date:20060718



Has somebody done that with a Grandstream  GXP-2000 or a BudgetTone 
100/101 ?

Has somebody even a list which SIP phones have this funtion?


Regards
Kai


It's called hotline or Private Line Auto Ringdown (PLAR).

SIP: It's a function of the phone, look for hotline in phone docs
Zap: immediate=yes, runs exten = when phone is picked up
Cisco and others: Look up PLAR






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Re: [asterisk-users] IVR DTMF

2006-07-18 Thread Filip Drągowski





cd /var/lib/asterisk/agi-bin
touch dtmf2txt.sh
chmod +x dtmfivr.sh
edit dtmf2txt.sh by Your favorite text editor
i'm using mc -e dtfmivr.sh
(this is onlu an example)
    #!/bin/bash
    digits=$1
    number=$2
    time=`date`
    echo "$time : $1"  /home/dtmf2txt/$2.txt

in extensions conf when DTMF are pressed You have to store pressed
numbers in variable
when full number is reached You have to execute bash script using
System(/var/lib/asterisk/agi-bin/stmfivr.sh
${variable_with_number_pressed} ${calleridnum})
or AGI(/var/lib/asterisk/agi-bin/stmfivr.sh
${variable_with_number_pressed} ${calleridnum})
then make some test : does System / AGI is executed ? doeas script
create proper files ? 

  
  
  

  
  
  I know may
be I am disturbing you ,but I am
too thanks full for your help 
  But can you
explain in detailed steps how
to do that 
   
  What I
understand from you I that I should
 put this line at asterisk.conf and its already exist 
  And create a
bash script     
#!/bin/bash
digits=$1
number=$2
echo "$1"  $2.txt
  
  
  
  At /var/lib/asterisk/agi-bin/dtmfivr.sh
for example 
  After that what should I do 
    
  
  Regards
  And really thanks 
   
   
  in asterisk.conf there is 
"astagidir = /var/lib/asterisk/agi-bin"
it can be used for storing any scripts/programs fo *, it is suggested
for
storiong AGI scripts there
example: /var/lib/asterisk/agi-bin/dtmf2text.file.sh
  
  
  
  
  Thanks
for your help but where is should put this bash script
,can you guide me please 
  
  Regards 
   
  "...receiving digits from IVR
through
dtmf and store it on a text file "
short idea:
 1 IVR start
 2 set(number=)
 3 playback(press_digit_or_#_to_finish)
 4 (pressed) set(number=${number}${digit_pressed})
 5 playback(press_another_digit_or_#_to_finish)
 6 if digit pressed goto(pressed[point 44])
 7 if # pressed execute
System(put_string_with_pressed_didgits_into_text_file.sh
${digit_pressed}
${calleridnum})
  #this
script###
sh script
#!/bin/bash
digits=$1
number=$2
echo "$1"  $2.txt
  
  
  
  
  Dear 
   
   I want to make a billing recharge through
receiving digits from IVR
through dtmf and store it on a text file ,
   
  How can todo that ?
   
  Regards
   
   
  




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RE: [asterisk-users] Polycom IP301 and Queues

2006-07-18 Thread Dean @ INKnBITs

The setup looks fine, I will run through what I did and the version, there
might be an easier way.

cd /usr/src
svn checkout
http://svn.digium.com/svn/asterisk/team/bweschke/polycom_acd_functions/
asterisk-poly -r 30432

this will checkout the 30432 release and put in the the asterisk-poly
directory.

cd /usr/src/asterisk-poly

make clean
make  - I found you had to run make (2 or 3 times), it does come up on the
screen and tells you to re-run. First run I think makes menuconfig,
second can't remember.
make mpg123 (if you want mp3 music on hold)
make install

The only problem I can find in this release is the meetme (conference
centre) does not compile, (but ACD does) and in the newer version the meetme
works but not ACD. So I'm going to have two servers one for ACD on old
software and one for conference on new software. Not great but least it
works.

Hope that helps.

Regards,
Dean.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Michael
Miller
Sent: 17 July 2006 23:16
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Polycom IP301 and Queues


Thanks for the response and information.

The Asterisk version that I am using is Asterisk
SVN-bweschke-polycom_acd_functions-r37228. I went one revision back
using the following command:

svn checkout -r37228
http://svn.digium.com/svn/asterisk/team/bweschke/polycom_acd_functions
PolycomACD-07172006

 With no results. I am not as familiar with svn as cvs. I am not sure if
the -r option just labels or checks out the requested version. I will do
some reading tonight on svn.

I have install zaptel and libpri from the latest version of trunk.

I am using a Polycom 601 SIP version 1.6.6.0036.

The Polycom reg tag includes the following for line button one:

reg.1.displayName=Helpdesk
reg.1.address=1000
reg.1.label=Agent
reg.1.type=private
reg.1.thirdPartyName=
reg.1.auth.userId=
reg.1.auth.password=1000
reg.1.server.1.address=
reg.1.server.1.port=
reg.1.server.1.transport=DNSnaptr
reg.1.server.2.transport=DNSnaptr
reg.1.server.1.expires=
reg.1.server.1.register=
reg.1.server.1.retryTimeOut=
reg.1.server.1.retryMaxCount=
reg.1.server.1.expires.lineSeize=
reg.1.acd-login-logout=1
reg.1.acd-agent-available=1
reg.1.ringType=2
reg.1.lineKeys=1
reg.1.callsPerLineKey=2

I assumed that the property reg.1.auth.userId= is what you meant by
not putting in a username on the Polycom. I tried it both ways with no
luck.

I set the server addrss in the Polycom sip.cfg file.

The sip.conf entry for the Polycom looks like:

[1000]
type= friend
secret  = 1000
context = default
callerid= Helpdesk 1000
accountcode = 1000

host= dynamic
nat = no
qualify = 1000
canreinvite = no

disallow= all
allow   = ulaw

dtmfmode= rfc2833

agentlogin  = yes
agentcbcontext  = default

I also have an agent defined in the agnt.conf as:

agent = 2000,1234,Test Agent


Thanks again for the assistance!

Michael



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dean @
INKnBITs
Sent: Monday, July 17, 2006 3:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom IP301 and Queues

I had the same problems, first of all, what version of asterisk are you
using? If you run the CLI whats the polycom_acd_functions verison 3.
If
you did a svn checkout http://polycom_acd_function, then you
most
likely got the newest version. I had trouble with that.

Have you installed and compiled the zaptel/libpri from the trunk?
http://svn.digium.com/svn/zaptel/trunk and
http://svn.digium.com/svn/libpri/trunk ? You need these for the ACD
part.

On the polycom setup, make sure the username field is blank and that set
a
password.

In the Sip.conf, make sure the secret is the same as the polycom, and
that
you do not put a username= or a authname=


I can get you all the release/version numbers to download from the svn
tomorrow when back in work. It would be easier to talk you through it
when
in front of the server, but I'm in the UK and the time differences might
get
in the way!

Regards,
Dean.

- Original Message -
From: Michael Miller [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, July 17, 2006 6:56 PM
Subject: RE: [asterisk-users] Polycom IP301 and Queues


I have been unable to get this branch of asterisk to work properly. I
can not get any SIP phone, Polycom or X-Lite, to register with the
server. If, on the same server, I recompile and install Trunk the phones
register properly. In doing this I made no changes to the conf files at
all. I simply recompiled and reinstalled.

Is there a trick to getting the phones to register? I made sure that the
phone SIP config and the agent config did no overlap. The phone will
register if I comment out the secret line.

I

Re: [asterisk-users] Email notification of voicemail

2006-07-18 Thread Wilson Pickett


Have you tried this?

2000 = 1234,User Nametz=eastern24
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[asterisk-users] don't hear start/begin of voiceprompts

2006-07-18 Thread Mein Name


Hi all,

I just want to setup new voiceprompts for serveral queues in our 
asterisk pbx (Version 1.2.41.2.4)


The Problem is, that I don't hear the start (or the first part) of the 
voiceprompt.

It makes no differece if I use the Playback or Background Command.

But it makes a difference if the prompt is played automatically during 
entering the queue ore if I playback the file manually by typing 3 
for example:



exten = s,7,Background(01_ni_asterisk-b)   - missing start part at 
beginning of  prompt
exten = 3,1,Playback(01_ni_asterisk-b)- sound is ok (whole 
soundfile is played)


Here is my extensions_queues.conf

[menu-it]
exten = s,1,Set(LANGUAGE()=de)
exten = s,2,system(/bin/echo ${LANGUAGE}  /tmp/LANGUAGE)
include = sipint
exten = s,3,Answer
exten = s,4,SetMusicOnHold(default)
exten = s,5,Set(TIMEOUT(digit)=3)
exten = s,6,Set(TIMEOUT(response)=16)
exten = s,7,Background(01_ni_asterisk-b)
exten = s,8,Background(queue-menu-announcement)
exten = s,9,SetCallerID(IT Queue: 0${CALLERID})
exten = s,10,Queue(queue-it|t)
exten = s,11,Playback(nbdy-avail-to-take-call)
exten = s,12,Voicemail(u1700) ;
exten = s,13,Set(LANGUAGE()=en)
exten = #,1,Hangup
exten = t,1,Hangup
exten = o,1,Goto(menu-office,s,1)
exten = i,1,Playback(invalid)
exten = i,2,WaitExten
exten = 3,1,Playback(01_ni_asterisk-b)
exten = 4,1,Playback(02_ni_asterisk-b)




Thanks for any help.

morel




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Re: [asterisk-users] don't hear start/begin of voiceprompts


Okay, i will be one of the 100 answering this question.

what about a wait (2) before the background()?

That should manage your problem.


Mein Name schrieb:

Hi all,

I just want to setup new voiceprompts for serveral queues in our 
asterisk pbx (Version 1.2.41.2.4)


The Problem is, that I don't hear the start (or the first part) of the 
voiceprompt.

It makes no differece if I use the Playback or Background Command.

But it makes a difference if the prompt is played automatically during 
entering the queue ore if I playback the file manually by typing 3 
for example:



exten = s,7,Background(01_ni_asterisk-b)   - missing start part at 
beginning of  prompt
exten = 3,1,Playback(01_ni_asterisk-b)- sound is ok (whole 
soundfile is played)


Here is my extensions_queues.conf

[menu-it]
exten = s,1,Set(LANGUAGE()=de)
exten = s,2,system(/bin/echo ${LANGUAGE}  /tmp/LANGUAGE)
include = sipint
exten = s,3,Answer
exten = s,4,SetMusicOnHold(default)
exten = s,5,Set(TIMEOUT(digit)=3)
exten = s,6,Set(TIMEOUT(response)=16)
exten = s,7,Background(01_ni_asterisk-b)
exten = s,8,Background(queue-menu-announcement)
exten = s,9,SetCallerID(IT Queue: 0${CALLERID})
exten = s,10,Queue(queue-it|t)
exten = s,11,Playback(nbdy-avail-to-take-call)
exten = s,12,Voicemail(u1700) ;
exten = s,13,Set(LANGUAGE()=en)
exten = #,1,Hangup
exten = t,1,Hangup
exten = o,1,Goto(menu-office,s,1)
exten = i,1,Playback(invalid)
exten = i,2,WaitExten
exten = 3,1,Playback(01_ni_asterisk-b)
exten = 4,1,Playback(02_ni_asterisk-b)




Thanks for any help.

morel




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Re: [asterisk-users] problems to call brazil from germany

2006-07-18 Thread Johann Steinwendtner


Sebastian,

This is possible and most likley the reason. To make sure, check the 
location code of the cause IE in your ISDN disconnect message.

You have two options:
1) call your provider and describe your problem.
2) Change your provider

Best regards

Hans

Sebastian Reitenbach schrieb:

Hi,

my problem in short:

I have a problem reaching a co-worker with the asterisk calling from Germany 
to Brazil. With a chance of about 90% I get a chanunavail message from the 
asterisk. Normally I try calling him in the afternoon Germany, when he is 
awake in Brazil.


so I tried to make calls to Brazil from Germany via the Asterisk telephone 
system in the morning, then everything is most likely fine (tried some numbers 
I found via google) when I do this in the afternoon then I get most likely 
a all channels unavalable. When I get the all channels unavalable message, 
then try to call the same number from a mobile, it can reach the number 
without problem.


my theory: I use a cheap preselected carrier to call out via the asterisk, 
that one has not much overseas lines, and therefore they are getting exhausted 
at the afternoon, when there are the people up in Brazil.
when calling at the same time from a mobile phone, then using another carrier 
that has more overseas lines and therefore I can reach the number.


I have no clue whether my theory is fine, or absolutely stupid. Its just out 
of the observations I have made so might be completely wrong.


so any idea whether my theory is right or not, and if not, any other theories?

hope its more clear now.

kind regards
Sebastian



Moises Silva [EMAIL PROTECTED] wrote: 


Callme stupid, but im not understanding your problem. Suggestions that
may help others to answer:

1. A little bit more clear in your examples? :)
2. Try describing the Asterisk behaviour under every circumstance.

Regards

On 7/17/06, Sebastian Reitenbach [EMAIL PROTECTED] wrote:


Hi,

I have problems to call to brazil, frome here in germany. the asterisk is
connected to the telephone system via a pri interface. I use a preselected
provider here to call out.

when I try to call a number in brazil, a mobile phone here in the germany 


in


the afternoon, when it is moring in brazil, then the chances to reach that
number are next to zero. taking a mobile phone and call that number works
fine.

when I try to call someone in brazil, taking numbers found by google, then 


i


can reach a lot of these numbers.

anybody has an explanation for this?
could it be that both carriers have different ways to route the call to 


brazil


and the preselection provider has not so many lines for overseas?



kind regards
Sebastian


--
Sebastian ReitenbachTel.: ++49-(0)3381-8904-451
RapidEye AG Fax: ++49-(0)3381-8904-101
Molkenmarkt 30  e-mail:[EMAIL PROTECTED]
D-14776 Brandenburg web:http://www.rapideye.de

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--
Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org;






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Re: [asterisk-users] ooh323c - cdr

2006-07-18 Thread Richard Scobie




antonio wrote:

I have a problem: when i make i call from a device h323 to sip, i have no
cdr, and i don't see cdr variables for the channnel ooh323.
Anyone can help me ??
Thanx


On my system, this lives in /var/log/asterisk/cdr-csv/ast_h323.csv.

