to build a unified messaging application integrated with SIP.

We have an E1 connection in Belgium with 100 msn's

We would think about having 2 servers :
 Server A : Asterisk
  PRI card (Digium TE410P)

 Server B : Fax server
PRI card (Eicon PRI30M)



 Call --- TE410P/1 --- Asterisk Extension --- 

Voice ?  --- Voicemail or Dial  
Fax ?--- TE410P/2 crossover to  --- Server B (Eicon PRI) 


Michael


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intended recipient only. Any use of the information contained herein ( including, but 
not limited to, total or partial reproduction, communication or distribution in any 
form ) by persons other than the designated recipient(s) is prohibited.If an 
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Re: [Asterisk-Users] IAX Jitter Buffer

[Andrew Kohlsmith]

On Friday 18 June 2004 02:46, George Pajari wrote:
 (b) other times we would experience no audio in one direction for between 1
 and 4 seconds and then things would seem to work fine;

I just had this problem with my * setup:

KSU - Adit600 - T100P - IAX2(Office) - IAX2(Colo) - IAX2(Nufone)

The *Colo box never steps out of the way since *Office is not routeable to 
*Nufone (no NAT, but rather two network interfaces at *Colo, one going 
directly to *Office.

The Colo box also has a TE405P in it going to the telco PRI for local calls, 
but dropouts never occured on those calls; only on calls to Nufone.  I turned 
off jitter buffer and moved to the GSM codec at the request of Nufone's 
technical support department (and turned on IAX2 trunking, I had it disabled 
since calls between *Office and *Colo would exhibit bursty audio) and the 
problem went away.

So no, I don't think jitter buffer's quite there yet, although I *never* had 
that problem before this week.  Perhaps it's a recent CVS fix.  :-)

Regards,
Andrew
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Re: [Asterisk-Users] TE410P / Eicon PRI

[Klaus-Peter Junghanns]

 We would think about having 2 servers :
  Server A : Asterisk
   PRI card (Digium TE410P)
   
  Server B : Fax server
   PRI card (Eicon PRI30M)
 
 
 
  Call --- TE410P/1 --- Asterisk Extension --- 
 
 Voice ?  --- Voicemail or Dial  
 Fax ?--- TE410P/2 crossover to  --- Server B (Eicon PRI) 
 

save 10k EUR and use spandDSP (www.opencall.org) for fax instead of the
second server with the Eicon PRI card.

 
 Michael

best regards

Klaus
-- 
Klaus-Peter Junghanns

CEO, CTO
Junghanns.NET GmbH
Breite Strasse 13a - 12167 Berlin - Germany
fon: (de) +49 30 79705390
fon: (uk) +44 870 1244692
fax: (de) +49 30 79705391
iaxtel: 1-700-157-8753
http://www.Junghanns.NET/asterisk/


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RE: [Asterisk-Users] TE410P / Eicon PRI

[Michael Devenijn]

i'dd like to but is it stable enough for production (receiving over 500 faxes a day ?)

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Klaus-Peter
Junghanns
Sent: vrijdag 18 juni 2004 13:58
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] TE410P / Eicon PRI


 We would think about having 2 servers :
  Server A : Asterisk
   PRI card (Digium TE410P)
   
  Server B : Fax server
   PRI card (Eicon PRI30M)
 
 
 
  Call --- TE410P/1 --- Asterisk Extension --- 
 
 Voice ?  --- Voicemail or Dial  
 Fax ?--- TE410P/2 crossover to  --- Server B (Eicon PRI) 
 

save 10k EUR and use spandDSP (www.opencall.org) for fax instead of the
second server with the Eicon PRI card.

 
 Michael

best regards

Klaus
-- 
Klaus-Peter Junghanns

CEO, CTO
Junghanns.NET GmbH
Breite Strasse 13a - 12167 Berlin - Germany
fon: (de) +49 30 79705390
fon: (uk) +44 870 1244692
fax: (de) +49 30 79705391
iaxtel: 1-700-157-8753
http://www.Junghanns.NET/asterisk/


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Re: [Asterisk-Users] IAX Jitter Buffer

[Rich Adamson]

 When we enabled jitterbuffer the sound quality seemed to improve but we
 noticed some problems:
 
 (a) sometime we would get only one-way audio;
 (b) other times we would experience no audio in one direction for between 1
 and 4 seconds and then things would seem to work fine;
 (c) some times users reported a clipped and almost half duplex sound
 quality as the flow of the conversation shifted back and forth.
 
 We also noticed some wingnut values for Lag and Jitter such as:
 Lag: -65476ms
 Jitter: 12897799ms
 
 PSTN gateway is CVS-04/20/04-01:11:29 
 Client machine is CVS-HEAD-06/02/04-07:56:41
 
 Searching the Asterisk bug lists shows some significant fixes (1696, 1643).
 
 Q2: Is jitterbuf working well enough to try again?
 
 Q3: Any other suggestions for improving voice quality with IAX links?

A google search of the asterisk-cvs list indicates there has been several
iax changes in the last several months. Iax2 with gsm is working very well
between * systems using the current cvs Head.

I was told specifically by Mark to include jitterbuffer=no in the iax.conf,
but with no explanation as to why. Although I'm not a programmer, causual
browsing of the source code would seem to suggest that some sort of 
dynamic jitter buffer function is in use and attempts to over-ride it
might not be a reasonable thing to do. 

I'd suggest bumping both systems up to current cvs Head, add the statement,
and eval the result.



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Re: [Asterisk-Users] IAX Jitter Buffer

[steve]

On Fri, 18 Jun 2004, Rich Adamson wrote:

 A google search of the asterisk-cvs list indicates there has been several
 iax changes in the last several months. Iax2 with gsm is working very well
 between * systems using the current cvs Head.
 
 I was told specifically by Mark to include jitterbuffer=no in the iax.conf,
 but with no explanation as to why. Although I'm not a programmer, causual
 browsing of the source code would seem to suggest that some sort of 
 dynamic jitter buffer function is in use and attempts to over-ride it
 might not be a reasonable thing to do. 
 
 I'd suggest bumping both systems up to current cvs Head, add the statement,
 and eval the result.

jitterbuffer=no turns off that dynamic jitter buffer function.

People recommend to turn that off because it doesn't work 100% at the 
moment.

Steve

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[Asterisk-Users] FXO Issues

[Greg Scasny]

All,



Experiencing some issues on my FXO lines. If a call comes in
on an FXO and then get transferred to another FXO (say to call someones cell phone), those two lines will stay tied
together indefinitely. This happens to us when we transfer an incoming call to
our on call guys after hours and on weekends. We have installed 3 other * boxes
and they do the same thing. 



We use a Adit Channel bank for all
incoming FXO and the other installs use multiple 1 port digium
FXO cards or a combination of the 1 port and 4 port FXO cards.



Has anyone else experienced this and if so how did you fix
it?



Thanks in
advance..Greg





Gregory P. Scasny

Golden Technologies Inc.

http://www.golden-tech.com

219-462-7200

[Asterisk-Users] FXO Issues - Sorry

[Greg Scasny]

I just saw that one of our techs posted the same
question - I apologize for the multiple
posts (as I put my asbestos suit on J ).



Greg



Gregory P. Scasny

Golden Technologies Inc.

http://www.golden-tech.com

219-462-7200

RE: [Asterisk-Users] Zap dropping calls

[Greg Scasny]

Tim,

busydetect=yes
callprogress=yes

Set these to no and it should stop the random hang-ups.

Gregory P. Scasny

Golden Technologies Inc.

http://www.golden-tech.com

219-462-7200

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Schlie
Sent: Thursday, June 17, 2004 11:55 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Zap dropping calls

I'm running Asterisk CVS-HEAD-05/24/04-17:37:48 on kernel
2.4.25-gentoo-r3.  I have a Digium TDM-400P card with 4 FXO ports.  Here
are the pertinent files:

zaptel.conf:
fxsks=1-4
loadzone = us
defaultzone=us

zapata.conf:
[channels]
context=north_in_pots_vip
group=1
signalling=fxs_ks
usecallerid=no
hidecallerid=no
callwaiting=no
restrictcid=no
threewaycalling=no
echocancel=1
echocancelwhenbridged=no
echotraining=1
rxgain=10.0
txgain=2.0
immediate=no
musiconhold=default
jitterbuffers=4
relaxdtmf=yes
busydetect=yes
callprogress=yes
callerid = 1234567
channel = 1-2

extensions.conf:
[north_in_pots_vip]
exten = s,1,Answer
exten = s,2,Dial(SIP/JGARVEYSIP/FDNORTH,10,rt)
exten = s,3,Dial(SIP/JGARVEYSIP/KRAFFERTYSIP/FDNORTH,10,rt)
exten = s,4,Background(vip_autoattendant)
exten = s,5,Voicemail(u600)
exten = s,6,Hangup

When the SIP phones (Grandstream BudgeTone-100's) answer a call there is
a random chance that the call will get disconnected.  It also happends
on outbound calls.  Sometimes it's within the first minute, sometimes
it's after 10 minutes.  All I ever see on the * console is Hungup
'Zap/1-1'.

Any ideas?



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RE: [Asterisk-Users] Compiling problem on Debian

[Asterisk Developer]

I had a bit of a problem compiling CVS Asterisk on Debian-Woody, but
www.voip-info.org has a debian-specific page that lists the debian
packages you will need to apt-get:

http://voip-info.org/wiki-Linux+Debian

...after installing these, it compiled without a hitch!

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RE: [Asterisk-Users] Compiling problem on Debian

[Asterisk Developer]

Also, make sure you have the kernel-headers package that matches your
kernel-source package.

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RE: [Asterisk-Users] LDAP synchronization script

[Stefan de Konink]

The base problem, I presume is not that there is no documentation, but how
to combine all those defacto standards, from an user and an application
point of view.
An Active Directory implementation in Linux (for users and application)
for me starts with the standard PAM/NSS stuff but why not extend that for
Jabber, Asterisk, Postfix/Sendmail, DHCPd, DNS and a zillion other stuff
like (a higher level) ENUM?

For most of the above application are 'dynamic' ldap backends made, which
are usable. Though what is the best thing to start with? Application with
users under it. Users with Application under it. Or the last type I think
it is the most usuable way of implementing:

Organisation/
Groups/
Applications(Group/Application Specific configs)
Users/
Applications(User/Application Specific configs)
Applications(Organisation Specific configs)

Applications (Basic configuration)
Name/   (Name like Asterisk)
ID/ (Which Asterisk server IP address etc.)


Which makes .application and /etc/application obsolete if well
implemented. Performance wise you would not want to poll the LDAP server
24/7 (though I want it ;) but only fetch while reloading.

In the combination and integration of those things I'm now writing a
thesis with a production proof-proof of concept, for Unified Messaging in
a Box. Though, importing all schema's like cosine, dhcpd, etc. the mess
only gets bigger eq. there need to be a basic structure and I would
like to have some feedback about it.