Regards,

Richard
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[Asterisk-Users] Forward call


go here

http://www.voip-info.org/wiki/view/Asterisk+call+forwarding

and look this

*sterisk 1.2*

[macro-stdexten]
; 
; Standard extension macro (with call forwarding): 
; ${ARG1} - Extension(we could have used ${MACRO_EXTEN} here as well 
; ${ARG2} - Device(s) to ring 
; 

---BeginMessage---

Hie,

I trie to use a simply call forward, found on this mailing list (:-), 
when i'm not near my phone:



i creat a global set:
olscell=123456789 ; my cell phone number

A macro for forwarding the call:

[macro-cell_user]
exten = s,1,Playback(Call_Transfer)
exten = s,2,Flash()
exten = s,3,SendDTMF(${ARG1})
exten = s,4,Hangup()


I put in m incoming context:
exten = 0470022762,1,Dial(IAX2/300,20,tr)
exten = 0470022762,2,Macro(cell_user,${olscell})

But, when the call is being, the phone is hangup!
What do i do in macro for forward the call??

Best regards,

--
Olivier Saulnier
STEGANUX
1er étage DIAMECANS
BEL AIR
03410 St-Victor
T: 04.70.02.27.62
F: 04.70.09.97.41
http://www.steganux.com

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---End Message---
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Re: [asterisk-users] IVR DTMF



At /var/lib/asterisk/agi-bin/dtmfivr.sh for example 

After that what should I do 

 
  


read this book?

http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11

this webpage
http://www.voip-info.org/

regards
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[asterisk-users] SIP ATA Channels for outbound calls - How to select in dialplan

2006-07-18 Thread Dean @ INKnBITs

I have setup 3 Linksys SPA-3000 devices to pass/send our analog voice calls
into/out of asterisk. The inbound calls work fine as I have set the
spa-3000's to forward all calls to an extension. I have added them to the
sip.conf as spa-3k1, spa-3k2, and spa-3k3. Is there a way for when some
picks up a phone to dial, it starts at 3k1, if congestion, move onto the
sk2, and so on. I'm looking for it to find the first available line to use.
Is this possible in the dialplan?

Thanks,
Dean.

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Re: [asterisk-users] Call-limit and internal transfer

2006-07-18 Thread random cluster


  Hi,

   Just a suggestion,

Why dont you use GROUP function to limit the calls??

Regards


2006/7/10, alexandre - aldeia digital [EMAIL PROTECTED]:

Hi,

I set the sip.conf parameter call-limit=1 to limit outbound calls and
'disable' call waiting.
But internally, I want to enable transfers. If the call-limit=1, the
transfers fails.

Any help ?

Thanks all,


Alexandre
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[asterisk-users] call forwarding to mobile phone

2006-07-18 Thread Lito Lampitoc

Hello all,Is it possible to forward a call received by the asterisk server to a mobile phone? If yes, how? a link or reference is highly appreciated.thanksLito
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RE: [asterisk-users] call forwarding to mobile phone

2006-07-18 Thread Sam Tam









Get an GSM Gateway from cyber-telecom.net











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lito Lampitoc
Sent: Tuesday, July 18, 2006 4:57
PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: [asterisk-users] call
forwarding to mobile phone





Hello all,

Is it possible to forward a call received by the asterisk server to a mobile
phone? 
If yes, how? a link or reference is highly appreciated.

thanks

Lito






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Re: [asterisk-users] don't hear start/begin of voiceprompts

2006-07-18 Thread [EMAIL PROTECTED]

You need two commands before playing back audio over a line:

Answer()
Wait(2)



On Tuesday 18 July 2006 4:15 am, Mein Name wrote:
 Hi all,

 I just want to setup new voiceprompts for serveral queues in our
 asterisk pbx (Version 1.2.41.2.4)

 The Problem is, that I don't hear the start (or the first part) of the
 voiceprompt.
 It makes no differece if I use the Playback or Background Command.

 But it makes a difference if the prompt is played automatically during
 entering the queue ore if I playback the file manually by typing 3
 for example:


 exten = s,7,Background(01_ni_asterisk-b)   - missing start part at
 beginning of  prompt
 exten = 3,1,Playback(01_ni_asterisk-b)- sound is ok (whole
 soundfile is played)

 Here is my extensions_queues.conf

 [menu-it]
 exten = s,1,Set(LANGUAGE()=de)
 exten = s,2,system(/bin/echo ${LANGUAGE}  /tmp/LANGUAGE)
 include = sipint
 exten = s,3,Answer
 exten = s,4,SetMusicOnHold(default)
 exten = s,5,Set(TIMEOUT(digit)=3)
 exten = s,6,Set(TIMEOUT(response)=16)
 exten = s,7,Background(01_ni_asterisk-b)
 exten = s,8,Background(queue-menu-announcement)
 exten = s,9,SetCallerID(IT Queue: 0${CALLERID})
 exten = s,10,Queue(queue-it|t)
 exten = s,11,Playback(nbdy-avail-to-take-call)
 exten = s,12,Voicemail(u1700) ;
 exten = s,13,Set(LANGUAGE()=en)
 exten = #,1,Hangup
 exten = t,1,Hangup
 exten = o,1,Goto(menu-office,s,1)
 exten = i,1,Playback(invalid)
 exten = i,2,WaitExten
 exten = 3,1,Playback(01_ni_asterisk-b)
 exten = 4,1,Playback(02_ni_asterisk-b)




 Thanks for any help.

 morel




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Re: [asterisk-users] call forwarding to mobile phone

2006-07-18 Thread Lito Lampitoc

is there a way I can do call forwarding to mobile phone without using a gsm gateway? my landline is capable of calling a gsm network.On 7/18/06, Sam Tam
 [EMAIL PROTECTED] wrote:













Get an GSM Gateway from 
cyber-telecom.net











From:
[EMAIL PROTECTED]

[mailto:[EMAIL PROTECTED]
] On Behalf Of Lito Lampitoc
Sent: Tuesday, July 18, 2006 4:57
PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: [asterisk-users] call
forwarding to mobile phone





Hello all,

Is it possible to forward a call received by the asterisk server to a mobile
phone? 
If yes, how? a link or reference is highly appreciated.

thanks

Lito







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[asterisk-users] how to enable users on other iax server call my iax users

2006-07-18 Thread Bobber Cheng


Hi,

I have two iax server, one is asterisk on ip 192.168.18.8, other is 
freeswitch on ip 192.168.18.180. I wanna asterisk iax users could accept 
or call freeswitch iax users. How could i do it in asterisk configure?


Bests,
Bobber Cheng
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Re: [asterisk-users] call forwarding to mobile phone

2006-07-18 Thread Rodrigo Mercado

I need information / documents or configurations of asterisk with other Telephonic head offices(plants), for your help , thank

sorry for my english, i speek spanish only.


atte,Rodrigo M
On 7/18/06, Lito Lampitoc [EMAIL PROTECTED] wrote:

is there a way I can do call forwarding to mobile phone without using a gsm gateway? my landline is capable of calling a gsm network.
On 7/18/06, Sam Tam 
[EMAIL PROTECTED] wrote: 




Get an GSM Gateway from 
cyber-telecom.net





From: 
[EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] ] On Behalf Of Lito LampitocSent: Tuesday, July 18, 2006 4:57 PM
To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [asterisk-users] call forwarding to mobile phone


Hello all,Is it possible to forward a call received by the asterisk server to a mobile phone? If yes, how? a link or reference is highly appreciated.
thanksLito
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[asterisk-users] Asterisk v/s other Telephonic plants

2006-07-18 Thread Rodrigo Mercado


On 7/18/06, Rodrigo Mercado [EMAIL PROTECTED] wrote:


I need information / documents or configurations of asterisk with other Telephonic head offices(plants), for your help , thank

sorry for my english, i speek spanish only.


atte,Rodrigo M

On 7/18/06, Lito Lampitoc 
[EMAIL PROTECTED] wrote: 

is there a way I can do call forwarding to mobile phone without using a gsm gateway? my landline is capable of calling a gsm network.
On 7/18/06, Sam Tam 
 [EMAIL PROTECTED] wrote: 




Get an GSM Gateway from 
cyber-telecom.net





From: 
[EMAIL PROTECTED] [mailto:
 [EMAIL PROTECTED] ] On Behalf Of Lito LampitocSent: Tuesday, July 18, 2006 4:57 PM
To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [asterisk-users] call forwarding to mobile phone


Hello all,Is it possible to forward a call received by the asterisk server to a mobile phone? If yes, how? a link or reference is highly appreciated. 
thanksLito
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[asterisk-users] usage of ast db

2006-07-18 Thread unplug


Hi,
Does anyone know the usage of ast db?
Does ast db will be useless if I use ARA in asterisk?
unplug
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Re: [asterisk-users] Provider UNREACHABLE

2006-07-18 Thread Wilson Pickett


How do I program the dialplan in extensions.conf to:

(a) try multiple provider to make an outgoing call  based on current
latency between my * box and the different providers ?

(b) have if provider 1 goes down, then someone can still call me at
number xxx- but now come in through provider 2 or provider 3 ?


For b) try using DIALSTATUS to determine where to go next. For a) I'm
not sure it's a good idea to route the call based on the latency of a
particular measurement so it would get complicated. Some of the
geniuses on this list may be able to set up an averaging ystem to
predict the reliability of a provider based on its response times
within the last hour or something. Even then, it can't really predict
the future. If computing could do that, we'd all be rich on the stock
market ;)
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Re: [asterisk-users] Provider UNREACHABLE

2006-07-18 Thread Wilson Pickett


(b) have if provider 1 goes down, then someone can still call me at
number xxx- but now come in through provider 2 or provider 3 ?

Oops, misread this one, yes you can have fallthrough numbers but this
must happen at the provider end, not yours.
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Re: [asterisk-users] zaptel on dual processor, How?

2006-07-18 Thread Matt Riddell (NZ)

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Warren (mailing lists) wrote:
 Olivier Picquenot wrote:
 Zeeshan Zakaria a écrit :

  
 It is CentOS 4.3 and kernel is 2.6.9-34.0.1-smp-i686

 Then you might want to use yum to install the apropriate package, the
 one that contains the kernel source, or at the very least the kernel
 headers .
 Or you might grab it on a Cent OS mirror, for exemple:
 ftp://ftp.dedibox.fr/centos/4.3/updates/i386/RPMS/kernel-devel-2.6.9-34.0.1.EL.i686.rpm


 I'm no Cent OS expert, but that should be the right rpm .
 
 The proper method is, as root, type:
 yum install kernel-devel

The problem is, the kernel headers will have the name 2.6.13-15.8
whereas uname -a will report 2.6.13-15.8-smp.

You may need to create a symbolic link.

- --
Cheers,

Matt Riddell
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[asterisk-users] Two security holes fixed in latest versions of Asterisk

2006-07-18 Thread Matt Riddell (NZ)

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

From: http://www.sineapps.com/news.php?rssid=1377

ISS Xforce has published details of two security issues in Asterisk 1.x
which were fixed in the recently release 1.2.10 version.

Asterisk IAX2 Protocol Denial of Service Attack

Summary:

ISS X-Force has discovered a denial of service vulnerability in the
Inter-Asterisk eXchange protocol version 2 (IAX2). IAX2 is used by
Asterisk PBX software to exchange Voice over IP call setup and call
content. If an attacker floods the PBX with call requests, the PBX will
be unable to handle new telephone calls.

IAX2 Protocol Denial of Service Amplification Attack

Summary:

ISS X-Force has discovered a traffic amplification vulnerability in the
Inter-Asterisk eXchange protocol version 2 (IAX2). IAX2 is used by
Asterisk PBX software to exchange Voice over IP call setup and call
content. An attacker can leverage accounts without passwords on an
Asterisk PBX to flood a third party with a large amount of UDP packets.
If the attack is properly constructed the amount of traffic generated
can saturate the victim's Internet connection. Networks do not have to
use Asterisk PBX to be the victim of this kind of traffic flood.

- --
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Matt Riddell
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Re: [asterisk-users] Re: Asterisk 1.2.10 and Zaptel 1.2.7 released!

2006-07-18 Thread Matt Riddell (NZ)

-BEGIN PGP SIGNED MESSAGE-
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trixter aka Bret McDanel wrote:
 On Mon, 2006-07-17 at 19:21 +1200, Matt Riddell (NZ) wrote:
 It will sometimes tell you that there are modules inside
 /var/lib/asterisk/modules which were not compiled for the version you
 are compiling.  If these are not asterisk-addons modules you will likely
 need to remove them.
 
 or modules from others that arent allowed to contribute to
 asterisk-addons or the tree itself for whatever reason, of which I have
 a few of those that have been specifically rejected for inclusion even
 though disclaimers are on file :/
 
 politics at its finest.  At least they work and it appears that some of
 them take less ram and cpu than default asterisk stuffs :)

:)

Which applications exist that have been disclaimed, well coded, are
patent unencumbered and are not accepted?

- --
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Matt Riddell
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[asterisk-users] realtime oracle dialplan select

2006-07-18 Thread René Enskat [Teamware GmbH]




somebody know a good
way howto select datas from * oracle database inside the
extensions?
for mysql there are
functions. are there for oracle similar ways?

regards
rene

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[asterisk-users] Re: Asterisk Database

2006-07-18 Thread Tomislav Parčina

In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
  Where can I find information's about maximum data that I can store in 
  internal * database?
 
 According to the Wiki:
 
 The Asterisk database uses version 1 of the Berkley DB
 
 So, you'd need to look up the information on the Berkeley website, to 
 find it's limitations.

Hi Doug!

Thank you for this information.


--
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Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
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[asterisk-users] Re: asterisk sending connects when it shouldn't

2006-07-18 Thread Tomislav Parčina

In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
 When asterisk receives those messages you hear when calling an 
 unreacheable cellular phone it sends a 'connect' over the terminating 
 PRI line (digium TE410P), making the call seen as billed from customer's 
 perspective.

Yes, this is definitely a problem. I hope there is solution. It could be solved 
if there is AOC (Advice Of Charge) support in Asterisk. But it seams that 
Asterisk developers aren't interested in this.

Please, be interested in developing AOC in Asterisk. That feature will provide 
you correct and accurate billing!



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Mob.: +385(91)1212148
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[asterisk-users] CentOS 4.3 and Zaptel-1.2.7

2006-07-18 Thread Tomislav Parčina

Can someone send me link with instructions how to install Zaptel 1.2.7 on 
CentOS 4.3? So far I have used Fedora Core 4 distribution and I didn't have any 
problems. I'm planning to use CentOS from now on.

Thank you!


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Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
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Re: [asterisk-users] zaptel on dual processor, How?

2006-07-18 Thread Tzafrir Cohen

On Tue, Jul 18, 2006 at 10:20:12AM +1200, Matt Riddell (NZ) wrote:
 -BEGIN PGP SIGNED MESSAGE-
 Hash: SHA1
 
 Warren (mailing lists) wrote:
  Olivier Picquenot wrote:
  Zeeshan Zakaria a écrit :
 
   
  It is CentOS 4.3 and kernel is 2.6.9-34.0.1-smp-i686
 
  Then you might want to use yum to install the apropriate package, the
  one that contains the kernel source, or at the very least the kernel
  headers .
  Or you might grab it on a Cent OS mirror, for exemple:
  ftp://ftp.dedibox.fr/centos/4.3/updates/i386/RPMS/kernel-devel-2.6.9-34.0.1.EL.i686.rpm
 
 
  I'm no Cent OS expert, but that should be the right rpm .
  
  The proper method is, as root, type:
  yum install kernel-devel

kernel-devel-smp ?

 
 The problem is, the kernel headers will have the name 2.6.13-15.8
 whereas uname -a will report 2.6.13-15.8-smp.
 
 You may need to create a symbolic link.

zaptel's makefile looks at /lib/modules/`uname -r`/build
by default. The kernel headers package also includes that link, IIRC.

-- 
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755  iax:[EMAIL PROTECTED]
+972-50-7952406  jabber:[EMAIL PROTECTED]
[EMAIL PROTECTED] http://www.xorcom.com
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Re: [asterisk-users] realtime oracle dialplan select

2006-07-18 Thread Filip Drągowski





AGI() using php, perl, c, bash, python or almost enything You like.

  
  
  somebody
know a good way howto select datas from * oracle database inside the
extensions?
  for
mysql there are functions. are there for oracle similar ways?
   
  regards
rene





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Re: [asterisk-users] Provider UNREACHABLE

2006-07-18 Thread Chris Mason (Lists)


Wilson Pickett wrote:

How do I program the dialplan in extensions.conf to:

(a) try multiple provider to make an outgoing call  based on current
latency between my * box and the different providers ?

To do this, you need a seperate application that would run something 
like fping on all your termination providers and would rate your 
providers. The an AGI would read the table and choose the providers in 
order of rating and route the call according. It would be relatively 
trivial to construct, whether it would work well is another matter, as I 
am sure we can have many discussions on what makes for the best route.
Another problem is that some providers will turn off the ability to ping 
the server, Teliax did that recently and I have to make a change to my 
monitoring setup.
My solution is to run smokeping to keep a watch on my routes, then rate 
those manually in the dialplan by using CHanIsAvail like this:


exten = s,n,ChanIsAvail(IAX2/teliaxIAX2/nufoneIAX2/sellvoip)

This way, I choose the historically highest quality provider first but 
roll over if they are down.


--
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(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 



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Re: [asterisk-users] Re: Asterisk 1.2.10 and Zaptel 1.2.7 released!

2006-07-18 Thread trixter aka Bret McDanel

On Tue, 2006-07-18 at 06:02 +1200, Matt Riddell (NZ) wrote:
 :)
 
 Which applications exist that have been disclaimed, well coded, are
 patent unencumbered and are not accepted?

res_js for example, which in my experience on a more or less fair
comparison (the javascript dialplan has more error control, better error
checking, and slightly more functionality but other than that it does
the same stuff) uses LESS ram and LESS cpu on the same hardware with the
same asterisk version when compared to extensions.conf dialplan
processing.  

That is just one example of something that could easily be placed in
asterisk-addons, was disclaimed, and wasnt wanted.  It has no patent and
the license for the code it uses is mozilla spidermonkey which depends
on the nspr stuff, both of which are tri-licensed - gpl,lgpl and mpl
(more like BSD).  

There are more examples, but this is one that doesnt break.  


-- 
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Belfast IE +44 28 9099 6461DE +49 801 777 555 3402
Utrecht NL +31 306 553058  US WA +1 360 207 0479
US NY +1 516 687 5200  FreeWorldDialup: 635378
http://www.trxtel.com the VoIP provider that pays you!


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[asterisk-users] GSM Module not picking up DTMF digits from VOIP FXO Gateway

2006-07-18 Thread Levis Kimotho

Hi,I've connected a GSM gateway to a WellTech 3702A FXO port and created an extension in Asterisk which mobile calls are forwarded to. When a call is made, the GSM module generates a tone instead of picking the digits from the Voip gateway and processing the calls. The VOip gateway keypad type is RFC2833.
When the GSM gateway generated the tone, i can then type and the call goes thru. Why is the GSM module not getting the digits automatically as passed from Asterisk via WellTech 3702A gateway?
-Levis


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Re: [asterisk-users] Provider UNREACHABLE

2006-07-18 Thread Chris Mason (Lists)




(b) have if provider 1 goes down, then someone can still call me at
number xxx- but now come in through provider 2 or provider 3 ?
The way to do this is to use a PSTN based DID provider such as Kall8 and 
use a rollover list to route to your DID provider. If your VOIP provider 
goes dead, you can log into Kall8 and change the forwarding number.


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Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 



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[asterisk-users] CDR related issue

2006-07-18 Thread dashy dude

Hi all
I am facing a strange problem related to CDRs.

I am using asterisk 1.2.4 and AMP

When I setup a 3 way call from  a phone, the CDRs are
generated quite strange.

e.g. Phone A calls phone B and Phone C
in CDRs it should appear that Phone A called phone B
and Phone A called phone B
Instead, it happens that there are 2 CDR records with
Phone B calling Phone C

Has anyone faced this?



Thanks
D

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[asterisk-users] Parked calls

2006-07-18 Thread harrygaillac-sip

Hello everybody,

I is possible to manage multiple call parked per line
.
I mean a caller (agent) have to park more than two
call . It is possible to retrieve caller one ,two
,three, ... with a aplliction which one display the
calling parked to a PC screen or a screen phone .

Regards

Harry 








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Re: [asterisk-users] LinkSys SPA 2002 ATA hardphone UNREACHABLE...!!!

2006-07-18 Thread Rich Adamson


voiplist wrote:

On 7/17/06, Luki [EMAIL PROTECTED] wrote:

 We have 6 or 7 SPA-2000's which all work with other installs of
 Asterisk but can't get a single one to receive calls using Asterisk
 1.2.4.

Ha! You're right. I just got some too and didn't even think of testing
the ringer. Outgoing calls work fine, but incoming calls say Call 1
State: Ringing on the web interface and the call details are
displayed but the phone does not ring. It obviously gets the SIP
message that it should ring but it does not. Asterisk CLI also
confirms that device is ringing. Increasing the ring voltage did not
help either.

Needless to say the same phone works fine with SPA 1000, 1001 and 
Grandstream.


Interesting... any ideas what the heck is up with that? This is
software version 3.1.9(LSa). I can't upgrade the software because the
unit thinks it's not idle and hence does not start the upgrade
process. Kind of disappointing.

--Luki


Not sure this is exactly the problem we have, our call gets rejected
by the device for some odd reason. Not 100% sure at this point because
it's been a while, I am going to do more testing in a few minutes I
think.

Can those having trouble confirm their Asterisk version? The version I
am having issues with is 1.2.4, I have a slightly older version of
Asterisk which rings these ATAs just fine..


Have you tried using ethereal to look at the sip packet contents?

It would be interesting to see what the packet differences are for 
v1.2.4, v1.2.10, etc.


Best guess is that recent changes in sip may be constructing the packets 
in slightly different ways, and the spa boxes may be sensitive to those. 
If that guess is correct, not sure a sip debug would reflect the same 
level of detail contained in an ethereal trace.




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Re: [asterisk-users] LinkSys SPA 2002 ATA hardphone UNREACHABLE...!!!

2006-07-18 Thread Patrick

On Mon, 2006-07-17 at 21:18 -0700, Luki wrote:
  We have 6 or 7 SPA-2000's which all work with other installs of
  Asterisk but can't get a single one to receive calls using Asterisk
  1.2.4.
 
 Ha! You're right. I just got some too and didn't even think of testing
 the ringer. Outgoing calls work fine, but incoming calls say Call 1
 State: Ringing on the web interface and the call details are
 displayed but the phone does not ring.
[snip]

Assuming you are talking about an analog phone hooked up to the ATA, I
remember reading somewhere that you may need to increase the voltage to
the phone. Try google.

Regards,
Patrick

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Re: [asterisk-users] LinkSys SPA 2002 ATA hardphone UNREACHABLE...!!!

2006-07-18 Thread Rich Adamson


Patrick wrote:

On Mon, 2006-07-17 at 21:18 -0700, Luki wrote:

We have 6 or 7 SPA-2000's which all work with other installs of
Asterisk but can't get a single one to receive calls using Asterisk
1.2.4.

Ha! You're right. I just got some too and didn't even think of testing
the ringer. Outgoing calls work fine, but incoming calls say Call 1
State: Ringing on the web interface and the call details are
displayed but the phone does not ring.

[snip]

Assuming you are talking about an analog phone hooked up to the ATA, I
remember reading somewhere that you may need to increase the voltage to
the phone. Try google.


If I recall correctly, the OP said he already tried that and it didn't 
impact the issue.


Almost sounds like the code for Do Not Disturb was entered by someone, 
but I don't think the spa's remember those settings after a reboot.


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Re: [asterisk-users] problems to call brazil from germany

2006-07-18 Thread Sebastian Reitenbach

Hi,

Johann Steinwendtner [EMAIL PROTECTED] wrote: 
 Sebastian,
 
 This is possible and most likley the reason. To make sure, check the 
 location code of the cause IE in your ISDN disconnect message.
I have a PRI interface, here ISDN with 30 channels. I am a bit unsure what you 
mean with the location code, for what shall I look for exactly? I was logged 
in to asterisk cli while calling, and for me it looked just like i had the 
wrong number.


 You have two options:
 1) call your provider and describe your problem.
done that, they will monitor the line, and probably see what happens.

 2) Change your provider
if nothing other helps, I'll do that.



 
 Best regards
thanks

Sebastian

  
  my problem in short:
  
  I have a problem reaching a co-worker with the asterisk calling from 
Germany 
  to Brazil. With a chance of about 90% I get a chanunavail message from 
the 
  asterisk. Normally I try calling him in the afternoon Germany, when he is 
  awake in Brazil.
  
  so I tried to make calls to Brazil from Germany via the Asterisk telephone 
  system in the morning, then everything is most likely fine (tried some 
numbers 
  I found via google) when I do this in the afternoon then I get most likely 
  a all channels unavalable. When I get the all channels unavalable 
message, 
  then try to call the same number from a mobile, it can reach the number 
  without problem.
  
  my theory: I use a cheap preselected carrier to call out via the asterisk, 
  that one has not much overseas lines, and therefore they are getting 
exhausted 
  at the afternoon, when there are the people up in Brazil.
  when calling at the same time from a mobile phone, then using another 
carrier 
  that has more overseas lines and therefore I can reach the number.
  
  I have no clue whether my theory is fine, or absolutely stupid. Its just 
out 
  of the observations I have made so might be completely wrong.
  
  so any idea whether my theory is right or not, and if not, any other 
theories?
  


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[asterisk-users] Other phone continues to ring when pick up a call with *8 on SVN HEAD

2006-07-18 Thread Chris Stenton

I've just found that picking up another phones call via *8#  gives me the call 
but the other phone keeps ringing. Anyone else seeing this on svn head (updated 
last Sunday).

Chris
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Re: [Asterisk-Users] Question setting up a bat phone extension.

2006-07-18 Thread Eric \ManxPower\ Wieling


Kai Ober wrote:


Has somebody done that with a Grandstream  GXP-2000 or a BudgetTone 
100/101 ?

Has somebody even a list which SIP phones have this funtion?


SIPura supports it, Cisco ATAs support it.  I assume that Cisco phones 
support it.


I don't know about Grandstream devices since they are banned from our 
network.


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Re: [asterisk-users] polycom 601 manual config?

2006-07-18 Thread Eric \ManxPower\ Wieling


Shaun wrote:
Is there not a way to manually configure these phones or at least configure 
them to use a diffrent tftp server rather than it attempting to ask the 
dhcp/bootp server?  For users at home with dinky linksys/dlink modems you 
cant set a tftp/bootp server.




Of course there is.  You do it on the phone.