The main objective is to make the user have a 'home' peer/server, though
it doesn't depends on this peer but it is like 'the first choice'. For
example two Asterisk servers, one crashes the other peer/server takes over
and starts accepting the other servers its users.

Ok, this basically implies there is a distributed filesystem around, at
the moment I use CodaFS for that. (Requires patching of some programs like
Postfix)


Stefan


On Fri, 18 Jun 2004, Lars Boegild Thomsen wrote:

 Hi,

  The what belongs were is my big question at the moment and I personally
  don't want to design anything LDAP-ish that would become my private tree
  instead of defacto implementation.

 You should definitely have a look at the defacto standards for storing users
 and groups (check http://www.padl.com/OSS/pam_ldap.html).  Would be rather
 cool to have a Linux network with users and groups defined in LDAP - and
 each user just having an extension defined in his record.  Asterisk base
 configuration should go in separate three.

 Regards,

   Lars...

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RE: [Asterisk-Users] TE410P / Eicon PRI

[Klaus-Peter Junghanns]

 i'dd like to but is it stable enough for production (receiving over 500 faxes a day 
 ?)

i think it is. at least i know someone who is using it in production on
a Digium E1 card.

If everything else fails you can buy that eicon card later on in the
worst case.

best regards

Klaus
-- 
Klaus-Peter Junghanns

CEO, CTO
Junghanns.NET GmbH
Breite Strasse 13a - 12167 Berlin - Germany
fon: (de) +49 30 79705390
fon: (uk) +44 870 1244692
fax: (de) +49 30 79705391
iaxtel: 1-700-157-8753
http://www.Junghanns.NET/asterisk/


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RE: [Asterisk-Users] Integration with SIEMENS HIPATH PBX

[ePyron Felix Deierlein]

Hi,

you can integrate it via PRI or BRI.

Regards


Felix




From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ronaldo
Sent: Friday, June 11, 2004 7:04 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Integration with SIEMENS HIPATH PBX


Hi,
 
 
I would like to know if Asterisk is able to be integrated with a
Siemens HIPATH PBX by VoIP or other ways.
 
Best regards,
 
Ronaldo S. Pereira
PRI Telemática.
 
 


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Re: [Asterisk-Users] FXO Issues

[Rich Adamson]

 Experiencing some issues on my FXO lines. If a call comes in on an FXO and then get 
transferred to another FXO (say to call someones cell
 phone), those two lines will stay tied together indefinitely. This happens to us 
 when we 
transfer an incoming call to our on call guys after
 hours and on weekends. We have installed 3 other * boxes and they do the same thing.
 
 We use a Adit Channel bank for all incoming FXO and the other installs use multiple 
 1 port 
digium FXO cards or a combination of the 1 port
 and 4 port FXO cards.
 
 Has anyone else experienced this and if so how did you fix it?

If you're using plan old pstn analog lines, put a voltmeter on the
analog line to see if you have call supervision coming from the telco.
You should see the voltmeter either dip to zero volts for about a
half second, or, voltage reverses for some short period of time.
That should occur within a few seconds after the caller hangs up.
Exactly when that occurs varys by central office switch manufacturer.

If you don't see any form of supervision, then you have to implement
some sort of timer, tone detection, etc. You can also talk to your
telco engineering/tech folks to see they have any options for call
supervision. (The sales office won't have a clue in most cases.)

Other choices might include changing your pstn lines from loop start
to EM signaling, etc.


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RE: [Asterisk-Users] Compiling problem on Debian

[Martin List-Petersen]

or a lot easier:

Pull the patch i use for my cvs snapshot Debian packages:
http://loke.home.marlow.dk/dists/sid/asterisk/patches/01-debian-marlow.diff

Apply it to latest cvs.
chmod +x debian/rules

And compile.

Have fun.

Kind regards,
Martin List-Petersen
martin (at) list (dash) petersen (dot) net

On Fri, 2004-06-18 at 14:05, Asterisk Developer wrote:
 I had a bit of a problem compiling CVS Asterisk on Debian-Woody, but
 www.voip-info.org has a debian-specific page that lists the debian
 packages you will need to apt-get:
 
 http://voip-info.org/wiki-Linux+Debian
 
 ...after installing these, it compiled without a hitch!
 
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Re: [Asterisk-Users] anyone use mailboxexists?

[Michael George]

Billy, looking at this more closely, I have some questions...
On Jun 15, 2004, at 9:45 PM, Billy Huddleston wrote:
Yes, I use it. Here's a sample extension of how to use it.
exten = 1234,1,Answer()
exten = 1234,2,MailboxExists(1234)
exten = 1234,3,Dial(SIP/1234,20) ; Try to ring for 20 seconds, no 
answer goto voicemail
From the docs, it looks like MailboxExists() will add 101 to the 
priority if the box *does* exist and goes to the next priority if not.

exten = 1234,4,Voicemail(b1234) ; send to voicemail if busy
Here's your next priority step and you go to VM.  However, if ME() gets 
to here, it seems that the box does not exist.

exten = 1234,103,Dial(SIP/1234) ; Try to ring till answered
Here is priority +101, you try to dial the line...
exten = 1234,104,Busy() ; Give busy tone if busy.
... and give the busy tone if the line is busy...
exten = 1234,204,Voicemail(u1234) ; send to voicemail if no answer
... but if there's no answer you go to voicemail.
It looks like you go to voicemail either way, so perhaps I'm 
misunderstanding how MailboxExists() works...

Thanks!
-Michael
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Re: [Asterisk-Users] embedded Asterisk

[Martin List-Petersen]

On Thu, 2004-06-17 at 09:11, Klaus-Peter Junghanns wrote:
 Hi,
 
  Actually, you the Geode CPU mentioned below is a 5x86 (486 platform) at
  233 MHz. If you take Pebble (http://www.nycwireless.net/pebble/), which
  is a downstripped Debian ( 64 MB) on a readonly ext2 filesystem, you
  should be grand. Installing asterisk + some extra stuff will probably
  require, that you have at least a 128MB or 256MB flash or so.
 
 Dont go for stripped down but complete distributions which include a
 lot of stuff that you dont need, e.g. gcc. Go for a rescue system, like
 i used the SuSE rescue system (14 mb), then you can add what you need
 (sshd,...) and compile asterisk on another box and then just copy it.
 My compressed ramdisk image is 32 mb, including all voice prompts and
 some mp3s for MOH.

The good thing about this stripped down image is that it's still
upgradable as regular (apt-get) and has the script that then removes
uneccessary documents etc.

It is a matter of convienience.

If it was a matter of space i probably go for a uclib/busybox from
scratch solution.

The rescue cd's often also contain much that you don't need.

Martin List-Petersen
martin (at) list (dash) petersen (dot) net


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[Asterisk-Users] Problems with X100P

[Adam Lewis]

All,

I'm having trouble getting the X100P working. 


Lsmod shows :

zaptel179808   0

I did a .

# modprobe zaptel

and here is my zaptel.conf (comments omitted)

__SNIP__

fxsks=1
loadzone = us
defaultzone=us

__SNIP__

Here is zapata.conf

__SNIP__

[trunkgroups]
[channels]
context=default
switchtype=national
signalling=fxo_ls
rxwink=300
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
group=1
callgroup=1
pickupgroup=1
immediate=no
context=sip
signalling=fxs_ks
callerid=Phone 1
channel=1

__SNIP__

ztcfg -vv gives the following output..

__SNIP__

Zaptel Configuration
==


Channel map:

Channel 01: FXS Kewlstart (Default) (Slaves: 01)

1 channels configured.

ZT_CHANCONFIG failed on channel 1: No such device or address (6)

__SNIP__

Any ideas,

Thanks,

Adam

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Re: [Asterisk-Users] Problems with X100P

[Martin List-Petersen]

On Fri, 2004-06-18 at 14:57, Adam Lewis wrote:
 All,
 
 I'm having trouble getting the X100P working. 
 
 
 Lsmod shows :
 
 zaptel179808   0
 
 I did a .
 
 # modprobe zaptel
 
 and here is my zaptel.conf (comments omitted)
 
 __SNIP__
 
 fxsks=1
 loadzone = us
 defaultzone=us
 
 __SNIP__
 
 Here is zapata.conf
 
 __SNIP__
 
 [trunkgroups]
 [channels]
 context=default
 switchtype=national
 signalling=fxo_ls

Problem is here: signalling for fxo cars is fxs_ls

Kind regards,
Martin List-Petersen
martin (at) list (dash) petersen (dot) net


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RE: [Asterisk-Users] Problems with X100P

[David J Carter]

Don't you need a 'modprobe wcfxs' also?

Dave

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Adam Lewis
Sent: 18 June 2004 14:57
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Problems with X100P


All,

I'm having trouble getting the X100P working. 


Lsmod shows :

zaptel179808   0

I did a .

# modprobe zaptel

and here is my zaptel.conf (comments omitted)

__SNIP__

fxsks=1
loadzone = us
defaultzone=us

__SNIP__

Here is zapata.conf

__SNIP__

[trunkgroups]
[channels]
context=default
switchtype=national
signalling=fxo_ls
rxwink=300
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
group=1
callgroup=1
pickupgroup=1
immediate=no
context=sip
signalling=fxs_ks
callerid=Phone 1
channel=1

__SNIP__

ztcfg -vv gives the following output..

__SNIP__

Zaptel Configuration
==


Channel map:

Channel 01: FXS Kewlstart (Default) (Slaves: 01)

1 channels configured.

ZT_CHANCONFIG failed on channel 1: No such device or address (6)

__SNIP__

Any ideas,

Thanks,

Adam

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[Asterisk-Users] Hwo to get CallerID: SIP - ISDN

[Bernie Hoeneisen]

Hi!

I trying to configure * in a way, that it uses a different CLIP (Caller-Id
in ISDN) per SIP user, when relaying the call from SIP to the ISDN. So far
always the main (1st) number of the number-block is sent to the ISDN.

I have a E100P from Digium and use the zapata stuff (chan_zap).
All SIP calls are coming through an SER.

One idea I had in mind is to assign userid's in SIP, that match the
extension of the number block, e.g. 854. * could then take
the user part of the From header field of the incoming SIP INVITE and
relay this numeric user part (e.g. 854) to the chan_zap, so that the
CLIP in the ISDN appears as the number assigned to SIP user.

Another idea I had was ENUM. But as in ENUM one can only resolve one way,
i.e. E.164-number - SIP address, *  would have to lookup the whole
number block (every entry) from time to time and cache it in a mapping
table. No so nice solution, I guess.

Does anybody have some experience in this?
Any hints, instructions and HowTo's are warmly welcome.

cheers,
 Bernie





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RE: [Asterisk-Users] Problems with X100P

[Adam Lewis]

That did it.  Thanks!