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[asterisk-users] Called party cannot hear caller

2006-07-18 Thread Giorgio Incantalupo


Hi,
I'm using Asterisk 1.2.1 on a Debian distro. It happens that sometimes 
the caller cannot hear the called party just after the called picks up 
the phone. This happens with inbound calls but also with calls from a 
SIP phone to another SIP phone. Asterisk and all the SIP phones are on 
the same net 192.168.1.x.


Is there anybody who has experienced and solved this problem?

TIA


Giorgio Incantalupo
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[asterisk-users] ooh323c - cdr problem

2006-07-18 Thread antonio


The configuration  is this:


H323 --  ASTERISK  ---  SIP


ooh323.conf

amaflags = billing
[xxx.xxx.xxx.xxx]
type=friend
context=h323-route
ip=xxx.xxx.xxx.xxx  
port=1720
allow=all
h323id=example
accountcode=5698742
rtptimeout=60
dtmfmode=rfc2833


extension.conf
...
[h323-route]
exten = 101,1,Answer()
exten = 101,2,Dial(SIP/101)
exten = 101,3,Hangup()

sip.conf
[101]
username=101
type=friend
record_out=Adhoc
record_in=Adhoc
qualify=no
port=5060
nat=never
[EMAIL PROTECTED]
host=dynamic
dtmfmode=rfc2833
context=from-internal-toh323
canreinvite=no
callerid=101

Why there are not cdr for these calls ??
If i make a call from sip to h323, the cdr is generated.
Thanx for the help 



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Re: [Asterisk-Users] Question setting up a bat phone extension.

2006-07-18 Thread Gonzalo Servat


On 6/11/06, James Harper [EMAIL PROTECTED] wrote:
[.snip.]

My dialplan in the pap2 is:

(:0S0)

Which causes it to dial a '0' to asterisk as soon as I gets picked up.
In my asterisk dialplan it then does a DISA to another context, which
means Asterisk is doing all the dialplan stuff. For what I want in a
dialplan, I could have configured it in the pap2 but I didn't want to
learn it. I think I'm at that age where everything new I learn means
something else gets overwritten :)

[.snip.]

This is pretty cool! Thanks James. Now I just keep the Asterisk
dialplan configured and can leave the PAP2 dialplan untouched. The
only functionality I'd loose is the ability to use the *xx codes for
the PAP2. right?

Cheers,
Gonzalo
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[asterisk-users] Reinvite and NAT - Problem

2006-07-18 Thread Roger Schreiter


Hi,

I have following setup:

++ ++
| asterisk A |-| asterisk B |-- PSTN-gateways ...
++ ++
  | .|
  |  .
=Router (NAT)=.==
  |.
++  .
| SIP phone  |-
++


asterisk A should do registration and call setup ...,
and asterisk B should handle the media.
Thus asterisk A should reinivite SIP phone and asterisk B
on any call.

I have asterisk 1.0.10, and on asterisk A the users are
stored in mysql/sipfriends, which works fine. I already
bugfixed in the source code, that chan_sip ignores the canreinvite-
setting from sip.conf, and now calls from SIP phone to the
PSTN gateway work perfekt:

Reinvite occours, and using tcpdump I can see, that after
call setup IP traffic is only between the router and asterisk
B, but no more between router and asterisk A (besides hangup).


The problem occours in the other direction, PSTN gateway
to SIP phone:

asterisk B is calling asterisk A, and than asterisk A is
calling the SIP phone, as intended.
Also as intended reinvite is taking place.
But unfortunately, asterisk B is addressing the private
(to be NATed) IP address of the SIP phone.
Thus, audio data are flowing from SIP phone to asterisk B, but
no audio data are flowing from asterisk B to SIP phone.
The NAT workaround of asterisk is not working as desired.

I assume, with a little source code modification the problem
would be solved (like the sipfriends/canreinvite problem).

Unfortunately I do not understand, who has to care about the
NAT workaround.
Is it asterisk A, who has to tell the right (SIP phones public) IP
address to asterisk B
(i.e. the one, where it gets IP traffic from instead of the one
SIP phone tells),
or is it asterisk B, who has to ignore the IP address, which SIP
phone tells, but has to take the IP address, where traffic
is coming from?


Please explain how reinvite with NAT workaround should work!


Roger.

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Re: [asterisk-users] link quality is poor

2006-07-18 Thread Alex Robar

Define poor link quality. Are you seeing latency delays, packetization... ?AlexOn 7/18/06, Alexandr Bondar 
[EMAIL PROTECTED] wrote:Hello.Problem is: Current configuration: (PSTN (UkrTeleCom)) - Е1 -
(TE210P) - T1 - (My own Lucent *MAX 4000*).I am testing different modem calls: (My own Lucent *MAX 4000*) - T1 - TE210P - T1 - (My own Lucent*MAX 4000*) - we have maximum quality of modem connection
 (My own Lucent *MAX 4000*) - T1 - TE210P - Е1 - (PSTN(UkrTeleCom)) - (Access Server UkrTeleCom) - we have maximum quality ofmodem connection (My own Lucent *MAX 4000*) - T1 - TE210P - Е1 - (PSTN
(UkrTeleCom)) - Е1 - TE210P - T1 - (My own Lucent *MAX 4000*) - linkquality is poorHow can we solve this problem?___--Bandwidth and Colocation provided by 
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Re: [Asterisk-Users] Question setting up a bat phone extension.


Eric ManxPower Wieling schrieb:


I don't know about Grandstream devices since they are banned from our 
network.
Banned? I didn't try any other devices, but whats wrong with the 
Grandstreams??



wondering

Kai



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Re: [Asterisk-Users] Question setting up a bat phone extension.

2006-07-18 Thread Eric \ManxPower\ Wieling


Kai Ober wrote:

Eric ManxPower Wieling schrieb:


I don't know about Grandstream devices since they are banned from our 
network.
Banned? I didn't try any other devices, but whats wrong with the 
Grandstreams??


Grandstream seems unable to produce stable firmware.  They have tried 
for *YEARS* and still people have to try many different versions of the 
firmware to find one that actually works in their environment.


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Re: [Asterisk-Users] Question setting up a bat phone extension.


Eric ManxPower Wieling schrieb:


Grandstream seems unable to produce stable firmware.  They have tried 
for *YEARS* and still people have to try many different versions of 
the firmware to find one that actually works in their environment.



okay, i see, thx :)
i will try to remember, if  i'm ever going to buy an VoIP-Phone.
any suggestions for this situation? (i.e. which devices do you prefer)

Kai



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Re: [asterisk-users] CentOS 4.3 and Zaptel-1.2.7

2006-07-18 Thread tracinet

Below is what I have in my notes regarding CentOS. I am not sure exactly where I originally found this - could have been right from this mailing list actually!

Patch CentOS
spinlock.h file before installing zaptel

Rebuilding Zaptel - Every
time there is a kernel update with yum (which is the case here), ZAP device
support needs to be rebuilt using the new kernel.

Unfortunately,
a RedHat bug caused the rebuilding process to fail. Here's the fix. Log into
your new server as root and issue the following commands:



uname -r (to findout what kernel you are running)cd /usr/src/kernels/'uname -r'/include/linux

mv spinlock.h spinlock.h.old
wget http:// nerdvittles.com/aah27/spinlock.hThen install zaptel as normalHope that helps!
On 7/18/06, Tomislav Parčina [EMAIL PROTECTED] wrote:
Can someone send me link with instructions how to install Zaptel 1.2.7 on CentOS 4.3? So far I have used Fedora Core 4 distribution and I didn't have any problems. I'm planning to use CentOS from now on.Thank you!
--Tomislav ParčinaLama Computers SplitStinice 12, 21000 SplitTel.: +385(21)495148Mob.: +385(91)1212148SIP: [EMAIL PROTECTED]e-mail: tparcina#lama.hr
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[asterisk-users] External call press 1

2006-07-18 Thread carl Lougher

Hi,
Running asterisk ver 1-0-9

Trying to send a call to a mobile phone and playback a
message to the user to press one to accept the call. 
If 1 isn't pressed then the call needs to be re-routed
back into the asterisk dialplan.

Tried various macros etc but if one isn't pressed the
call still gets accepted?

Any clues???

exten = 333,1,Macro(test)
exten = 333,2,Hangup

exten = 334,1,Dial(SIP/XXX)


[macro-test]
exten = s,1,Wait(1)
exten = s,2,Read(ACCEPT|press-one |1)
exten = s,3,GotoIf($[${ACCEPT} = 1 ]?4:5)
exten = s,4,NoOp(Caller accepted)
exten = s,5,Goto(client,334,1)

exten = i,1,Set(MACRO_RESULT=CONTINUE)
exten = t,1,Set(MACRO_RESULT=CONTINUE)




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Re: [asterisk-users] zaptel on dual processor, How?

2006-07-18 Thread Warren (mailing lists)

Matt Riddell (NZ) wrote:
 Warren (mailing lists) wrote:
The proper method is, as root, type:
yum install kernel-devel
 
 
 The problem is, the kernel headers will have the name 2.6.13-15.8
 whereas uname -a will report 2.6.13-15.8-smp.
 
 You may need to create a symbolic link.

Sorry...
yum install kernel-smp-devel

You will then have a directory of /usr/src/kernels/2.6.13-15.8-smp-i686
(or somthing like that).  You might want to create a link to
/usr/src/linux from there to keep things simple.

W
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[asterisk-users] Asterisk Trunk Name Problem

2006-07-18 Thread Ondrej P.


Hey guys.
I have a little problem. My provider requires me to make the trunk name 
of my SIP connection i2telecom.com (otherwise it won't register). 
Unfortunately, this name also becomes the identifier for the connection. 
Now, when I want to dial through it Asterisk think I am trying to dial 
through the domain i2telecom.com and not the actual connection.
I tried both, Dial(SIP/i2telecom.com/5551234) and 
Dial(SIP/[EMAIL PROTECTED]). But neither one works.


Is there anything I can do?

Thanks,
Andrew
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Re: [asterisk-users] show channels

2006-07-18 Thread Moises Silva


The show channels output is always truncated.

On 7/17/06, marek cervenka [EMAIL PROTECTED] wrote:

hi,

i have problem with showing actual channels

asteriskshow chanels
SIP/123456789-b6c4b2 [EMAIL PROTECTED] Up  Busy()
(last 2 chars are NOT showed)

but the name of channel is longer
asterisk show channel SIP/123456789-b6c4b290

how can i get full name of channel with asterisk -rqnx ?

thanks

---
Marek Cervenka
===

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[Asterisk-Users] GSM gateway flooded ce ll - how to detect?

2006-07-18 Thread Colin Anderson

We are using an Ateus VoiceBlue to GSM gateway calls on our * 1.0.9 server.
It works perfectly fine, except at peak periods, say, 10 AM and 3 PM. At
that point, calls get dropped (not gateway'd) and Asterisk jumps to the next
priority in the dialplan. Our interpretation of this is that the local GSM
cell is flooded with other calls and can't service our request, so nothing
to to with Asterisk or the gateway. No matter how hard we try, during
off-hours, we can't replicate this behavior. My question is how to detect
this behavior and relay the call out to our PRI instead. I've had a couple
of ideas so far, but nothing has panned out:

1. Use the ${DIALSTATUS} variable, however when the condition occurs, the
variable is set to NOANSWER which is the same setting if the guy doesn't
pick up his phone, so it does me no good, since I can't correctly detect
whether it is the gateway or whatever. Maybe an AGI which sets a timer to
detect ringtime? More information: This is different than if the gateway is
full and can't service the request, which I am already successfully testing
for before the dialplan makes the determination to use the gateway in the
first place or not.

2. Dial the target cell using the gateway and the PRI simultaneously, so
this masks the condition. If the gateway kacks, then the call would still go
through the PRI to the target cell.  This would work, however I am using the
'r' option to dial, in order to detect early audio if the user has his cell
off to advance the dialplan. When I do this, and the user answers, the PRI
channel gets an early-audio indicator from the GSM provider (The person you
are calling can't answer blablabla ), and Asterisk drops to the next
priority in the dialplan, which I do *not* want to do, until the user has
hung up or doesn't answer. Getting rid of the 'r' option is not in the
cards. 

Another idea which just occured to me is to physically move the gateway to
another location a few km away that we have a VPN tunnel to, and just route
calls over there - another cell, maybe not so saturated, right? The danger
there is that the gateway is not on the LAN so the side effect is that our
infrastructure becomes more fragile i.e. if the VPN is down the gateway
doesn't work. Still, I think it's worth a try.

Anybody have any spitballs about how to work around this issue? When the
gateway works, it saves is $2-4K a month in airtime, so I definitely don't
want to abandon it. My GSM provider (Rogers) could care less about working
with me to address this, since it is more revenue for him. 

tia
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RE: [asterisk-users] Hitting # to Transfer out of a Queue

 -Original Message-
 From: Nic Bellamy [mailto:[EMAIL PROTECTED]
 Sent: Monday, July 17, 2006 11:31 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Hitting # to Transfer out of a Queue

 Douglas Garstang wrote:

 Why? I don't want the variable to be global, or inherit to 
 other channels. I only want it to be persistent for the 
 current call in progress.

 It'll only inherit to channels *created from this one*, eg. Agent 
 channels, Local channels and the like.

 It doesn't make it a global variable - see doc/README.variables for 
 further information.

Nic,

Still no luck with this. I have:

exten = oe_ccare,1,NoOp(Queue oe_ccare called)
exten = oe_ccare,n,Set(TIMEOUT(response)=5)
exten = oe_ccare,n,
GotoIfTime(8:00-17:30|mon-fri|*|*?one_queue_acd,oe_ccare-open,1)
exten = oe_ccare,n,Goto(oe_ccare-shut,1)
exten = oe_ccare-open,1,   Answer
exten = oe_ccare-open,n,   Set(__TRANSFER_CONTEXT=one_start)
exten = oe_ccare-open,n,   NoOp(${__TRANSFER_CONTEXT})
exten = oe_ccare-open,n(queue1),   Queue(oe_custcare30)

and I'm still getting this console message when someone hits a digit...