Adam

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason Williams
Sent: Friday, June 18, 2004 10:08 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Problems with X100P

At 09:57 18/06/2004 -0400, you wrote:

I did a .

# modprobe zaptel

You need to carry out the following commands in this order

modprobe zaptel
modprobe wcfxo
ztcfg


without the modprobe wcfxo it will not work



Jason 

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[Asterisk-Users] ATT CallVantage Asterisk

[Kubat, Philip]

I am trying to connect directly to ATT VoIP service
CallVanage. I have ATTs ATA (D-Link DVG-1120M). They use mgcp. I have
traces of the connects from the Dlink and hoping to setup Asterisk the same.
It looks like I need to have Asterisk be a MGCP endpoint (gateway). How do I
configure this? Does the mgcp.conf support register like sip
etc? What is the syntax?



Thanks!

[Asterisk-Users] C7960 g729 question

[Rich Adamson]

I have multiple voiceage g729 licenses installed on a RH9 box, and have
a remote C7960 configured to use it (low bandwidth). In calls like:

  Remote C7960 - g729 - asterisk - g711 - C7960

the audio is oftentimes rather choppy. Changing the remote 7960 to use
g711 seems to eliminate/reduce the choppyness. Any ideas on what might
be behind this?



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Re: [Asterisk-Users] Hwo to get CallerID: SIP - ISDN

[Martin List-Petersen]

On Fri, 2004-06-18 at 15:16, Bernie Hoeneisen wrote:
 Hi!
 
 I trying to configure * in a way, that it uses a different CLIP (Caller-Id
 in ISDN) per SIP user, when relaying the call from SIP to the ISDN. So far
 always the main (1st) number of the number-block is sent to the ISDN.
 
 I have a E100P from Digium and use the zapata stuff (chan_zap).
 All SIP calls are coming through an SER.

Have you tried just to use SetCallerID in * before you dispatch the call
to your ZAP channel ?

 One idea I had in mind is to assign userid's in SIP, that match the
 extension of the number block, e.g. 854. * could then take
 the user part of the From header field of the incoming SIP INVITE and
 relay this numeric user part (e.g. 854) to the chan_zap, so that the
 CLIP in the ISDN appears as the number assigned to SIP user.

You can also maintain a database (astdb etc.) which matches the
phoneno.'s against you SIP id's, but your suggestion is easier.
Maintainance free.

It depends a bit on what userbase you have for your SIP users. How much you manage 
them or if
they are created/maintained by third party.

Kind regards,
Martin List-Petersen
martin (at) list (dash) petersen (dot) net


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[Asterisk-Users] X100P in Switzerland

[Reto Stauss]

Hi

Does anybody if the X100P works in Switzerland? We can't get a line to PSTN.

When I run zttool it shows me always a red alert. I can make and receive calls with an
anlog phone plugged in the phone connector.

I've compiled and configured the card according to the wiki. Everything seemed to be 
ok.

Is there a way to debug this?

Regards
Reto





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Re: [Asterisk-Users] asterisk hardware selection question

[creslin]

On Thu, Jun 17, 2004 at 05:02:26PM -0500, Erick Perez wrote:
 10 analog extension using conventional phones (lets say Panasonic kx-ts3
 analog)
 4 analog lines coming from our telco
 
 So i will need 3 TDM40B (total 12 FXS and none FXO so i can have 2 extra FXS
 ports for future)
 and one TDM04B Quad FXO.
 
 Right?

Ideally, you probably don't want to have more than 2 (and at the max 3)
of the TDM cards in your system.  Each of the TDM cards should have it's
own interrupt in your system.  With (on most PCI buses) only 4
interrupts available to the PCI bus, it's extremely difficult to get 4
cards working well on a single system.

 
 and what is the Asterisk support for Digital phones?
 

SIP, H.323, MGCP, and (of course) IAX :-) (you can find more info about
all this on the various websites http://www.asterisk.org,
http://www.voip-info.org/).

I prefer SIP or IAX :-).  The rest are a little bit more interesting to
set up.

Matthew Fredrickson
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RE: [Asterisk-Users] anyone use mailboxexists?

[Jeremy Jones]

Michael,

  From the docs, it looks like MailboxExists() will add 101 to the 
 priority if the box *does* exist and goes to the next priority if not.

I think the show application mailboxexists documentation is wrong.  I
believe it's the other way around.  It does exits? Jump to next
priority.  It doesn't?  Jump to n+101.  Here's my extension macro (sift
out the forwarding stuff if you don't like that), and it works:

[macro-stdexten]
exten=s,1,MailboxExists(${MACRO_EXTEN:[EMAIL PROTECTED]);If
mailbox exists continue at 2, otherwise goto 102
exten=s,2,NoOp  ;Filler
exten=s,3,NoOp  ;Filler
exten=s,4,NoOp  ;Filler
exten=s,5,NoOp  ;Filler
exten=s,6,DBget(temp=CFIM/${ARG1})  ;Get
CFIM key, if not existing, goto 107
exten=s,7,Dial(${TRUNK}/9${temp})
;Unconditional forward
exten=s,8,NoOp  ;Filler
exten=s,9,Dial(${ARG2},25,rtT)  ;Dial device for
25 seconds, goto 10 if busy, goto 110 if unavailable
exten=s,10,NoOp ;Filler
exten=s,11,DBget(temp=CFBS/${ARG1}) ;Get
CFBS key, if not existing, goto 112
exten=s,12,Dial(${TRUNK}/9${temp})  ;Forward
on busy or unavailable
exten=s,102,DBget(temp=CFIM/${ARG1});Get CFIM key,
if not existing, goto 203
exten=s,103,Dial(${TRUNK}/9${temp})
;Unconditional forward
exten=s,104,Dial(${ARG2},120,rtT)   ;Dial
device for 120 seconds, goto 105 if busy, goto 205 if unavailable
exten=s,105,DBget(temp=CFBS/${ARG1});Get CFBS key,
if not existing, goto 206
exten=s,106,Dial(${TRUNK}/9${temp}) ;Forward
on busy or unavailable
exten=s,107,Goto(s,9)   ;Goto 9
exten=s,110,Voicemail(u${MACRO_EXTEN:[EMAIL PROTECTED]) ;To VM
if unavailable
exten=s,111,Hangup  ;Hang up
the channel when vm exits
exten=s,112,Voicemail(b${MACRO_EXTEN:[EMAIL PROTECTED]) ;To VM
if busy
exten=s,113,Hangup  ;Hang up
the channel when vm exits
exten=s,203,Goto(s,104) ;Goto
104 for accounts w/out vm
exten=s,205,Busy()  ;Busy
signal if busy  no vm
exten=s,206,Busy()  ;Busy
signal if no answer in 2 min  no vm

It's a little ugly w/all those NoOps, but I think I need those to get
the priorities right.

Jeremy Jones



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[Asterisk-Users] Re: SJphone regestration problem - Help!

[ruixun wu]

Gonzalo Gasca wrote:
 Create the profile
 And a new windows appears:
 Profile name
 File name
 Profile type Calls through SIP proxy 
 Then in SIP proxy, 
 click the sip proxy option
 enter the Ip address of the proxy domain port
 user domain
 and proxy for nat and also the port (5060)
 
 be sure u have the sip.conf file correct
 Otherwise try to reinstall it
 
 

Ty Purcell wrote:
 Edit the profile, and on the Initialization tab
and make sure the Inquired
 box is checked by the fields you listed above. (Mine
also has all of the other
 boxes checked under saved and required.) 


Thank Gonzlo Gasca and Ty Purcell very much!  It does
work. I can enter 
the server IP address now and I can do further test
with Asterisk and 
SJphone. Thank for you help again.

Rui

__ 
Post your free ad now! http://personals.yahoo.ca
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Re: [Asterisk-Users] Need guides on setting up PDA on asterisk server

[Jason A. Pattie]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Holger Schurig wrote:
|I should follow this up to accurately state that audio was not
|operational in my test calls from the PDA.  I have patched the
|iaxclient library with the changes available from ZiaxPhone that word
|align the IAX2 library on the ARM platform.  I haven't finished
|compiling a new binary to test with.
|
|
| If you want to even patch ZiaxPhone, you can't: there's no source.
|
| There is something similar at my homepage,
| http://www.holgerschurig.de/qtiax.html. It doesn't yet run on my PDA) and
| lacks a config file support, but it's all source code.
He says he will release the sourcecode when he gets to a stable
working release.
Do you think your QtIAX client will run on a 206MHz StrongARM processor?
- --
Jason A. Pattie
[EMAIL PROTECTED]
Xperience, Inc. (http://www.xperienceinc.com)
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.4 (GNU/Linux)
Comment: Using GnuPG with Debian - http://enigmail.mozdev.org
iD8DBQFA0wK5uYsUrHkpYtARAlRxAJ4wLCEVox3OfxdQ21o4mapgCDBxLwCfcrha
vi4EUYRB+qe3PUWZa2UlnwU=
=lloy
-END PGP SIGNATURE-
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Re: [Asterisk-Users] TE410P / Eicon PRI

[Lee Howard]

If you would rather use HylaFAX instead of spandsp and have $10K to 
throw around, then may I suggest hiring an Asterisk channel author to 
write a T.38-supporting channel driver?  That way you could just use 
t38modem with HylaFAX, and you wouldn't need all the duplicate hardware.

Lee.
On 2004.06.18 05:17 Michael Devenijn wrote:
i'dd like to but is it stable enough for production (receiving over
500 faxes a day ?)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Klaus-Peter
Junghanns
Sent: vrijdag 18 juni 2004 13:58
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] TE410P / Eicon PRI

save 10k EUR and use spandDSP (www.opencall.org) for fax instead of
the
second server with the Eicon PRI card.

 Michael
best regards
Klaus
--
Klaus-Peter Junghanns
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[Asterisk-Users] Possible chan_skinny problems - no ringtone, no moh and no queue messages

[Steve Hanselman]

We're using Cisco phones running skinny protocol.



When I call other extensions I don't get a ringtone,
although the remote end does ring and when answered we get clear two way audio.

When I call a queue from a skinny phone then I don't
hear the announcements.

Likewise we don't hear music on hold on these phones,
although we can see mpg123 in the process list and ls -l the fd shows a pipe
open from asterisk to mpg123.



I created a dummy extension that played back the queue
message, called it from a skinny and it's fine.



Any ideas?



Also, I read somewhere that the two skinny implementations
(chan_sccp and chan_skinny) were going to be merged, any news on this?