Jul 18 08:27:37 VERBOSE[26274] logger.c: -- Unable to find extension '4' in 
context ''

Don't know why the context is '', null.

Douglas.
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[asterisk-users] PAP2 TUI Configuration Menu

2006-07-18 Thread Jamin W. Collins

Does anyone know of a way to disable access to the TUI interface 
(accessed via ) on the PAP2 devices?  I'm looking at using these 
devices for lobby and door phones and would like to remove/disable the 
TUI interface if at all possible.


--
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Re: [asterisk-users] Hitting # to Transfer out of a Queue

2006-07-18 Thread Doug Lytle


Douglas Garstang wrote:

Jul 18 08:27:37 VERBOSE[26274] logger.c: -- Unable to find extension '4' in 
context ''

Don't know why the context is '', null.
  


Silly question,

Has a context been defined in the queues.conf?

; A context may be specified, in which if the user types a SINGLE
; digit extension while they are in the queue, they will be taken out
; of the queue and sent to that extension in this context.
;
;context = qoutcon

Doug

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deserve neither Liberty nor Safety.


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Re: [asterisk-users] don't hear start/begin of voiceprompts

2006-07-18 Thread Mein Name


Hi,

[EMAIL PROTECTED] wrote:


You need two commands before playing back audio over a line:

Answer()
Wait(2)

 



thanks a lot for answering!  This solves my problem perfectly.

ciao,
morel
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Re: SV: [Asterisk-Users] Nokia E61

2006-07-18 Thread Dinesh Nair




On 07/18/06 04:03 Fredrik Emil Jensen said the following:

the packet too, but when the firewall/router loses its table (usually it
will timeout after xx sec/min) you will only be able to dial outgoing


can't you use qualify to get the nat device to keep the mapping ?

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| for a in past present future; do|
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|   echo The opinions here in no way reflect the opinions of my $a $b.  |
| done; done  |
+=+
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RE: [asterisk-users] Hitting # to Transfer out of a Queue

 -Original Message-
 From: Doug Lytle [mailto:[EMAIL PROTECTED]
 Sent: Tuesday, July 18, 2006 8:43 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Hitting # to Transfer out of a Queue

 Douglas Garstang wrote:
  Jul 18 08:27:37 VERBOSE[26274] logger.c: -- Unable to 
 find extension '4' in context ''

  Don't know why the context is '', null.

 Silly question,

 Has a context been defined in the queues.conf?

 ; A context may be specified, in which if the user types a SINGLE
 ; digit extension while they are in the queue, they will be taken out
 ; of the queue and sent to that extension in this context.
 ;
 ;context = qoutcon

Doug,

This is not the same thing. The 'context' parameter in queues.conf is used to 
allow a caller, while waiting in a queue, to dial an extension and be taken 
somewhere else. That's not what I am trying to do. I am trying to get Asterisk 
assisted transfers to work with Queues. That is, someone dials into a queue, 
the AgentCallBackLogin() function calls the agent, and the agent wants to 
transfer the caller somewhere else. Quite different. :)

We'd use SIP transfers initiated from the phones, but this seems to cause 
Asterisk to completely lock up. See these bugs:
http://bugs.digium.com/view.php?id=7458 
http://bugs.digium.com/view.php?id=6626 
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Re: [asterisk-users] CentOS 4.3 and Zaptel-1.2.7

2006-07-18 Thread varun

I have problems compiling zaptel 1.2.6 on my CentOS 4.3. CentOS is
updated and I believe I have installed all the dependencies.

did you fix spinlock.h?

Go into your kernel source directory(or directories if you have more 
than one kernel source on your system) and edit the file spinlock.h
Then goto line 407

Change this line from :

#define DEFINE_RWLOCK(x) rw_lock_t x = RW__LOCK_UNLOCKED
To:
#define DEFINE_RWLOCK(x) rwlock_t x = RW__LOCK_UNLOCKED 

Save the changes to all copies of this file you have, after these
changes have been made your zaptel drivers should compile just fine.

OR

http://aussievoip.com.au/wiki/freePBX-Centos

OR

www.voip-info.org/wiki/view/Asterisk+Zaptel
+Installationview_comment_id=11286

Varun


On Tue, 2006-07-18 at 12:30 +0200, Tomislav Parčina wrote:
 Can someone send me link with instructions how to install Zaptel 1.2.7 on 
 CentOS 4.3? So far I have used Fedora Core 4 distribution and I didn't have 
 any problems. I'm planning to use CentOS from now on.
 
 Thank you!
 
 
 --
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 Stinice 12, 21000 Split
 Tel.: +385(21)495148
 Mob.: +385(91)1212148
 SIP: [EMAIL PROTECTED]
 e-mail: tparcina#lama.hr
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Re: [asterisk-users] Setting a threshold for asterisk to take ZAP line off hook ?

I think when a PSTN line says 'Ring' it's simply for aesthetics... The 
line is 'answered' the instant * connects to it for two-way audio... 
(well not that instant but somewhere in the connection process.  When 
you are hearing ringing from the PSTN through a zap card, the rings are 
coming from the phone company and are just sound.  * doesn't decode that 
and act on it yet.)


Maxim Vexler wrote:

On 7/16/06, Martin Joseph [EMAIL PROTECTED] wrote:

On Jul 16, 2006, at 11:36 AM, Maxim Vexler wrote:


Hello list

I'm trying to setup asterisk as an answering machine.

How can I set asterisk to Answer() incoming call ONLY after specified
count of ring cycles ?

In the current situation I have the PBX connected to a home line,
where POTS device are also connected on the same circuit. What I'm
trying to do is allow a grace period where a POTS device could be
picked up and those stop the ring indication on the line by this
causing asterisk to not answer the call.

In present situation even if the incoming phone call is taken off hook
by a POST device asterisk still starts playing its incoming call IVR
after the specified(where?) number of seconds.


I don't think you can do that, since asterisk has no way to know when
the shared PSTN line is answered by your analog phones...

I don't think asterisk counts the rings, as much as it waits for
answered status, which it is never going to see in your current
configuration.

I am a relative newb though,  so maybe someone else here has a
brilliant idea for you?


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You have a point but no way am I going to accept that as an answer.

Here's the log off such case :

Jul 16 21:59:20 DEBUG[4387] chan_zap.c: Monitor doohicky got event
Ring Begin on channel 1
Jul 16 21:59:21 DEBUG[4387] chan_zap.c: Monitor doohicky got event
Ring/Answered on channel 1
Jul 16 21:59:21 DEBUG[4387] dsp.c: dsp busy pattern set to 0,0
Jul 16 21:59:21 VERBOSE[4411] logger.c: -- Starting simple switch
on 'Zap/1-1'
Jul 16 21:59:21 DEBUG[4373] devicestate.c: Changing state for Zap/1 -
state 2 (In use)
Jul 16 21:59:21 DEBUG[4412] app_queue.c: Device 'Zap/1' changed to
state '2' (In use)
Jul 16 21:59:29 WARNING[4411] chan_zap.c: CallerID returned with error
on channel 'Zap/1-1'
Jul 16 21:59:29 DEBUG[4411] pbx.c: Launching 'Answer'
Jul 16 21:59:29 VERBOSE[4411] logger.c: -- Executing
Answer(Zap/1-1, ) in new stack
Jul 16 21:59:29 DEBUG[4411] chan_zap.c: Took Zap/1-1 off hook
Jul 16 21:59:29 DEBUG[4411] chan_zap.c: Enabled echo cancellation on channel 1
Jul 16 21:59:29 DEBUG[4411] chan_zap.c: No echo training requested
Jul 16 21:59:29 DEBUG[4411] pbx.c: Launching 'Set'

As you can see, the first two events are event Ring and event Ring/Answered.
What I need is the driver of chan_zap.c counting 5 event Ring before
starting Ring/Answered.

It can't be that hard (I think).
Thank you for your answer.



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Re: [asterisk-users] Injecting prerecorded audio into active call

Yes, use a web interface to re-generate features.conf on the fly?  hmm.. 
then tell asterisk to reload it.  (Should be OK if the web interface has 
some manner of a mutex to keep other instances of the web interface from 
stomping on its features.conf and if reload res_features.so does what 
you want it to)




Matt Riddell (NZ) wrote:

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Nick wrote:

Yeah a bit messy I guess. I had been hoping for a simple solution, but
knew there most likely wasn't!

The one idea I did have would be to use some kind of SIP api on the web
backend. Then bring the backend extension into a conference, then from
the web api you would have to select the call to play audio in.

This idea would work well I think, as it would mean the system can be
use regardless of the training call being active on the asterisk box, as
long as their system supported conference calls.

This is where I fall down though, I'm no developer! Anyone know of an
api that would allow this?


If you don't mind the call centre staff member pressing some buttons to
request help in the middle of the call you could use a featuremap using
features.conf and the playback application.

- --
Cheers,

Matt Riddell
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-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.2 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iD8DBQFEuzwjS6d5vy0jeVcRAuOLAJ9xBWUKiuFN2yLqxnnsYIXqig2XMQCfchOu
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!DSPAM:500,44bb3d1b139628513218385!



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[asterisk-users] Buch Bekanntmachung: Der Weg zu VoIP Asterisk von A bis Z

2006-07-18 Thread Silvio Schneider



Buch Bekanntmachung: Der Weg zu VoIP Asterisk von A bis Z

For english see bellow.

Als ich Oktober 2005 angefangen habe mit Asterisk zu arbeiten, gab es nur wenig 
zusammenhängende Informationen zu Asterisk. Es gab bereits ein Buch zu 
Asterisk, jedoch wurden dort einige Themen ausgelassen. Auch im neuen Asterisk 
von Oreilly wird nur Asterisk besprochen, jedoch nicht auf Telefonkarten, VoIP 
Telefone, Zusatzprogramme eingegangen.

Der Buch Inhalt ist auf die deutschsprachigen Länder, Deutschland, Österreich 
und die Schweiz, bezogen.

Das Buch Der Weg zu VoIP Asterisk von A bis Z beschreibt nicht nur Asterisk 
sondern auch viele Dinge, welche mit Asterisk verwendet werden können. Hier 
einige Beispiele aus dem Buchinhalt.

- Einführung in Asterisk
- Benötigte Hardware für Asterisk
- Asterisk installation
--  Das erste Gespräch
--  Echo Call test
--  die sprechende Uhr

- die Konfigurationsdateien
- Sicherheit

- Anrufbeantworter
- Sprachausgabe auf deutsch wechseln
- makeln, weiterleiten
- Music on Hold
- Automatisch weiterleiten

- PSTN(POTS) integration
- ISDN integration

- SIP Telefone
- Sprachcodecs
- Asterisk Konfigurationsprogramme
- Asterisk und Billing
- Rollout in einem Unternehmen
- Alternativen zu Asterisk
- VoIP Protokolle
- ...

Das Buch ist bereits teilweise frei erhältlich im Internet verfügbar.
Die ersten 30 Seiten finden man jetzt schon unter 
http://www.suvi.org/theory/asterisk.html

Jede Woche, oder beim Verkauf eines Buch Exemplares, wird eine Seite 
freigegeben. Das Buch wird also Zeit abhängig freigegeben. Zudem unterstützt 
man mit dem Kauf des Buches die freie Erhältlichkeit des Buches.

Das Buch hat rund 243 Seiten und kann unter 
https://www.lulu.com/commerce/index.php?fBuyContent=359309  bezogen werden.

Gruss
Silvio

Book announcement: Der Weg zu VoIP Asterisk von A bis Z

First at all: The book is only aviable in the german language. The book does 
describe Asterisk as well as things that are usefull for Asterisk: 
VoIP-Telefons, Configuration Software, Billing Software, Telefonycards, ...

The book content is suitable for the german speaking countries: Germany, 
Austria and Switzerland.

Parts of the book are allready free avialbe on the internet.
The first 30 pages can be found at http://www.suvi.org/theory/asterisk.html

Everyweek, and if someone buys the book, one page more will be free aviable. 
With the buy of the book the freedom of the book is supported.

The book has 243 pages and can be bought at 
https://www.lulu.com/commerce/index.php?fBuyContent=359309 

Best Regards
Silvio

P.S. Sorry for the cross-posting.
-- 


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[asterisk-users] Reload clears queue stats

Has anyone noticed that doing a 'reload' on the Asterisk console clears all the 
stats shown by the 'show queues' command?
I'd like to report a bug, but would probably get my head chewed off for not 
testing it in the latest SVN code first.

Doug.
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Re: [asterisk-users] Polycom - simpler transfers?

Put the extensions or a wildcard that matches them into the polycom's
digitmap (dialplan?) I find info about this on page 79 of the 1.5.2 IP
Guide. My offices are like 110, 112, 113, 114, 115, 116 so I use
something like 110|11[2-9]| to match these instantly without softkey
afterwards. or if there's a T in there, 110T then that means timeout
after a few seconds and accept 110 if they dial it. This works in the
polycom's transfer feature as well.

Moj

Brian Vincent (C) wrote:
We’re using Polycom 601’s and I was wondering if there was a way to do
transfers by simply pressing the “Transfer” button followed by the
extension. Currently you need to hit “Transfer”, extension, and then a
transfer soft key. That extra soft key is really confusing the users.

---
Brian Vincent
Copper Mountain Telecom
[EMAIL PROTECTED]

Confidentiality Warning: This message and any attachments are intended only for the use of the intended recipient(s),
are confidential, and may be privileged. If you are not the intended recipient, you are hereby notified that any review,
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Re: [asterisk-users] Cisco 7960 SIP 8-3-0

2006-07-18 Thread Mailing List


Are you using the Non-CallManager version?