Here's a skinny debug of a call being made to a queue:



Linux3*CLI skinny debug 

Skinny Debugging Enabled

 -- Starting simple switch on '[EMAIL PROTECTED]'

Collected digit: [6]

 -- Asked to indicate 'Stop tone'
condition on channel Skinny/[EMAIL PROTECTED]

Collected digit: [2]

 -- Asked to indicate 'Stop tone' condition
on channel Skinny/[EMAIL PROTECTED]

Collected digit: [2]

 -- Executing Queue(Skinny/[EMAIL PROTECTED],
Sales) in new stack

 -- Started music on hold, class 'random',
on Skinny/[EMAIL PROTECTED]

 -- Stopped music on hold on
Skinny/[EMAIL PROTECTED]

 -- Playing 'queue-youarenext' (language
'en')

 -- Told Skinny/[EMAIL PROTECTED] in Sales their
queue position (which was 1)

 -- Playing 'queue-thankyou' (language
'en')

 -- Started music on hold, class 'random',
on Skinny/[EMAIL PROTECTED]

Jun 18 16:08:21 NOTICE[63531]:
app_queue.c:668 wait_for_answer: No one is answering queue 'Sales'

Jun 18 16:08:26 NOTICE[63531]:
app_queue.c:668 wait_for_answer: No one is answering queue 'Sales'

Skinny [EMAIL PROTECTED] went on hook

 -- Stopped music on hold on
Skinny/[EMAIL PROTECTED]

 -- User disconnected when they almost
made it

 == Spawn extension (default, 622, 1) exited non-zero
on 'Skinny/[EMAIL PROTECTED]'

skinny_hangup(Skinny/[EMAIL PROTECTED]) on [EMAIL PROTECTED]



Here's the debug of the call to the dummy extension:



 -- Starting simple switch on '[EMAIL PROTECTED]'

Collected digit: [2]

 -- Asked to indicate 'Stop tone'
condition on channel Skinny/[EMAIL PROTECTED]

Collected digit: [0]

 -- Asked to indicate 'Stop tone' condition
on channel Skinny/[EMAIL PROTECTED]

Collected digit: [5]

 -- Executing Playback(Skinny/[EMAIL PROTECTED],
queue-thankyou) in new stack

skinny_answer(Skinny/[EMAIL PROTECTED]) on [EMAIL PROTECTED]

 -- Playing 'queue-thankyou' (language
'en')







Steve Hanselman

Brendata (UK) Ltd



Tel: +44 (0)1268 466111

Fax: +44 (0)870 1387283

Mob: +44 (0)7973 750993












The information contained in this email is intended for the personal and confidential useof the addressee only. It may also be privileged information. If you are not the intendedrecipient then you are hereby notified that you have received this document in error andthat any review, distribution or copying of this document is strictly prohibited. If you have received  this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) LtdNevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX  UKRegistered Office as above. Registered in England No. 2764339See our current vacancies at www.brendata.co.uk

[Asterisk-Users] Voicemail

[Joseph]

Which voicemail is current and latest?

Voicemail
  or
Voicemail2

I thot it was voicemail2 but this link sort of indicates otherwise...at
the bottome of the page it says:
  Old version:
   . Asterisk cmd VoiceMail2

http://www.voip-info.org/wiki-Asterisk+cmd+VoiceMail2#comments

-- 
respectfully, Joseph - (606) 477-2355 x140
   --=

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Re: [Asterisk-Users] TE410P / Eicon PRI

[Klaus-Peter Junghanns]

better send the EUR 10k (not $10k... :)  ) to the author of spandDSP.
Nobody needs HylaFAX for receiving faxes. Converting a tiff to pdf and
storing it somewhere is not rocket science. ;)

best regards

Klaus
 
Am Fr, 2004-06-18 um 17.08 schrieb Lee Howard:
 If you would rather use HylaFAX instead of spandsp and have $10K to 
 throw around, then may I suggest hiring an Asterisk channel author to 
 write a T.38-supporting channel driver?  That way you could just use 
 t38modem with HylaFAX, and you wouldn't need all the duplicate hardware.
 
 Lee.
 


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[Asterisk-Users] UK install

[Tim Guy]

Well I'm slowly learning my way around asterisk although as yet I
haven't had the chance to actually hook the system up to an ISDN line.

I am going to migrate from an Argent Office setup. My only problem is
keeping costs down on the phones.

The Argent system is running about 30 POTS phones. Can someone suggest
the cheapest option? Should I get some kind of large scale FXS box or
would the cost of doing that on a large scale work out the same as
getting cheap SIP phones?

I have a large number of POTS phones with headsets so I would have to
take that into account if I replaced the phones with SIP's

In an ideal world Id like to convert a number of POTS to soft phones but
as always its persuading the users that they can operate in the same
way.

Our Telco is NTL offering us an ISDN 30 style package. I assume this is
a E100P card requirement? Any suggestions for good UK reseller or shall
I get it direct from Digium?

Anyhow, as I say I'm getting more functionality out of Asterisk than I
ever did with (personally thinking) a very confusing Argent setup. I
just hope that I can make it financially viable to do the install

Cheers

Tim

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Re: [Asterisk-Users] TE410P / Eicon PRI

[Andrew Kohlsmith]

On Friday 18 June 2004 11:08, Lee Howard wrote:
 If you would rather use HylaFAX instead of spandsp and have $10K to
 throw around, then may I suggest hiring an Asterisk channel author to
 write a T.38-supporting channel driver?  That way you could just use
 t38modem with HylaFAX, and you wouldn't need all the duplicate hardware.

I am gearing up to write a character port emulator which will telnet to an 
Ascend Max modem bank for HylaFax.  It's based on code from ttywatch which 
does the opposite.  (it is a telnet daemon that connects to a character port) 
-- rtty, ser2net and conserver are all apps which do the opposite.

Hopefully it will work alright, as the Ascend Max will give you a direct 
connection to its modem bank when enabled.

I was going to use T38modem but, like practically everything else h.323, the 
code is disgustingly hard to wade through.  :-(

Regards,
Andrew
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Re: [Asterisk-Users] oh323

[Jeremy McNamara]

T. Chan wrote:
Jeremy,
Yes, I felt that it was important to report my trouble and I did it three
times, reporting to the asterisk community, but for some reasons, I was not
being responded to at all. I thought my messages were embedded among the
hundreds of them and were missed out or everyone was having the same problem
and was not able to help.

Ok...What bug number?  I haven't paid very close attention to Mantis, 
but I thought I had it setup to email me when someone assigned a bug to me.



Jeremy, I have followed all instructions of yours by compiling the correct
verson of pwlib and openh323 (by doing make clean opt under each directory),
I have then gone into H323 and done a 'make' before going back to
/usr/src/asterisk to do a 'make install'. I tried using sjphone, I tried
using another asterisk, I tried using cisco to call into it, but I just was
not able to get any audio at all, when using the old version, I was able to
do so no problem with all the equipment above.

I just tried sjphone and chan_h323 and it worked on the very first call. 
  cvs -head.


Jeremy, I don't know if there is any change on the h323.conf or any other
file that I need to do, please let me know, because I have not changed any
configuration files.

Look at the h323.conf.sample

Jeremy McNamara





-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jeremy
McNamara
Sent: Friday, June 18, 2004 3:57 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] oh323
T. Chan wrote:

Jeremy
I speak for myself, I have been testing with oh323 driver as well, because
in my case, your h323 driver is not working, it was working before, but
then
when I started to upgrade to 0.7.0 version of asterisk and from that point
onwards (beginning of January), calls have had no audio. I tried making
calls and I was getting no audio at all when the call was connected. Since
then, I have not been able to upgrade the asterisk version, because if so,
I
would not be able to run h323. That is why in my case, I have been trying
to
explore the other alternative. If you have some idea to it, please let me
know, thanks alot

So you didn't feel it was important to report your trouble anywhere?
I have tested the cvs -head of asterisk with many different types of
H.323 gateways and cannot make it fail.
Jeremy McNamara
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Re: [Asterisk-Users] C7960 g729 question

[Dominique Kull]

What does your sip.conf look like? Always make sure that you have the 
following codec order for G.729 pass-thru:

[general]
disallow=all
allow=g729
allow=ulaw
allow=alaw
you don't need to force your C7960 (SIP settings) to use G.729 with the 
above config.

see also:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20G.729%20pass-thru
Dominique
Rich Adamson wrote:
I have multiple voiceage g729 licenses installed on a RH9 box, and have
a remote C7960 configured to use it (low bandwidth). In calls like:
  Remote C7960 - g729 - asterisk - g711 - C7960
the audio is oftentimes rather choppy. Changing the remote 7960 to use
g711 seems to eliminate/reduce the choppyness. Any ideas on what might
be behind this?

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--
Dominique Kull
The Old Lodge, London SW6 6EE UK
t: +44 207 731 1562
v: fwd 268167
e: [EMAIL PROTECTED]
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Re: [Asterisk-Users] TE410P / Eicon PRI

[Darren Nickerson]

You don't even need spandsp - fax is dead, remember? ;-)

-d

- Original Message - 
From: Klaus-Peter Junghanns [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, June 18, 2004 11:10 AM
Subject: Re: [Asterisk-Users] TE410P / Eicon PRI


 better send the EUR 10k (not $10k... :)  ) to the author of spandDSP.
 Nobody needs HylaFAX for receiving faxes. Converting a tiff to pdf and
 storing it somewhere is not rocket science. ;)

 best regards

 Klaus

 Am Fr, 2004-06-18 um 17.08 schrieb Lee Howard:
  If you would rather use HylaFAX instead of spandsp and have $10K to
  throw around, then may I suggest hiring an Asterisk channel author to
  write a T.38-supporting channel driver?  That way you could just use
  t38modem with HylaFAX, and you wouldn't need all the duplicate hardware.
 
  Lee.
 


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Re: [Asterisk-Users] TE410P / Eicon PRI

[Andrew Kohlsmith]

On Friday 18 June 2004 11:10, Klaus-Peter Junghanns wrote:
 better send the EUR 10k (not $10k... :)  ) to the author of spandDSP.
 Nobody needs HylaFAX for receiving faxes. Converting a tiff to pdf and
 storing it somewhere is not rocket science. ;)

Incorrect.  I've been unable to get spandsp operating consistently with 
Slackware 9.1 and libtiff 3.6.0.  Some faxes receive great, some are 
completely corrupted and the biggest problem is that some (most) fax 
reception segfaults asterisk.  :-(

Regards,
Andrew
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[Asterisk-Users] Re: 7960 straight through?

[Randy Bush]

 Anyway, it appears as though the two contexts you have listed below have
 the exact same name in-internal,

sorry, my error in anonymizing the stuff.  the dupe is not in
the real config.

randy

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RE: [Asterisk-Users] UK install

[mattf]

If you're already using POTS phones and want the flexibility of SIP you may
just want to get SIP adapters that you can continue to use your POTS phones
with. I recommend the Sipura SPA-2000 dual analog adapter(www.sipura.com).
You can get them for about $92(if you get more than 10 in one order) and
each one handles two analog phones. They have a lot of features and great
support. We are currently using them on over 80 analog phones across 3
Asterisk servers and they work great. 

A cheaper option might be using channel banks for your POTS phones but that
may only be cheaper if you buy a used one, and then you wouldn't have the
flexibility of using SIP adapters, and you would probably need to get a quad
T1 card if you were planning on only getting a single T1 card, which would
make it much more expensive overall.

Hope this helps. Good luck

MATT---


-Original Message-
From: Tim Guy [mailto:[EMAIL PROTECTED]
Sent: Friday, June 18, 2004 11:37 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] UK install


Well I'm slowly learning my way around asterisk although as yet I
haven't had the chance to actually hook the system up to an ISDN line.

I am going to migrate from an Argent Office setup. My only problem is
keeping costs down on the phones.

The Argent system is running about 30 POTS phones. Can someone suggest
the cheapest option? Should I get some kind of large scale FXS box or
would the cost of doing that on a large scale work out the same as
getting cheap SIP phones?

I have a large number of POTS phones with headsets so I would have to
take that into account if I replaced the phones with SIP's

In an ideal world Id like to convert a number of POTS to soft phones but
as always its persuading the users that they can operate in the same
way.

Our Telco is NTL offering us an ISDN 30 style package. I assume this is
a E100P card requirement? Any suggestions for good UK reseller or shall
I get it direct from Digium?

Anyhow, as I say I'm getting more functionality out of Asterisk than I
ever did with (personally thinking) a very confusing Argent setup. I
just hope that I can make it financially viable to do the install

Cheers

Tim

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[Asterisk-Users] Re: 7960 straight through?

[Scott Laird]

On Jun 18, 2004, at 8:56 AM, Randy Bush wrote:
Err, it works for me, with a 7940 and 6.3.  I've never bothered with
'NewCall' or 'Dial'; you can get around them if you can set up a 
decent
dialplan.xml.
aha.  ok.  thanks.  on to sorting out a dialplan.xml.  any simple
one that sez just give it all to asterisk?
Here's mine.  It's not terrifically complicated.  If you remove 
everything but the last TEMPLATE line, then it'll timeout on 
everything after 5 seconds.

The single-digit lines are new; once I've had time to verify that they 
work, I'll remove the 425 and 206 entries.  I'm obviously not using 
'dial 9 for an outside line' here.

DIALTEMPLATE
  TEMPLATE MATCH=22.. Timeout=0/
  TEMPLATE MATCH=425... Timeout=0/
  TEMPLATE MATCH=206... Timeout=0/
  TEMPLATE MATCH=2. Timeout=0/
  TEMPLATE MATCH=3. Timeout=0/
  TEMPLATE MATCH=4. Timeout=0/
  TEMPLATE MATCH=5. Timeout=0/
  TEMPLATE MATCH=6. Timeout=0/
  TEMPLATE MATCH=7. Timeout=0/
  TEMPLATE MATCH=8. Timeout=0/
  TEMPLATE MATCH=9. Timeout=0/
  TEMPLATE MATCH=1.. Timeout=0/
  TEMPLATE MATCH=* Timeout=5/
/DIALTEMPLATE
Scott
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[Asterisk-Users] trouble compiling zaptel-0.9.1 on YellowDog (PowerMac)

[Artur Jasowicz]

I am running asterisk on an old PowerComputing Mac clone running  
YellowDog 3.0 (Red Hat clone for PowerMacs) I've decided to try adding  
a generic winmodem card and compile zaptel-0.9.1 for it.

First I tried to just unpack zaptel archive and do make clean; make  
install. Compiled fine, but during insallation I got the unresolved  
symbols error messages from depmod -a

I did some research and followed instructions at
http://www.voip-info.org/tiki-index.php? 
page=Asterisk%20Zaptel%20Installation

I copied my config file. BTW: instructions above direct to:
cp /boot/config-2.4.28 /usr/src/.config
Shouldn't that be:
cp /boot/config-`uname -r` /usr/src/linux-`uname -r`/.config
/usr/src does not seem like the right location to put .config into.
I did menuconfig and make dep steps, then removed old untarred zaptel  
dir, untarred fresh copy, make clean; make  log.make 21. Now my  
log.make file contains a looong list of complaints about problems in:
/usr/src/linux-2.4/include/linux/kernel.h
/usr/src/linux-2.4/include/asm/processor.h
/usr/src/linux-2.4/include/asm/cache.h
/usr/src/linux-2.4/include/asm/atomic.h
/usr/src/linux-2.4/include/linux/module.h
/usr/src/linux-2.4/include/linux/dcache.h
/usr/src/linux-2.4/include/asm/pci.h
while trying to compile tor2.c. Finally, it ends with:
/usr/src/linux-2.4/include/asm/pci.h:98: warning: implicit declaration  
of functi
on `printk_Rdd132261'
tor2.c: In function `init_spans':
tor2.c:274: warning: implicit declaration of function  
`sprintf_R1d26aa98'
make: *** [tor2.o] Error 1

Compiling asterisk as well as other software on this machine went well.  
What should I check to resolve this? I can do some basic compile  
troubleshooting, but this one seems like too much for me to handle on  
my own. Anyone care to see the entire log.make I generated? It is 140  
lines long.

Artur
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RE: [Asterisk-Users] Soekris Engineering net4801

[John Bittner]

Hi,

We used 512meg compact flash running debian. 

John Bittner
Simlab.net
 

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 W. Kevin Hunt
 Sent: Thursday, June 17, 2004 8:54 PM
 To: [EMAIL PROTECTED]
 Subject: RE: [Asterisk-Users] Soekris Engineering net4801
 
 
 John Bittner wrote:
  Hi,
  
  I have it working great. I have debian running on it with music on 
  hold disabled. I setup 10 cisco 7960 phones and tested the 
 4801 with 
  calls on all 10 phones at the same time through voicepulse with no 
  issues. I ran top with all the phones running and I was only up to
  45% cpu. Seems to run ok but I am still in the testing phase.   
 
 What storage medium did you use, compact flash for 2.5 HD ?
 What OS/flavor did you use? 
 
 W. Kevin Hunt
 CCIE #11841
 
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RE: [Asterisk-Users] UK install

[Chris Bond]

We're thinking of doing the same with our argent office system at the
moment.

 The Argent system is running about 30 POTS phones. Can someone suggest
 the cheapest option? Should I get some kind of large scale FXS box or
 would the cost of doing that on a large scale work out the same as
 getting cheap SIP phones?

Best bet is to use an IAXy or supr to convert the phone into an IAX2.  The
supr's have passthrough Ethernet ports so easier to do.  I suggest you get
one to try first if you have headsets that's what we're in the process of
doing soon.

The other way to do it would be get an ADTRAN 650 or 750, you can pick them
up cheap on ebay.  But this requires an extra PRI interface for each ADTRAN
box (unless there linked so they run via single T1 termination).

 Our Telco is NTL offering us an ISDN 30 style package. I assume this is
 a E100P card requirement? Any suggestions for good UK reseller or shall
 I get it direct from Digium?

Should work yes, digium direct are good or telappliant.

Would be really good if you could post your config files on a website once
you've got it all up wouldn't mind seeing the config, as we're about 3
months off before we think of converting ours.

Kind Regards,
Chris Bond

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Re: [Asterisk-Users] TE410P / Eicon PRI

[Klaus-Peter Junghanns]

Am Fr, 2004-06-18 um 17.53 schrieb Darren Nickerson:
 You don't even need spandsp - fax is dead, remember? ;-)
 
Why do YOU sell hylafax servers then? ;)

best regards

Klaus

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[Asterisk-Users] Lingo and *

[Andreas Schiffler]

Hi,

just found out about the great lingo.com service offerings.

Could this be used with Asterisk? I have a couple of Sipuras on the LAN
and would like to use * to route this to Lingo or my POTS adapter.

People report that Lingo is using SIP although they say it can only be
used with their ATA. They claim PBX compatibility on their website
though.

Regards
Andreas
 


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Re: [Asterisk-Users] TE410P / Eicon PRI

[Steve Underwood]

Andrew Kohlsmith wrote:
On Friday 18 June 2004 11:10, Klaus-Peter Junghanns wrote:
 

better send the EUR 10k (not $10k... :)  ) to the author of spandDSP.
Nobody needs HylaFAX for receiving faxes. Converting a tiff to pdf and
storing it somewhere is not rocket science. ;)
   

Incorrect.  I've been unable to get spandsp operating consistently with 
Slackware 9.1 and libtiff 3.6.0.  Some faxes receive great, some are 
completely corrupted and the biggest problem is that some (most) fax 
reception segfaults asterisk.  :-(
 

The segfaults I have followed up on have all been due to libtiff 
versions. Are you sure there isn't some other version of libtiff lurking 
on your machine? If there isn't I would like to follow up with you and 
find why this happens. Many people are getting reliable performance.

Regards,
Steve
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[Asterisk-Users] Thousands of contexts?

[Manuel Wenger]

By reading the Wiki's I found out that an Asterisk server with many (1) 
extensions and/or SIP users can become slow when reloading. But what happens when you 
also have many contexts in extensions.conf? More precisely, one context for each SIP 
user?

I need this because I will have users with random usernames that they can choose, but 
I obviously cannot set that username as the outgoing caller ID when passing the call 
to our PSTN gateway. I need to change the CLI before dialling out. Now, every SIP user 
has his CLI, so I thought of creating a context for every user, where I would 
SetCallerID() before issuing the Dial() command. Obviously I would use some sort of 
script reading from a database to re-create the extensions.conf and sip.conf after 
making changes.

Do you see any issues which could arise? Is Asterisk going to crash, or is it just 
going to be slow when reloading?

Thank you for your help
-Manuel


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[Asterisk-Users] Re: 7960 straight through?

[Randy Bush]

my current, inherited, dialplan.xml is

DIALTEMPLATE
 TEMPLATE MATCH=00,1.. Timeout=0 User=Phone /
 TEMPLATE MATCH=00,*   Timeout=5 User=Phone /
 TEMPLATE MATCH=* Timeout=5 User=Phone /
/DIALTEMPLATE

the last of the three entries would seem to be the significant
one.

but my problem is that * is wanting the cisco to prepend its
own extension number to the dialed string.  see my original
message (corrected) below.

a ether dump of the sipura's invite shows

From: biwa phone sip:[EMAIL PROTECTED];tag=3a553a2b9373c699
To: sip:[EMAIL PROTECTED]
 ^^^

while cisco demands that i dial the 142 before it will send the invite
at all

randy

---

From: Randy Bush [EMAIL PROTECTED]
To: splatters [EMAIL PROTECTED]
Subject: 7960 straight through?
Date: Thu, 17 Jun 2004 17:42:36 -0700

if i go off hook and dial 666 from an internal sipura spa-x000
(at extn 141), it rings straight through to extn 666.

using the same dialplan, from a cisco 7960 with 7.1 sip code
(at extn 142), i have to
   go off hook
   hit NewCall
   punch 142  (or any valid extn in the dialplan)   the problem ***
   hit Dial
   then dial 666

sip.conf for crisco

[fiji]
callerid=crisco 142
type=friend
host=dynamic
port=5060
secret=pfui
qualify=1000
dtmfmode=rfc2833
canreinvite=yes
context=in-internal

extensions.conf

[in-internal]
exten = s,1,Answer
exten = 141,1,GoTo(int-extns,s,1)   ; spa-x000
exten = 142,1,GoTo(int-extns,s,1)   ; 7960

[in-extns]
exten = s,1,Answer
exten = s,2,DigitTimeout,5
exten = s,3,ResponseTimeout,10
exten = s,4,PlayTones(dial)
exten = 141,1,Macro(dial-extension,marais)
exten = 142,1,Macro(dial-extension,fiji)
exten = 666,1,Macro(dial-extension,downthere)

-30-

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[Asterisk-Users] Asterisk as Media Gateway (was: ATT CallVantage Asterisk)

[Stewart Nelson]

Hi Philip,

Unfortunately, * speaks MGCP only as the Call Agent, rather
than as the Media Gateway.  MGCP is a master/slave protocol,
and it would take some effort to make * work as the slave.