_
Mobilcom
http://www.mobilcom.net


- Original Message - 
From: Tong [EMAIL PROTECTED]

To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Monday, July 17, 2006 8:56 PM
Subject: Re: [asterisk-users] Cisco 7960 SIP 8-3-0



if you don't report it to cisco they won't know that bug exisit.


- Original Message - 
From: Daryl Johnson [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com

Sent: Monday, July 17, 2006 4:05 PM
Subject: Re: [asterisk-users] Cisco 7960 SIP 8-3-0



Tim,

I have seen the same 400 errors and the broken MWI...  I backed up to 
7.3...  We'll see if Cisco corrects these in the next release...


Daryl

- Original Message - 
From: Tim Connolly [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com

Sent: Monday, July 17, 2006 12:06 PM
Subject: [asterisk-users] Cisco 7960 SIP 8-3-0



Looks like the MWI broke on 8-3 also...


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Re: [asterisk-users] Polycom config file location


K, here's something a phone coughed up the other day:
?xml version=1.0 standalone=yes?
PHONE_CONFIG
OVERRIDES reg.1.ringType=17/

/PHONE_CONFIG

and here's another chunk from another phone.

?xml version=1.0 standalone=yes?
PHONE_CONFIG
OVERRIDES reg.1.fwdContact=312 reg.1.fwdStatus=0 
call.callsPerLineKey=1 reg.2.callsPerLineKey=8 
reg.2.thirdPartyName= reg.2.type=private reg.2.label= 
reg.2.displayName= reg.2.address= reg.1.thirdPartyName= 
reg.1.ringType=17/




/PHONE_CONFIG

madness, huh?  I think these are situations in which configurations made 
to the phone had to be written here to override the global provisioning 
files.  These exist in the ftp account's home dir as *-phone.cfg


Moj

 I'd love to see ngrep output of the communication between the phone
 and the FTP server for this.

why? I think the xmelly cfg files are proof enough.



Douglas Garstang wrote:
Been working with Polycom 301/501/601 for almost a year now and I've 
_never_ seen that behaviour!
I'd love to see ngrep output of the communication between the phone and 
the FTP server for this.


-Original Message-
*From:* Alex Robar [mailto:[EMAIL PROTECTED]
*Sent:* Monday, July 17, 2006 6:48 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Polycom config file location

Our 501's upload their configs to the server by themselves... Is
this uncommon? Seems to me that if you had no config on the server
at all but pointed the phones there anyways, they should upload
their current set of files there and then default to using that set
of configs until the server is updated.

Alex

On 7/17/06, *Jerry Jones* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:

If you at least setup your ftp server, and point the phones to it,
they will save a copy of their contact database so that will not be
lost.

Just edit and save an entry after server is ready and it will
create
the file.

No too hard to use the web browser and look at each phone to get its
current settings and manually create a config file.


On Jul 16, 2006, at 5:04 PM, Avi Miller wrote:

  Stephen Murphy wrote:
  My question is: How do I get the current config files the
phone is
  using off the phone?
 
  AFAIK, you can't. :( You can only provide new configuration files
  from your FTP/TFTP server. However, the Polycoms do strange
things
  when they've been configured in multiple locations. You might
find
  the phone overwriting the configuration files with its original
  configuration.
 
  That is not confirmed though. I've just seen my Polycoms do
weird
  stuff in the wild. :)
 
 
  --
  National Manager - Special Projects
 
   Melbourne / Sydney / Canberra / Hobart / London /
2/340 Gore StreetT: 1 300 SQUIZ (77859)
Fitzroy, VIC T: 03 9235 5400
3065 F: 03 9235 5444
 W: http://www.squiz.net/
 
  . Open Source  - Own it  -   Squiz.net
http://Squiz.net ./
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-- 
Alex Robar
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] 


!DSPAM:500,44bba8b032611804284693!




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!DSPAM:500,44bba8b032611804284693!


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RE: [asterisk-users] Polycom config file location

This is an override file. They are an addendum to the main sip.cfg and 
phone1.cfg files.

 -Original Message-
 From: Mojo with Horan  Company, LLC [mailto:[EMAIL PROTECTED]
 Sent: Tuesday, July 18, 2006 10:02 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Polycom config file location

 K, here's something a phone coughed up the other day:
 ?xml version=1.0 standalone=yes?
 PHONE_CONFIG
  OVERRIDES reg.1.ringType=17/

 /PHONE_CONFIG

 and here's another chunk from another phone.

 ?xml version=1.0 standalone=yes?
 PHONE_CONFIG
  OVERRIDES reg.1.fwdContact=312 reg.1.fwdStatus=0 
 call.callsPerLineKey=1 reg.2.callsPerLineKey=8 
 reg.2.thirdPartyName= reg.2.type=private reg.2.label= 
 reg.2.displayName= reg.2.address= reg.1.thirdPartyName= 
 reg.1.ringType=17/

 /PHONE_CONFIG

 madness, huh?  I think these are situations in which 
 configurations made 
 to the phone had to be written here to override the global 
 provisioning 
 files.  These exist in the ftp account's home dir as *-phone.cfg

 Moj

   I'd love to see ngrep output of the communication between the phone
   and the FTP server for this.

 why? I think the xmelly cfg files are proof enough.

 Douglas Garstang wrote:
  Been working with Polycom 301/501/601 for almost a year now 
 and I've 
  _never_ seen that behaviour!
  I'd love to see ngrep output of the communication between 
 the phone and 
  the FTP server for this.

  -Original Message-
  *From:* Alex Robar [mailto:[EMAIL PROTECTED]
  *Sent:* Monday, July 17, 2006 6:48 AM
  *To:* Asterisk Users Mailing List - Non-Commercial Discussion
  *Subject:* Re: [asterisk-users] Polycom config file location

  Our 501's upload their configs to the server by themselves... Is
  this uncommon? Seems to me that if you had no config on 
 the server
  at all but pointed the phones there anyways, they should upload
  their current set of files there and then default to 
 using that set
  of configs until the server is updated.

  Alex

  On 7/17/06, *Jerry Jones* [EMAIL PROTECTED]
  mailto:[EMAIL PROTECTED] wrote:

  If you at least setup your ftp server, and point 
 the phones to it,
  they will save a copy of their contact database so 
 that will not be
  lost.

  Just edit and save an entry after server is ready 
 and it will
  create
  the file.

  No too hard to use the web browser and look at each 
 phone to get its
  current settings and manually create a config file.

  On Jul 16, 2006, at 5:04 PM, Avi Miller wrote:

Stephen Murphy wrote:
My question is: How do I get the current config 
 files the
  phone is
using off the phone?

AFAIK, you can't. :( You can only provide new 
 configuration files
from your FTP/TFTP server. However, the Polycoms 
 do strange
  things
when they've been configured in multiple 
 locations. You might
  find
the phone overwriting the configuration files 
 with its original
configuration.

That is not confirmed though. I've just seen my 
 Polycoms do
  weird
stuff in the wild. :)

--
National Manager - Special Projects

 Melbourne / Sydney / Canberra / Hobart / London /
  2/340 Gore StreetT: 1 300 SQUIZ (77859)
  Fitzroy, VIC T: 03 9235 5400
  3065 F: 03 9235 5444
   W: http://www.squiz.net/

. Open Source  - Own it  -   Squiz.net
  http://Squiz.net ./
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  -- 
  Alex Robar
  [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] 

  !DSPAM:500,44bba8b032611804284693!

 --
 --

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Re: [asterisk-users] Two security holes fixed in latest versions of Asterisk

2006-07-18 Thread Tzafrir Cohen

On Tue, Jul 18, 2006 at 10:13:58AM +1200, Matt Riddell (NZ) wrote:
 -BEGIN PGP SIGNED MESSAGE-
 Hash: SHA1
 
 From: http://www.sineapps.com/news.php?rssid=1377
 
 ISS Xforce has published details of two security issues in Asterisk 1.x
 which were fixed in the recently release 1.2.10 version.
 
 Asterisk IAX2 Protocol Denial of Service Attack
 
 Summary:
 
 ISS X-Force has discovered a denial of service vulnerability in the
 Inter-Asterisk eXchange protocol version 2 (IAX2). IAX2 is used by
 Asterisk PBX software to exchange Voice over IP call setup and call
 content. If an attacker floods the PBX with call requests, the PBX will
 be unable to handle new telephone calls.
 
 IAX2 Protocol Denial of Service Amplification Attack
 
 Summary:
 
 ISS X-Force has discovered a traffic amplification vulnerability in the
 Inter-Asterisk eXchange protocol version 2 (IAX2). IAX2 is used by
 Asterisk PBX software to exchange Voice over IP call setup and call
 content. An attacker can leverage accounts without passwords on an
 Asterisk PBX to flood a third party with a large amount of UDP packets.
 If the attack is properly constructed the amount of traffic generated
 can saturate the victim's Internet connection. Networks do not have to
 use Asterisk PBX to be the victim of this kind of traffic flood.

If you wish to find more information and follow the links to ISS
Xforce's site, you'll actually get irrelevant and misleading
information.

I remember the issue of amplification raised in the dev list a number of
monthes ago regarding both SIP and IAX2. It is still not clear from
those texts what version 1.2.10 has actually fixed here. Where can I
find more details?

-- 
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755  iax:[EMAIL PROTECTED]
+972-50-7952406  jabber:[EMAIL PROTECTED]
[EMAIL PROTECTED] http://www.xorcom.com
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RE: [asterisk-users] Hitting # to Transfer out of a Queue

 -Original Message-
 From: Massimo Nuvoli [mailto:[EMAIL PROTECTED]
 Sent: Monday, July 17, 2006 8:30 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Hitting # to Transfer out of a Queue

 Douglas Garstang ha scritto:
  I have dialled into a Queue, and an agent has answered the 
 call with AgentcallbackLogin().
  The agent hits #1, to transfer the call. Asterisk responds 
 with 'Transfer', followed by dial tone.
  As soon as I enter a digit, Asterisk responds with 'I am 
 sorry. That is not a valid extension'

  This is working for regular user-user dialling, not going 
 through Queues. The queue app has Tt passed to it.

  Anyone got any ideas?

 In the queue configuration there is a context used when dialing
 (also in this case).

 Also, check the console, something like unable to find XY extension
 in KZ context must come out with the error.

Asterisk is logging:

Jul 18 10:02:38 VERBOSE[28172] logger.c: -- Unable to find extension '1' in 
context ''

I don't know why the context is empty, because I am setting it...

exten = oe_ccare,1,NoOp(*** Incoming call from ${CALLERID} to 
queue oe_ccare)
exten = oe_ccare,n,Set(TIMEOUT(response)=5)
exten = oe_ccare,n,
GotoIfTime(8:00-17:00|mon-fri|*|*?one_queue_acd,oe_ccare-open,1)
exten = oe_ccare,n,Goto(oe_ccare-shut,1)
exten = oe_ccare-open,1,   Answer
exten = oe_ccare-open,n,   Set(__TRANSFER_CONTEXT=one_start)
exten = oe_ccare-open,n,   NoOp(${__TRANSFER_CONTEXT})
exten = oe_ccare-open,n(queue1),   Queue(oe_custcare30) 

but... it's still empty!

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Re: [asterisk-users] Polycom config file location

I understand, I was helping Alex to make the connection too.  I thought 
this was what he was talking about.


Douglas Garstang wrote:

This is an override file. They are an addendum to the main sip.cfg and 
phone1.cfg files.


-Original Message-
From: Mojo with Horan  Company, LLC [mailto:[EMAIL PROTECTED]
Sent: Tuesday, July 18, 2006 10:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom config file location


K, here's something a phone coughed up the other day:
?xml version=1.0 standalone=yes?
PHONE_CONFIG
 OVERRIDES reg.1.ringType=17/

/PHONE_CONFIG

and here's another chunk from another phone.

?xml version=1.0 standalone=yes?
PHONE_CONFIG
 OVERRIDES reg.1.fwdContact=312 reg.1.fwdStatus=0 
call.callsPerLineKey=1 reg.2.callsPerLineKey=8 
reg.2.thirdPartyName= reg.2.type=private reg.2.label= 
reg.2.displayName= reg.2.address= reg.1.thirdPartyName= 
reg.1.ringType=17/




/PHONE_CONFIG

madness, huh?  I think these are situations in which 
configurations made 
to the phone had to be written here to override the global 
provisioning 
files.  These exist in the ftp account's home dir as *-phone.cfg


Moj

  I'd love to see ngrep output of the communication between the phone
  and the FTP server for this.

why? I think the xmelly cfg files are proof enough.



Douglas Garstang wrote:
Been working with Polycom 301/501/601 for almost a year now 
and I've 

_never_ seen that behaviour!
I'd love to see ngrep output of the communication between 
the phone and 

the FTP server for this.

-Original Message-
*From:* Alex Robar [mailto:[EMAIL PROTECTED]
*Sent:* Monday, July 17, 2006 6:48 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Polycom config file location

Our 501's upload their configs to the server by themselves... Is
this uncommon? Seems to me that if you had no config on 

the server

at all but pointed the phones there anyways, they should upload
their current set of files there and then default to 

using that set

of configs until the server is updated.

Alex

On 7/17/06, *Jerry Jones* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:

If you at least setup your ftp server, and point 

the phones to it,
they will save a copy of their contact database so 

that will not be

lost.

Just edit and save an entry after server is ready 

and it will

create
the file.

No too hard to use the web browser and look at each 

phone to get its

current settings and manually create a config file.


On Jul 16, 2006, at 5:04 PM, Avi Miller wrote:

  Stephen Murphy wrote:
  My question is: How do I get the current config 

files the

phone is
  using off the phone?
 
  AFAIK, you can't. :( You can only provide new 

configuration files
  from your FTP/TFTP server. However, the Polycoms 

do strange

things
  when they've been configured in multiple 

locations. You might

find
  the phone overwriting the configuration files 

with its original

  configuration.
 
  That is not confirmed though. I've just seen my 

Polycoms do

weird
  stuff in the wild. :)
 
 
  --
  National Manager - Special Projects
 
   Melbourne / Sydney / Canberra / Hobart / London /
2/340 Gore StreetT: 1 300 SQUIZ (77859)
Fitzroy, VIC T: 03 9235 5400
3065 F: 03 9235 5444
 W: http://www.squiz.net/
 
  . Open Source  - Own it  -   Squiz.net
http://Squiz.net ./
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-- 
Alex Robar
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] 







--
--

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!DSPAM:500,44bba8b032611804284693!

--
Mojo [EMAIL PROTECTED]
Office Manger, Horan  Company, LLC
(907) 747- x112

[asterisk-users] Examples of handeling input from phones with PHP

2006-07-18 Thread Chuck Bunn


Hi,

Can anyone direct me to where I might find examples of handling 
interactive input from a phone using PHP and AGI. I want to have someone 
dial an extension and then have the system request input from the user, 
take that input and put it into a database.


Thanks

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[asterisk-users] Asterisk 1.2.7.1 Crashing

2006-07-18 Thread Dan Brummer




Hello,
Well I had an issue 
this morning where the Asterisk process unexpectedly stopped. Below is an 
output from the full log:



Jul 18 09:18:51 
DEBUG[4892] chan_sip.c: (Provisional) Stopping retransmission (but retaining 
packet) on '[EMAIL PROTECTED]' 
Request 103: Not Found
Jul 18 09:18:51 
DEBUG[4892] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' 
of Request 251: Match Found
Jul 18 09:18:51 
DEBUG[4892] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' 
of Response 2: Match Found
Jul 18 09:18:51 
DEBUG[4892] chan_sip.c: Sending pending reinvite on '[EMAIL PROTECTED]'
Jul 18 09:18:51 
DEBUG[30430] channel.c: Scheduling timer at 0 sample 
intervals
Jul 18 09:18:51 
DEBUG[30430] channel.c: Scheduling timer at 0 sample 
intervals
Jul 18 09:18:51 
VERBOSE[30430] logger.c: == Parsing 
'/var/spool/asterisk/voicemail/default/2195/Old/msg.txt': Jul 18 09:18:51 
VERBOSE[30430] logger.c:
 == Parsing 
'/var/spool/asterisk/voicemail/default/2195/Old/msg.txt': 
Found
Jul 18 09:18:51 
DEBUG[30430] app_voicemail.c: VM-Duration: duration is: 1737 seconds converted 
to: 28 minutes
Jul 18 09:18:51 
DEBUG[30430] channel.c: Scheduling timer at 160 sample 
intervals
Jul 18 09:18:51 
VERBOSE[30430] logger.c: -- Playing 'digits/20' 
(language 'en')
Jul 18 09:18:51 
DEBUG[4892] chan_sip.c: Acked pending invite 102
Jul 18 09:18:51 
DEBUG[4892] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' 
of Request 102: Match Found
Jul 18 09:18:51 
DEBUG[4892] chan_sip.c: build_route: Contact hop: 
sip:[EMAIL PROTECTED]
Jul 18 09:18:51 
DEBUG[30430] channel.c: Scheduling timer at 0 sample 
intervals
Jul 18 09:18:51 
DEBUG[30430] channel.c: Scheduling timer at 0 sample 
intervals
Jul 18 09:18:51 
DEBUG[30430] channel.c: Scheduling timer at 160 sample 
intervals
Jul 18 09:18:51 
VERBOSE[30430] logger.c: -- Playing 'digits/8' (language 
'en')
Jul 18 09:18:52 
DEBUG[30430] channel.c: Scheduling timer at 0 sample 
intervals
Jul 18 09:18:52 
DEBUG[30430] channel.c: Scheduling timer at 0 sample 
intervals
Jul 18 09:18:52 
DEBUG[30430] channel.c: Scheduling timer at 160 sample 
intervals
Jul 18 09:18:52 
VERBOSE[30430] logger.c: -- Playing 'vm-minutes' 
(language 'en')
Jul 18 09:18:53 
DEBUG[30430] channel.c: Scheduling timer at 0 sample 
intervals
Jul 18 09:18:53 
DEBUG[30430] channel.c: Scheduling timer at 0 sample 
intervals
Jul 18 09:18:53 
WARNING[30430] file.c: File 
/var/spool/asterisk/voicemail/default/2195/Old/msg does not exist in any 
format
Jul 18 09:18:53 
WARNING[30430] file.c: Unable to open 
/var/spool/asterisk/voicemail/default/2195/Old/msg (format ulaw): No such 
file ordirectory
Jul 18 09:18:53 
DEBUG[30430] app.c: Locked path 
'/var/spool/asterisk/voicemail/default/2195/Old'
Jul 18 09:18:53 
DEBUG[30430] app.c: Unlocked path 
'/var/spool/asterisk/voicemail/default/2195/Old'
### SERVER 
STOPPED


### SERVER 
STARTED
Jul 18 09:32:30 
NOTICE[30730] cdr.c: CDR simple logging enabled.
Jul 18 09:32:30 
ERROR[30730] res_config_mysql.c: MySQL RealTime: Failed to connect database 
server on . Check debug for more info.
Jul 18 09:32:30 
WARNING[30730] res_config_mysql.c: MySQL RealTime: Couldn't establish 
connection. Check debug.
Jul 18 09:32:30 
NOTICE[30730] config.c: Registered Config Engine mysql
Jul 18 09:32:30 
WARNING[30730] cdr_addon_mysql.c: Unable to load config for mysql CDR's: 
cdr_mysql.conf
Jul 18 09:32:36 
VERBOSE[30745] logger.c: -- Saved useragent 
"PolycomSoundPointIP-SPIP_501-UA/1.6.3.0067" for peer 2179
Jul 18 09:32:36 
VERBOSE[30745] logger.c: -- Saved useragent 
"PolycomSoundPointIP-SPIP_601-UA/1.6.3.0067" for peer 
2109sh2198
Jul 18 09:32:36 
VERBOSE[30745] logger.c: -- Saved useragent 
"PolycomSoundPointIP-SPIP_501-UA/1.6.3.0067" for peer 
2148sh6353
Jul 18 09:32:36 
VERBOSE[30745] logger.c: -- Saved useragent 
"PolycomSoundPointIP-SPIP_501-UA/1.6.3.0067" for peer 5553
Jul 18 09:32:36 
DEBUG[30818] channel.c: Scheduling timer at 0 sample 
intervals
Jul 18 09:32:36 
DEBUG[30818] channel.c: Scheduling timer at 0 sample 
intervals


As you can see there 
is nothing in the log to help me troubleshoot this issue. Any 
ideas?

Thank 
you!
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[asterisk-users] extensions.conf 4 digit dialing question

2006-07-18 Thread Jerry Bonner









Hi all,



Does anyone have any tips on how I would
accomplish a plan where if a user dials 4 digits, then prefix 6 digits, then if
there is a local extension configured for that number dial it, otherwise send
it out another sip gateway ( my pstn gateway)?



Perhaps more specifically, are there any
construtcs that would dial extension if exists? Because I want to
make sure I dial a sip extension before routing it out to the pstn.



~jerry






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[asterisk-users] Net::CSTA on CPAN

2006-07-18 Thread Gabriel Millerd


A pretty cool package was released on CPAN today from Leif Johansson.

http://search.cpan.org/search?query=Net%3A%3ACSTA

NAME:
Net::CSTA - Perl extension for ECMA CSTA

SYNOPSIS:
 use Net::CSTA;

 # Connect to the CSTA server
 my $csta = Net::CSTA-new(Host='csta-server',Port='csta-server-port');
 # Create a monitor for '555'
 my $number = 555;
 $csta-request(serviceID=71,

serviceArgs={monitorObject={device={dialingNumber=$number}}})

 for (;;)
 {
my $pdu = $csta-receive();
print $pdu-toXML();
 }

DESCRIPTION ^

ECMA CSTA is an ASN.1 based protocol for Computer Integrated Telephony
(CTI) using CSTA it is possible to write code that communicates with a
PBX. Typical applications include receiving notifications for incoming
calls, placing calls, redirecting calls or placing conference calls.

BUGS

This module currently implements CSTA phase I - mostly because my PBX
(MD110 with Application Link 4.0) only supports phase I. Supporting
multiple versions will require some thought since the versions are
largly incompatible.

The CSTA client opens a UDP port on  to receive incoming
usolicited notifications. This is not implemented yet.

SECURITY CONSIDERATIONS ^

CSTA is a protocol devoid of any form of security. Take care to
firewall your CSTA server and throw away the key.
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[asterisk-users] TxFax

2006-07-18 Thread Giordano Grandis




Hi 
all,
I'm using spandsp 
0.0.2pre21 with tiff 3.7.1 and asterisk 1.0.9
With rxfax 
application, everythinghs is ok, but when i try to send a fax whit txfax 
applicationthe channel got hangup. I'm testing send fax towards an ATA 
grandstrem 488, but if i sent it to a normal fax i got the same 
error.
This is what 
happen:

 
-- Executing TxFAX("Zap/1-1", "/fax07182006190657.tif|caller|debug") in new 
stackFLOW Slow carrier upFLOW Slow carrier downFLOW Slow carrier 
upFLOW  NSF: 20 ad 00 36 20 00 00 00 00FLOW NSF without 
final frame tagFLOW The remote was made by 'HP'FLOW  CSI: 40 
37 37 34 39 35 34 34 20 35 38 30 20 39 33 2b 20 20 20 20 20FLOW CSI without 
final frame tagFLOW Remote fax gave CSI as: "+39 085 4459477"FLOW 
 DIS: 80 00 ee f8 c4 80 92 80 80 98 00FLOW DIS with final frame 
tagFLOW In state 10FLOW ???:FLOW 3rd generation mobile 
networkFLOW V.8 capableFLOW Prefer 64 octet 
blocksFLOW Reserved: 0x98FLOW Supported data 
signalling rates: V.29FLOW R8x7.7lines/mm and/or 
200x200pels/25.4mmFLOW 2D codingFLOW Scan line 
length: 215mmFLOW Recording length: A4 
(297mm)FLOW Receiver's minimum scan line time: 20ms at 3.85 
l/mm: T7.7 = T3.85FLOW Uncompressed modeFLOW 
Reserved: 0x10FLOW Minimum scan line time for higher 
resolutions: T15.4 = T7.7FLOW Binary file transfer 
(BFT)FLOW Reserved: 1FLOW ???:FLOW Prefer 
256 octet blocksFLOW Reserved: 0x80FLOW 
Supported data signalling rates: V.27ter fallback modeFLOW 2D 
codingFLOW Scan line length: 215mmFLOW Recording 
length: A4 (297mm)FLOW Receiver's minimum scan line time: 20ms 
at 3.85 l/mm: T7.7 = T3.85FLOW Start sending documentTIFFOpen: 
/fax07182006190657.tif: Cannot open.FLOW Cannot open source TIFF file 
'/fax07182006190657.tif'FLOW DIS nothing to send [0]FLOW 
???:FLOW Real-time Internet fax (T.38)FLOW V.8 
capableFLOW Prefer 64 octet blocksFLOW Reserved: 
0x90FLOW Supported data signalling rates: V.27ter fallback 
modeFLOW 2D codingFLOW Scan line length: 
215mmFLOW Recording length: A4 (297mm)FLOW 
Receiver's minimum scan line time: 20ms at 3.85 l/mm: T7.7 = 
T3.85FLOW Reserved: 0x1FLOW Minimum scan line 
time for higher resolutions: T15.4 = T7.7FLOW Character 
modeFLOW Reserved: 0x10FLOW DIS nothing to receive 
[0]FLOW Changed from phase 2 to 4FLOW  DCN: fbFLOW HDLC 
underflow in state 2FLOW DisconnectingFLOW Changed from phase 4 to 
7FLOW Changed from phase 7 to 8 == Spawn extension (incoming, 
X, 2) exited non-zero on 'Zap/2-1' -- Hungup 
'Zap/2-1' -- Channel 0/1, span 1 got 
hangup -- Hungup 'Zap/1-1'
I'm going to check 
it towards several fax, but i think i will get the same 
problem.
Anyone can help 
me?

Thanks very 
much

Giordano
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Re: [asterisk-users] Asterisk 1.2.7.1 Crashing

2006-07-18 Thread Doug Lytle


Dan Brummer wrote:

Hello,
Well I had an issue this morning where the Asterisk process 
unexpectedly stopped.  Below is an output from the full log:
 
Jul 18 09:18:51 VERBOSE[30430] logger.c:   == Parsing 
'/var/spool/asterisk/voicemail/default/2195/Old/msg.txt': Jul 18 
09:18:51 VERBOSE[30430] logger.c:
  == Parsing 
'/var/spool/asterisk/voicemail/default/2195/Old/msg.txt': Found
Jul 18 09:18:53 WARNING[30430] file.c: File 
/var/spool/asterisk/voicemail/default/2195/Old/msg does not exist 
in any format
Jul 18 09:18:53 WARNING[30430] file.c: Unable to open 
/var/spool/asterisk/voicemail/default/2195/Old/msg (format ulaw): 
No such file ordirectory
Jul 18 09:18:53 DEBUG[30430] app.c: Locked path 
'/var/spool/asterisk/voicemail/default/2195/Old'
Jul 18 09:18:53 DEBUG[30430] app.c: Unlocked path 
'/var/spool/asterisk/voicemail/default/2195/Old'


This particular issue was fixed in 1.2.10,  You'll want to upgrade.