I have the same problem: Free Telecom here in Paris includes
MGCP service with their DSL.  You can call any fixed phone in
France at no charge!  Rates to mobiles and international are
quite aggressive, too.  Various ISPs around the world have
similar offers, so I'm surprised that nobody has yet implemented
a solution for *.

In the short term, you could connect the FXS ports on the D-Link
to some FXO interfaces (PCI, TDM, SIP, H.323, or even MGCP).
Of course, this impairs voice quality, increases delay, may
disrupt some functions, is a hassle to administer, etc.
But it's better than nothing.

I am considering adding MG capability to the * MGCP stack.
Do you or does anyone have an interest in helping with this?

--Stewart

 Date: Fri, 18 Jun 2004 10:30:15 -0400
 From: Kubat, Philip [EMAIL PROTECTED]
 Subject: [Asterisk-Users] ATT CallVantage  Asterisk

 I am trying to connect directly to ATT VoIP service CallVanage.  I have
 ATT's ATA (D-Link DVG-1120M).  They use mgcp.  I have traces of the connects
 from the Dlink and hoping to setup Asterisk the same.  It looks like I need
 to have Asterisk be a MGCP endpoint (gateway).   How do I configure this?
 Does the mgcp.conf support register like sip etc?  What is the syntax?
 

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Re: [Asterisk-Users] TE410P / Eicon PRI

[Lee Howard]

On 2004.06.18 08:34 Andrew Kohlsmith wrote:
On Friday 18 June 2004 11:08, Lee Howard wrote:
 If you would rather use HylaFAX instead of spandsp and have $10K to
 throw around, then may I suggest hiring an Asterisk channel author
to
 write a T.38-supporting channel driver?  That way you could just use
 t38modem with HylaFAX, and you wouldn't need all the duplicate
hardware.
I am gearing up to write a character port emulator which will telnet
to an
Ascend Max modem bank for HylaFax.  It's based on code from ttywatch
which
does the opposite.  (it is a telnet daemon that connects to a
character port)
-- rtty, ser2net and conserver are all apps which do the opposite.
Hopefully it will work alright, as the Ascend Max will give you a
direct
connection to its modem bank when enabled.
I was going to use T38modem but, like practically everything else
h.323, the
code is disgustingly hard to wade through.  :-(
Well, if you don't like t38modem, then a really cool thing would be if 
you wrote a T.38 driver for HylaFAX also.  So then Asterisk and HylaFAX 
could play together without t38modem, without the AT command-response 
language limitations.

Lee.
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RE: [Asterisk-Users] oh323

[T. Chan]

Jeremy

I did not report that to the bug tracker, I did not even think that was a
bug, I just thought may be I did something wrong, and I reported my problem
3 times to this mailing list, trying to get some light to my problem, I did
not get any response.

This time, at least I got some response, but I don't think it helps much.
May be that is why the other gentlemen Michael was trying the other driver
as well.

Thanks

TC

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jeremy
McNamara
Sent: Friday, June 18, 2004 11:40 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] oh323



T. Chan wrote:

 Jeremy,

 Yes, I felt that it was important to report my trouble and I did it three
 times, reporting to the asterisk community, but for some reasons, I was
not
 being responded to at all. I thought my messages were embedded among the
 hundreds of them and were missed out or everyone was having the same
problem
 and was not able to help.


Ok...What bug number?  I haven't paid very close attention to Mantis,
but I thought I had it setup to email me when someone assigned a bug to me.




 Jeremy, I have followed all instructions of yours by compiling the correct
 verson of pwlib and openh323 (by doing make clean opt under each
directory),
 I have then gone into H323 and done a 'make' before going back to
 /usr/src/asterisk to do a 'make install'. I tried using sjphone, I tried
 using another asterisk, I tried using cisco to call into it, but I just
was
 not able to get any audio at all, when using the old version, I was able
to
 do so no problem with all the equipment above.



I just tried sjphone and chan_h323 and it worked on the very first call.
   cvs -head.



 Jeremy, I don't know if there is any change on the h323.conf or any other
 file that I need to do, please let me know, because I have not changed any
 configuration files.


Look at the h323.conf.sample



Jeremy McNamara











 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Behalf Of Jeremy
 McNamara
 Sent: Friday, June 18, 2004 3:57 AM
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] oh323


 T. Chan wrote:


Jeremy

I speak for myself, I have been testing with oh323 driver as well, because
in my case, your h323 driver is not working, it was working before, but

 then

when I started to upgrade to 0.7.0 version of asterisk and from that point
onwards (beginning of January), calls have had no audio. I tried making
calls and I was getting no audio at all when the call was connected. Since
then, I have not been able to upgrade the asterisk version, because if so,

 I

would not be able to run h323. That is why in my case, I have been trying

 to

explore the other alternative. If you have some idea to it, please let me
know, thanks alot



 So you didn't feel it was important to report your trouble anywhere?


 I have tested the cvs -head of asterisk with many different types of
 H.323 gateways and cannot make it fail.


 Jeremy McNamara
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RE: [Asterisk-Users] Thousands of contexts?

[Chris Bond]

Why not use mysql as it should be faster I'd suspect

-Original Message-
From: Manuel Wenger [mailto:[EMAIL PROTECTED] 
Sent: 18 June 2004 5:43 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Thousands of contexts?

By reading the Wiki's I found out that an Asterisk server with many (1)
extensions and/or SIP users can become slow when reloading. But what happens
when you also have many contexts in extensions.conf? More precisely, one
context for each SIP user?

I need this because I will have users with random usernames that they can
choose, but I obviously cannot set that username as the outgoing caller ID
when passing the call to our PSTN gateway. I need to change the CLI before
dialling out. Now, every SIP user has his CLI, so I thought of creating a
context for every user, where I would SetCallerID() before issuing the
Dial() command. Obviously I would use some sort of script reading from a
database to re-create the extensions.conf and sip.conf after making changes.

Do you see any issues which could arise? Is Asterisk going to crash, or is
it just going to be slow when reloading?

Thank you for your help
-Manuel


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[Asterisk-Users] Re: X100P in Switzerland

[Aaron Clauson]

Hi,

I had a similar problem for a while in Ireland.
Eventually after much hair tearing I decided it must
be something to do with the phone socket and commenced
to make a direct conenction between the twisted pair
and the X100P socket. Low and behold it worked.

After more mucking around I found I could get the card
to work, and get the red alarm removed, by jiggling
the RJ11 cable in the phone socket. I would plug a
analogue phone into the X100P and then a cable from
the line in on the card to the phone socket. By moving
the cable in and out of the socket I could get the
signal passed through to the phone and at the same
time clear the red alarm. I am pretty sure this has
something to do with the line impedance but despite
having a dim distant electronic engineering degree
don't really understand it??

hth,
Aaron

quote
Hi

Does anybody if the X100P works in Switzerland? We
can't get a line to 
PSTN.

When I run zttool it shows me always a red alert. I
can make and receive calls with an
anlog phone plugged in the phone connector.

I've compiled and configured the card according to the
wiki. Everything 
seemed to be ok.

Is there a way to debug this?

Regards
Reto
/quote





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Re: [Asterisk-Users] TE410P / Eicon PRI

[Steve Underwood]

Klaus-Peter Junghanns wrote:
Am Fr, 2004-06-18 um 17.53 schrieb Darren Nickerson:
 

You don't even need spandsp - fax is dead, remember? ;-)
   

Why do YOU sell hylafax servers then? ;)
best regards
Klaus
 

Working with the dead never stopped undertakers making a living :-)
Regards,
Steve
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Re: [Asterisk-Users] TE410P / Eicon PRI

[Lee Howard]

On 2004.06.18 08:10 Klaus-Peter Junghanns wrote:
better send the EUR 10k (not $10k... :)  ) to the author of spandDSP.
Nobody needs HylaFAX for receiving faxes.
Firstly, I'm not just talking about receiving faxes.
If my choices are between HylaFAX and spandsp and if I want outbound 
queueing and a client-server interface for networked usage, then 
spandsp will not cut it alone.

So yes, anyone who wants these features will need to use HylaFAX.  And 
to use HylaFAX with Asterisk currently one must send the fax calls to 
an FXS port and then to a HylaFAX-controlled modem.

This is not a pretty configuration, I completely agree.  And, I 
completely agree that there are a myriad of beautiful ways to do this, 
in theory.  But the coding does not exist for those to be reality.  So 
unless someone wants to code it or pay to have it coded, then those who 
want outbound queueing and a client-server interface must put up with 
the cumbersome configuration.

Furthermore, even if you assumed that spandsp was as stable as HylaFAX, 
there is a vast feature-set difference between them as far as the 
faxing itself goes.  Steve has already made it clear that he sees no 
future in fax, and that he does not intend to bridge that feature-set 
gap at all.

So, show me a T.38 channel driver for Asterisk.  And if you think that 
using t38modem is ugly, then show me a T.38 driver for HylaFAX.

Lee.
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Re: [Asterisk-Users] TE410P / Eicon PRI

[Andrew Kohlsmith]

On Friday 18 June 2004 12:37, Steve Underwood wrote:
 The segfaults I have followed up on have all been due to libtiff
 versions. Are you sure there isn't some other version of libtiff lurking
 on your machine? If there isn't I would like to follow up with you and
 find why this happens. Many people are getting reliable performance.

Yup I am positive.  I *did* have an older version of libtiff (3.7.9?) hanging 
around but after I found out about it I blew it away and made sure I rebuilt 
the libraries from scratch (making sure I had eliminated header files, 
libraries, everything).

Once I did that, as I said, I was able to receive faxes but only sporadically.  
I posted the data to this list earlier:

http://lists.digium.com/pipermail/asterisk-users/2004-June/049405.html
http://lists.digium.com/pipermail/asterisk-users/2004-June/049414.html
http://lists.digium.com/pipermail/asterisk-users/2004-June/049415.html
http://lists.digium.com/pipermail/asterisk-users/2004-June/049418.html
http://lists.digium.com/pipermail/asterisk-users/2004-June/049443.html

The most recent crash audio files are at 
http://www.mixdown.ca/~andrew/dump/akohlmith-faxsegfault2.tgz -- this occured 
on the same system; some fax receives work fine, some don't, and some (like 
this one) crash asterisk.  :-)

Regards,
Andrew
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Re: [Asterisk-Users] trouble compiling zaptel-0.9.1 on YellowDog (PowerMac)

[Steven Critchfield]

On Fri, 2004-06-18 at 11:06, Artur Jasowicz wrote:
 I am running asterisk on an old PowerComputing Mac clone running  
 YellowDog 3.0 (Red Hat clone for PowerMacs) I've decided to try adding  
 a generic winmodem card and compile zaptel-0.9.1 for it.
 
 First I tried to just unpack zaptel archive and do make clean; make  
 install. Compiled fine, but during insallation I got the unresolved  
 symbols error messages from depmod -a

Use -ae for depmod to find out what it is complaining about, then you
can move forward. My bet is there isn't support for the hardware on non
x86 hardware.

-- 
Steven Critchfield  [EMAIL PROTECTED]

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Re: [Asterisk-Users] TE410P / Eicon PRI

[Darren Nickerson]

- Original Message - 
From: Klaus-Peter Junghanns [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, June 18, 2004 12:03 PM
Subject: Re: [Asterisk-Users] TE410P / Eicon PRI


 Am Fr, 2004-06-18 um 17.53 schrieb Darren Nickerson:
  You don't even need spandsp - fax is dead, remember? ;-)
  
 Why do YOU sell hylafax servers then? ;)

Because the customers keep calling us wanting more FAX!! It's horrible.

;-)

-d

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Re: [Asterisk-Users] Thousands of contexts?

[Jeremy McNamara]

Manuel Wenger wrote:
By reading the Wiki's I found out that an Asterisk server with many (1) 
extensions and/or SIP users can become slow when reloading. But what happens when you 
also have many contexts in extensions.conf? More precisely, one context for each SIP user?
I need this because I will have users with random usernames that they can choose, but 
I obviously cannot set that username as the outgoing caller ID when passing the call 
to our PSTN gateway. I need to change the CLI before dialling out. Now, every SIP user 
has his CLI, so I thought of creating a context for every user, where I would 
SetCallerID() before issuing the Dial() command. Obviously I would use some sort of 
script reading from a database to re-create the extensions.conf and sip.conf after 
making changes.
Do you see any issues which could arise? Is Asterisk going to crash, or is it just 
going to be slow when reloading?

You need to learn more about Asterisk, especially power of Asterisk's 
dial plan.  There is absolutely no need for thousands of contexts on one 
box.

We have a tremendous amount of endpoints on our various systems, yet we 
only have 4 or 5 contexts.

If you cannot use the callerid directive in the sip.conf (or equivalent) 
to set the callerid once and forget it, you can always use astdb to 
store and have the ability to update callerid in real-time.

Then again, you could do what we do and let the customer specify their 
own callerid, until we receive any complaints then we would simply 
override it with a callerid directive in the appropriate config file on 
our system.

Jeremy McNamara
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RE: [Asterisk-Users] Thousands of contexts?

[Kevin Walsh]

Manuel Wenger [EMAIL PROTECTED] wrote:
 By reading the Wiki's I found out that an Asterisk server with many
 (1) extensions and/or SIP users can become slow when reloading. But
 what happens when you also have many contexts in extensions.conf? More
 precisely, one context for each SIP user?  
 
 I need this because I will have users with random usernames that they can
 choose, but I obviously cannot set that username as the outgoing caller
 ID when passing the call to our PSTN gateway. I need to change the CLI
 before dialling out. Now, every SIP user has his CLI, so I thought of
 creating a context for every user, where I would SetCallerID() before
 issuing the Dial() command. Obviously I would use some sort of script
 reading from a database to re-create the extensions.conf and sip.conf
 after making changes.   
 
 Do you see any issues which could arise? Is Asterisk going to crash, or
 is it just going to be slow when reloading? 
 
I don't quite understand your Caller*ID dilemma.

In your sip.conf, you'd have a block for each user, say [abc123].
That's your random username, yes?  The same block would also define
the password and other directives.  Why can't you simply include the
callerid directive to set the Caller*ID name and number?

The following should do the trick:

callerid = Kevin Walsh 1234567890

I don't know whether Asterisk would slow down when reloading thousands
of contexts, but it sounds reasonable to me - I wouldn't expect it to
get any quicker. :-)

-- 
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  _/_/_/   _/_/  _/_/_/_/_/  _/   K e v i n   W a l s h
 _/ _/_/  _/ _/ _/_/  _/_/[EMAIL PROTECTED]
_/   _/  _/_/_/_/  _/_/_/_/  _/_/

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Re: [Asterisk-Users] Lingo and *

[Simon Dorfman]

On 6/18/04 11:27 AM, Andreas Schiffler [EMAIL PROTECTED] wrote:
 Hi,
 
 just found out about the great lingo.com service offerings.
 
 Could this be used with Asterisk? I have a couple of Sipuras on the LAN
 and would like to use * to route this to Lingo or my POTS adapter.
 
 People report that Lingo is using SIP although they say it can only be
 used with their ATA. They claim PBX compatibility on their website
 though.
 
 Regards
 Andreas

Hi Andreas,
I just received my Lingo ATA yesterday and plan on tackling the same
question.  From what I understand, Lingo uses MGCP, not SIP.  Where did you
read that they are using SIP?

So far, Lingo has been great, I just made a 2 hour call to Germany from the
U.S.  Call quality was no different from POTS to my ear.  I will post a
thorough review of Lingo after using it further and after I try to figure
out how to get it working with asterisk.

Also if you decide to sign up and you feel like giving me a $25 credit to my
account (that would be very nice! :D), enter my name and email when you sign
up:
Simon Dorfman
simon (AT) simondorfman.com

Thanks,
Simon in New Orleans


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RE: [Asterisk-Users] Problems with X100P

[Kevin Walsh]

David J Carter [EMAIL PROTECTED] wrote:
 Don't you need a 'modprobe wcfxs' also?
 
Not for an FXO device, such as the X100P, no.

-- 
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 _/ _/_/  _/ _/ _/_/  _/_/[EMAIL PROTECTED]
_/   _/  _/_/_/_/  _/_/_/_/  _/_/

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[Asterisk-Users] Grandstream HT-286 and NAT

[Nathan Martinez]

I have 2 Grandstream HT-286 devices and an Asterisk server.  The *
Server is not using NAT and has port 5060 opened up.  One HT-286 is
using traditional NAT and the other HT-286 is behind a residential DSL
router/firewall.  I have the HT-286 setup as the DMZ Host in the
router/firewall so that all incoming connections are forwarded to the
HT-286.

HT-286-1 == NAT FW == * Server === Router/FW == HT-286-2

In the setup for HT-286-2 , I have filled in the Use NAT IP field with
the public IP for that location.  I did the same thing for HT-286-1 and
then I mapped a public IP to its private IP in the NAT FW.  At this
point, the two devices can call each other without any problems.

I want to use the HT-286 for our traveling users who will never know
what their IP is.  When I remove the Use NAT IP entry on HT-286-1 as
well as remove its direct IP mapping from the NAT FW, HT-286-1 can
register with the * Server, but when I try to call HT-286-2, all I get
is silence.  If I do a 'sip show channels' it shows that the call is
connected.  Here is what I have in my sip.conf for these two units:

[305]
type=friend
host=dynamic
nat=yes
qualify=100

[307]
type=friend
host=dynamic
nat=yes
qualify=100

Has anyone used these units in this scenario?  Does anyone have any
hints as to what I can try to get this working?  Your help is much
appreciated.

Nathan Martinez
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Re: [Asterisk-Users] Re: 7960 straight through?

[Joshua M. Thompson]

On Fri, 2004-06-18 at 13:03, Randy Bush wrote:
 if i go off hook and dial 666 from an internal sipura spa-x000
 (at extn 141), it rings straight through to extn 666.
 
 using the same dialplan, from a cisco 7960 with 7.1 sip code
 (at extn 142), i have to
go off hook
hit NewCall
punch 142  (or any valid extn in the dialplan)   the problem ***
hit Dial
then dial 666
 
 sip.conf for crisco
 
 [fiji]
 callerid=crisco 142
 type=friend
 host=dynamic
 port=5060
 secret=pfui
 qualify=1000
 dtmfmode=rfc2833
 canreinvite=yes
 context=in-internal
 
 extensions.conf
 
 [in-internal]
 exten = s,1,Answer
 exten = 141,1,GoTo(int-extns,s,1)   ; spa-x000
 exten = 142,1,GoTo(int-extns,s,1) ; 7960
 
 [in-extns]
 exten = s,1,Answer
 exten = s,2,DigitTimeout,5
 exten = s,3,ResponseTimeout,10
 exten = s,4,PlayTones(dial)
 exten = 141,1,Macro(dial-extension,marais)
 exten = 142,1,Macro(dial-extension,fiji)
 exten = 666,1,Macro(dial-extension,downthere)

The reason  you're getting this behavior from the Cisco is that you have
assigned it to the in-internal context. That context has no way out
other than to dial a valid extension. Once you do that it transfers to
the in-extns context, where 666 is valid. I bet yourother phone is set
to be in the in-extns context so it doesn't need to do this to dial
out.

Just out of curiosity why do you have this strange setup? I usually use
a setup something like this:

[extensions]

exten = 101,1,Macro(vmextension,101,${EXTEN101})
exten = 102,1,Macro(vmextension,102,${EXTEN102})

[pstn]

exten = _NX,1,Macro(route,${EXTEN})

[applications]

exten = *98,1,VoicemailMain(${CALLERIDNUM})

[speeddials]

exten = #01,1,Macro(route,2345678901)

[internal]

include = extensions
include = applications
include = speeddials
include = pstn

(where the 'route' macro is a macro that looks up the NPA/NXX via dbodbc
and routes local calls to my analog trunks and long distance calls to my
VoIP trunk)

And then all my cisco phones are set to be in the internal context and
they can dial any internal extension as 1XX or dial a plain ten digit
PSTN number. There won't be a conflict because my dialplan uses strict
10D dialing (no 1+number) so anything beginning with 1 cannot be a valid
PSTN number. So my dialplan.xml is set to allow 1XX to dial immediately.

If you need more help with your dialing plan email me off list. I have
four Cisco phones in my house (1 7960G and three 7940Gs) and they're all
working just fine without problems using SIP with firmware version 7.1.

-- 
Joshua M. Thompson [EMAIL PROTECTED]

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Re: [Asterisk-Users] TE410P / Eicon PRI

[Andrew Kohlsmith]

On Friday 18 June 2004 13:20, Lee Howard wrote:
 Well, if you don't like t38modem, then a really cool thing would be if
 you wrote a T.38 driver for HylaFAX also.  So then Asterisk and HylaFAX
 could play together without t38modem, without the AT command-response
 language limitations.

I wasn't intending to write any t38 code; my intention was to write a pseudo 
char device and set up a telnet connection to the Ascend Max modem bank.  I 
agree that a t38 driver would be cool but I'm stretching my programming 
abilities as is :-)

Also Darren has provided a link to an unoffical patch for HylaFax which allows 
it to contact TCP modems which I intend to play with shortly.  :-)

-A.
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[Asterisk-Users] cisco 924 config

[Gabriel C Millerd]

hello

i have a cisco 924 router (its a router with a cable modem
interface, ethernet interface hublet, two pots jacks/fxs and one
pstn jack/fxo). i am not using the cable modem interface. i
merely want to use it as an ata device, possibly just a fxs if
thats all that can be done.

as some may know its a flaky device and never was very well
supported by cisco because of flaws with the cable modem
interface i gather. it is stuck at ios 12.2 i believe.

i have asterisk-oh323 working with asterisk (i am able to make
open phone calls just like any other device) however i cannot
seem to lick the cs924 hurdle does anyone have a configuration
that works with this device.

thanks very much

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Re: [Asterisk-Users] TE410P / Eicon PRI

[Klaus-Peter Junghanns]

Am Fr, 2004-06-18 um 19.56 schrieb Lee Howard:
 Firstly, I'm not just talking about receiving faxes.
 
 If my choices are between HylaFAX and spandsp and if I want outbound 
 queueing and a client-server interface for networked usage, then 
 spandsp will not cut it alone.
 
 So yes, anyone who wants these features will need to use HylaFAX.  And 
 to use HylaFAX with Asterisk currently one must send the fax calls to 
 an FXS port and then to a HylaFAX-controlled modem.

Theoretically chan_capi could also be modified for fax support, since
that is already part of the CAPI specs. But spanDSP works for all 
channel types so i dont see the need for this.

For outbound spooling pbx_spool is your friend. If you want to take
total control of the spooling yourself you can also build something
very nice and scalable with the manager interface.

 
 This is not a pretty configuration, I completely agree.  And, I 
 completely agree that there are a myriad of beautiful ways to do this, 
 in theory.  But the coding does not exist for those to be reality.  So 
 unless someone wants to code it or pay to have it coded, then those who 
 want outbound queueing and a client-server interface must put up with 
 the cumbersome configuration.
 
I agree that the hylafax clients are really nice and very useful.

 Furthermore, even if you assumed that spandsp was as stable as HylaFAX, 
 there is a vast feature-set difference between them as far as the 
 faxing itself goes.  Steve has already made it clear that he sees no 
 future in fax, and that he does not intend to bridge that feature-set 
 gap at all.
 

Correct me if I am wrong, but hylafax and spanDSP are two totally
different pairs of shoes. Hylafax relies on the modem device to 
actually provide the fax capability. SpanDSP is pure software solution.
You can fax with any Asterisk channel driver even VoIP.

Apart from the missing network client you can build any feature you
can dream about with Asterisk.

Oh, and btw, i receive all my faxes with capi4hylafax and HylaFAX of
course, just because SuSE comes with such a nice configruation tool
for it (and i am lazy!). :)

best regards

Klaus


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RE: [Asterisk-Users] Thousands of contexts?

[Carlton J. O'Riley]

Is there any reason you can't use the callerid=name number in sip.conf
instead of a ton of contexts to do this? 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Bond
Sent: Friday, June 18, 2004 1:47 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Thousands of contexts?

Why not use mysql as it should be faster I'd suspect

-Original Message-
From: Manuel Wenger [mailto:[EMAIL PROTECTED]
Sent: 18 June 2004 5:43 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Thousands of contexts?

By reading the Wiki's I found out that an Asterisk server with many (1)
extensions and/or SIP users can become slow when reloading. But what happens
when you also have many contexts in extensions.conf? More precisely, one
context for each SIP user?

I need this because I will have users with random usernames that they can
choose, but I obviously cannot set that username as the outgoing caller ID
when passing the call to our PSTN gateway. I need to change the CLI before
dialling out. Now, every SIP user has his CLI, so I thought of creating a
context for every user, where I would SetCallerID() before issuing the
Dial() command. Obviously I would use some sort of script reading from a
database to re-create the extensions.conf and sip.conf after making changes.

Do you see any issues which could arise? Is Asterisk going to crash, or is
it just going to be slow when reloading?

Thank you for your help
-Manuel
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Fwd: Re: [Asterisk-Users] Disable IAX1 Registrations

[Christopher Lewis]

Just in case anyone else is looking for this information, I'm posting the 
answer here.  I do need to double check if this works with asterisk = 0.9.0.  
I've heard rumors that IAX1 support has been removed in newer versions.

--  Forwarded Message  --

Subject: Re: [Asterisk-Users] Disable IAX1 Registrations
Date: Thursday 17 June 2004 04:43 pm
From: Chris A. Icide
To: Christopher Lewis 

Christopher,

Create a iax1.conf file (iax.conf is actually the config file for IAX2 in
versions of asterisk that have both iax implementations).  Then set the
registrations you want for IAX2 in the iax.conf file and registrations you
want for IAX1 in the iax1.conf file.  Shutdown and restart asterisk.

-Chris
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RE: [Asterisk-Users] Thousands of contexts?

[Steven Critchfield]

On Fri, 2004-06-18 at 12:46, Chris Bond wrote:
 Why not use mysql as it should be faster I'd suspect

I doubt it would be faster as asterisk will keep it all in memory, only
changes might be slowed.

But the thought is correct, use a database to store the data and one
context that does a lookup into the database and populates your
callerid. It is a better way of doing things. You could even host it in
the ast_db and then it shouldn't be too slow as you aren't spawning any
outside apps.

 -Original Message-
 From: Manuel Wenger [mailto:[EMAIL PROTECTED] 
 Sent: 18 June 2004 5:43 PM
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] Thousands of contexts?
 
 By reading the Wiki's I found out that an Asterisk server with many (1)
 extensions and/or SIP users can become slow when reloading. But what happens
 when you also have many contexts in extensions.conf? More precisely, one
 context for each SIP user?
 
 I need this because I will have users with random usernames that they can
 choose, but I obviously cannot set that username as the outgoing caller ID
 when passing the call to our PSTN gateway. I need to change the CLI before
 dialling out. Now, every SIP user has his CLI, so I thought of creating a
 context for every user, where I would SetCallerID() before issuing the
 Dial() command. Obviously I would use some sort of script reading from a
 database to re-create the extensions.conf and sip.conf after making changes.
 
 Do you see any issues which could arise? Is Asterisk going to crash, or is
 it just going to be slow when reloading?
 
 Thank you for your help
 -Manuel
 
 
 ___
 Ticinocom SA - Via Stazione 5 - 6600 Muralto
 Tel 0844 007070 - Fax 0844 007071
 http://www.ticinocom.com
 
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-- 
Steven Critchfield  [EMAIL PROTECTED]

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[Asterisk-Users] Iaxy issue

[Glen Hinkle]

Folks, 

Randomly, when the phone is taken off-hook, the the Iaxy produces a
irritating banshee scream as opposed to a dial-tone.  Cycling the power
fixes the issue,  sometimes it magically goes away by itself.  

Has anyone experienced this issue  potentially fixed it?  

I'm using asterisk CVS head as of jun 17 2004.  

Thanks, 
Glen


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Re: [Asterisk-Users] anyone use mailboxexists?

[Michael George]

On Jun 18, 2004, at 10:57 AM, Jeremy Jones wrote:
 From the docs, it looks like MailboxExists() will add 101 to the
priority if the box *does* exist and goes to the next priority if not.
I think the show application mailboxexists documentation is wrong.  I
believe it's the other way around.  It does exits? Jump to next
priority.  It doesn't?  Jump to n+101.  Here's my extension macro (sift
out the forwarding stuff if you don't like that), and it works:
Odd...  I did a make update and how the MailboxExists works fine.  
However, it works just as the docs say: add 101 to priority if the box 
*does* exist, add 1 if not.  I have tested it and this seems to be how 
it works.  You may wish to test your flow and make sure it works as you 
think it does.

[macro-stdexten]
exten=s,1,MailboxExists(${MACRO_EXTEN:[EMAIL PROTECTED]);If
mailbox exists continue at 2, otherwise goto 102
exten=s,2,NoOp  ;Filler
exten=s,3,NoOp  ;Filler
exten=s,4,NoOp  ;Filler
exten=s,5,NoOp  ;Filler
exten=s,6,DBget(temp=CFIM/${ARG1})  ;Get
CFIM key, if not existing, goto 107
exten=s,7,Dial(${TRUNK}/9${temp})
;Unconditional forward
exten=s,8,NoOp  ;Filler
exten=s,9,Dial(${ARG2},25,rtT)  ;Dial device for
25 seconds, goto 10 if busy, goto 110 if unavailable
exten=s,10,NoOp ;Filler
exten=s,11,DBget(temp=CFBS/${ARG1}) ;Get
CFBS key, if not existing, goto 112
exten=s,12,Dial(${TRUNK}/9${temp})  ;Forward
on busy or unavailable
exten=s,102,DBget(temp=CFIM/${ARG1});Get CFIM key,
if not existing, goto 203
exten=s,103,Dial(${TRUNK}/9${temp})
;Unconditional forward
exten=s,104,Dial(${ARG2},120,rtT)   ;Dial
device for 120 seconds, goto 105 if busy, goto 205 if unavailable
exten=s,105,DBget(temp=CFBS/${ARG1});Get CFBS key,
if not existing, goto 206
exten=s,106,Dial(${TRUNK}/9${temp}) ;Forward
on busy or unavailable
exten=s,107,Goto(s,9)   ;Goto 9
exten=s,110,Voicemail(u${MACRO_EXTEN:[EMAIL PROTECTED]) ;To VM
if unavailable
exten=s,111,Hangup  ;Hang up
the channel when vm exits
exten=s,112,Voicemail(b${MACRO_EXTEN:[EMAIL PROTECTED]) ;To VM
if busy
exten=s,113,Hangup  ;Hang up
the channel when vm exits
exten=s,203,Goto(s,104) ;Goto
104 for accounts w/out vm
exten=s,205,Busy()  ;Busy
signal if busy  no vm
exten=s,206,Busy()  ;Busy
signal if no answer in 2 min  no vm
It's a little ugly w/all those NoOps, but I think I need those to get
the priorities right.
Jeremy Jones

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-Michael
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