Doug

--

Ben Franklin quote:

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


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Re: [asterisk-users] extensions.conf 4 digit dialing question

2006-07-18 Thread Bruce Reeves

I think asterisk will take care of this for you. Asterisk will take the most complete match in a pattern match. For example if you have a local extension 1234567890 and a pattern match _XX then Asterisk will match the 1234567890 to the exact match if it exist, if not then it will got the closest pattern it can find. Does that make sense or help?
On 7/18/06, Jerry Bonner [EMAIL PROTECTED] wrote:



















Hi all,



Does anyone have any tips on how I would
accomplish a plan where if a user dials 4 digits, then prefix 6 digits, then if
there is a local extension configured for that number dial it, otherwise send
it out another sip gateway ( my pstn gateway)?



Perhaps more specifically, are there any
construtcs that would "dial extension if exists"? Because I want to
make sure I dial a sip extension before routing it out to the pstn.



~jerry







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http://lists.digium.com/mailman/listinfo/asterisk-users-- BruceNortex Networks
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RE: SV: [Asterisk-Users] Nokia E61

2006-07-18 Thread Fredrik Emil Jensen

I'm not completely sure if it will. 

As I under stand the qualify options will request a SIP OPTIONS call
every minute from the phone. This solved my NAT problem with one phone
through a linux firewall running ipfilter, I am going to test more phone
through the firewall as I can see it uses the default port 5060, not
quite sure if this will work when you have many phone behind the same
NAT router/firewall. 

/Fredrik Jensen

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dinesh
Nair
Sent: 18. juli 2006 16:36
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: SV: [Asterisk-Users] Nokia E61



On 07/18/06 04:03 Fredrik Emil Jensen said the following:
 the packet too, but when the firewall/router loses its table (usually
it
 will timeout after xx sec/min) you will only be able to dial outgoing

can't you use qualify to get the nat device to keep the mapping ?

-- 
Regards,   /\_/\   All dogs go to heaven.
[EMAIL PROTECTED](0 0)
http://www.openmalaysiablog.com/
+==oOO--(_)--OOo
==+
| for a in past present future; do
|
|   for b in clients employers associates relatives neighbours pets; do
|
|   echo The opinions here in no way reflect the opinions of my $a $b.
|
| done; done
|
+===
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RE: [asterisk-users] Asterisk 1.2.7.1 Crashing

2006-07-18 Thread Dan Brummer

Thank you Doug for the response.  Do you know if the 1.2.10 release
fixes the warm transfer issue I experienced in 1.2.9.1?

Thank you,
Dan 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle
Sent: Tuesday, July 18, 2006 10:23 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 1.2.7.1 Crashing

Dan Brummer wrote:
 Hello,
 Well I had an issue this morning where the Asterisk process 
 unexpectedly stopped.  Below is an output from the full log:
  
 Jul 18 09:18:51 VERBOSE[30430] logger.c:   == Parsing 
 '/var/spool/asterisk/voicemail/default/2195/Old/msg.txt': Jul 18
 09:18:51 VERBOSE[30430] logger.c:
   == Parsing
 '/var/spool/asterisk/voicemail/default/2195/Old/msg.txt': Found 
 Jul 18 09:18:53 WARNING[30430] file.c: File 
 /var/spool/asterisk/voicemail/default/2195/Old/msg does not exist 
 in any format Jul 18 09:18:53 WARNING[30430] file.c: Unable to open 
 /var/spool/asterisk/voicemail/default/2195/Old/msg (format ulaw):
 No such file ordirectory
 Jul 18 09:18:53 DEBUG[30430] app.c: Locked path 
 '/var/spool/asterisk/voicemail/default/2195/Old'
 Jul 18 09:18:53 DEBUG[30430] app.c: Unlocked path 
 '/var/spool/asterisk/voicemail/default/2195/Old'

This particular issue was fixed in 1.2.10,  You'll want to upgrade.

Doug

-- 
 
Ben Franklin quote:

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


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[asterisk-users] rxfax Got hangup

2006-07-18 Thread Jan Fousek

Hi all,
 I'm trying to setup asterisk and spandsp to recieve fax transmissions. I
got Asterisk to detect fax calls, it even tries to communicate, but the
other side doesn't seem to send the main data. Instead it ends the
communication with hangup. Have anybody got an idea? 

Thanks a lot.
Jan Fousek

This is a relevant part of the log:


Jul 18 19:58:30 VERBOSE[20671] logger.c: -- Executing 
Answer(SIP/420543254384-b5dd, ) in new stack
Jul 18 19:58:30 VERBOSE[20671] logger.c: -- Executing 
AbsoluteTimeout(SIP/420543254384-b5dd, 35) in new stack
Jul 18 19:58:30 VERBOSE[20671] logger.c: -- Set Absolute Timeout to 35
Jul 18 19:58:30 VERBOSE[20671] logger.c: -- Executing 
Set(SIP/420543254384-b5dd, 
FAXFILE=/var/spool/asterisk-fax/1153245510.15.tif) in new stack
Jul 18 19:58:30 VERBOSE[20671] logger.c: -- Executing 
RxFAX(SIP/420543254384-b5dd, 
/var/spool/asterisk-fax/1153245510.15.tif|debug) in new stack
Jul 18 19:58:30 DEBUG[31032] chan_sip.c: Stopping retransmission on '[EMAIL 
PROTECTED]' of Response 23395782: Match Found
Jul 18 19:58:32 DEBUG[31032] chan_sip.c: Auto destroying call '[EMAIL 
PROTECTED]'
Jul 18 19:58:33 DEBUG[31032] chan_sip.c: Auto destroying call '[EMAIL 
PROTECTED]'
Jul 18 19:58:33 DEBUG[20671] app_rxfax.c: FLOW Changed from phase 1 to 4
Jul 18 19:58:33 DEBUG[20671] app_rxfax.c: FLOW  DIS:Jul 18 19:58:33 
DEBUG[20671] app_rxfax.c:  80Jul 18 19:58:33 DEBUG[20671] app_rxfax.c:  00Jul 
18 19:58:33 DEBUG[20671] app_rxfax.c:  ceJul 18 19:58:33 DEBUG[20671] 
app_rxfax.c:  f4Jul 18 19:58:33 DEBUG[20671] app_rxfax.c:  80Jul 18 19:58:33 
DEBUG[20671] app_rxfax.c:  80Jul 18 19:58:33 DEBUG[20671] app_rxfax.c:  81Jul 
18 19:58:33 DEBUG[20671] app_rxfax.c:  80Jul 18 19:58:33 DEBUG[20671] 
app_rxfax.c:  80Jul 18 19:58:33 DEBUG[20671] app_rxfax.c:  80Jul 18 19:58:33 
DEBUG[20671] app_rxfax.c:  18Jul 18 19:58:33 DEBUG[20671] app_rxfax.c: 
Jul 18 19:58:35 DEBUG[20671] app_rxfax.c: FLOW HDLC underflow in state 9
Jul 18 19:58:35 DEBUG[20671] app_rxfax.c: FLOW Changed from phase 4 to 3
Jul 18 19:58:36 DEBUG[20671] app_rxfax.c: FLOW HDLC carrier up
Jul 18 19:58:36 DEBUG[20671] app_rxfax.c: FLOW HDLC framing OK
Jul 18 19:58:37 DEBUG[20671] app_rxfax.c: FLOW  ???:Jul 18 19:58:37 
DEBUG[20671] app_rxfax.c:  1aJul 18 19:58:37 DEBUG[20671] app_rxfax.c: 
Jul 18 19:58:37 DEBUG[20671] app_rxfax.c: FLOW ??? with final frame tag
Jul 18 19:58:37 DEBUG[20671] app_rxfax.c: FLOW In state 9
Jul 18 19:58:37 DEBUG[20671] app_rxfax.c: FLOW HDLC carrier down
Jul 18 19:58:38 DEBUG[20671] app_rxfax.c: FLOW T4 timeout in state 9
Jul 18 19:58:38 DEBUG[20671] app_rxfax.c: FLOW Changed from phase 3 to 4
Jul 18 19:58:38 DEBUG[20671] app_rxfax.c: FLOW  DIS:Jul 18 19:58:38 
DEBUG[20671] app_rxfax.c:  80Jul 18 19:58:38 DEBUG[20671] app_rxfax.c:  00Jul 
18 19:58:38 DEBUG[20671] app_rxfax.c:  ceJul 18 19:58:38 DEBUG[20671] 
app_rxfax.c:  f4Jul 18 19:58:38 DEBUG[20671] app_rxfax.c:  80Jul 18 19:58:38 
DEBUG[20671] app_rxfax.c:  80Jul 18 19:58:38 DEBUG[20671] app_rxfax.c:  81Jul 
18 19:58:38 DEBUG[20671] app_rxfax.c:  80Jul 18 19:58:38 DEBUG[20671] 
app_rxfax.c:  80Jul 18 19:58:38 DEBUG[20671] app_rxfax.c:  80Jul 18 19:58:38 
DEBUG[20671] app_rxfax.c:  18Jul 18 19:58:38 DEBUG[20671] app_rxfax.c: 
Jul 18 19:58:40 DEBUG[20671] app_rxfax.c: FLOW HDLC underflow in state 9
Jul 18 19:58:40 DEBUG[20671] app_rxfax.c: FLOW Changed from phase 4 to 3
Jul 18 19:58:43 DEBUG[20671] app_rxfax.c: FLOW T4 timeout in state 9
Jul 18 19:58:43 DEBUG[20671] app_rxfax.c: FLOW Changed from phase 3 to 4
Jul 18 19:58:43 DEBUG[20671] app_rxfax.c: FLOW  DIS:Jul 18 19:58:43 
DEBUG[20671] app_rxfax.c:  80Jul 18 19:58:43 DEBUG[20671] app_rxfax.c:  00Jul 
18 19:58:43 DEBUG[20671] app_rxfax.c:  ceJul 18 19:58:43 DEBUG[20671] 
app_rxfax.c:  f4Jul 18 19:58:43 DEBUG[20671] app_rxfax.c:  80Jul 18 19:58:43 
DEBUG[20671] app_rxfax.c:  80Jul 18 19:58:43 DEBUG[20671] app_rxfax.c:  81Jul 
18 19:58:43 DEBUG[20671] app_rxfax.c:  80Jul 18 19:58:43 DEBUG[20671] 
app_rxfax.c:  80Jul 18 19:58:43 DEBUG[20671] app_rxfax.c:  80Jul 18 19:58:43 
DEBUG[20671] app_rxfax.c:  18Jul 18 19:58:43 DEBUG[20671] app_rxfax.c: 
Jul 18 19:58:45 DEBUG[20671] app_rxfax.c: FLOW HDLC underflow in state 9
Jul 18 19:58:45 DEBUG[20671] app_rxfax.c: FLOW Changed from phase 4 to 3
Jul 18 19:58:48 DEBUG[20671] app_rxfax.c: FLOW T4 timeout in state 9
Jul 18 19:58:48 DEBUG[20671] app_rxfax.c: FLOW Changed from phase 3 to 4
Jul 18 19:58:48 DEBUG[20671] app_rxfax.c: FLOW  DIS:Jul 18 19:58:48 
DEBUG[20671] app_rxfax.c:  80Jul 18 19:58:48 DEBUG[20671] app_rxfax.c:  00Jul 
18 19:58:48 DEBUG[20671] app_rxfax.c:  ceJul 18 19:58:48 DEBUG[20671] 
app_rxfax.c:  f4Jul 18 19:58:48 DEBUG[20671] app_rxfax.c:  80Jul 18 19:58:48 
DEBUG[20671] app_rxfax.c:  80Jul 18 19:58:48 DEBUG[20671] app_rxfax.c:  81Jul 
18 19:58:48 DEBUG[20671] app_rxfax.c:  80Jul 18 19:58:48 DEBUG[20671] 
app_rxfax.c:  80Jul 18 19:58:48 DEBUG[20671] app_rxfax.c:  80Jul 18 19:58:48 
DEBUG[20671] app_rxfax.c:  18Jul 18 19:58:48 DEBUG[20671]

Re: [asterisk-users] Asterisk 1.2.7.1 Crashing

2006-07-18 Thread Doug Lytle


Dan Brummer wrote:

Thank you Doug for the response.  Do you know if the 1.2.10 release
fixes the warm transfer issue I experienced in 1.2.9.1?

  


I have no idea.  I would suggest reading the change log at:

http://ftp.digium.com/pub/telephony/asterisk/releases/ChangeLog-1.2.10

Doug

--

Ben Franklin quote:

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


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[asterisk-users] Tf.voipmich.com - Broken?

2006-07-18 Thread Brent Torrenga

Anyone notice that tf.voipmich.com (ENUM for US toll free service) will
connect you successfully, but then disconnect after what seems like 30
seconds or so? Anyone know what might be going on here? I googled the hell
out of voipmich and did not get very far.

Sincerely,

Brent A. Torrenga

Torrenga Engineering, Inc.
907 Ridge Road
Munster, Indiana 46321-1771

tel:+1 219 836 8918 x325
fax:+1 219 836 1138
email:[EMAIL PROTECTED]
web:www.torrenga.com

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RE: [asterisk-users] Hitting # to Transfer out of a Queue

2006-07-18 Thread Patrick

On Tue, 2006-07-18 at 10:29 -0600, Douglas Garstang wrote:
[snip]
 exten = oe_ccare,1,NoOp(*** Incoming call from ${CALLERID} 
 to queue oe_ccare)
 exten = oe_ccare,n,Set(TIMEOUT(response)=5)
 exten = oe_ccare,n,
 GotoIfTime(8:00-17:00|mon-fri|*|*?one_queue_acd,oe_ccare-open,1)
 exten = oe_ccare,n,Goto(oe_ccare-shut,1)
 exten = oe_ccare-open,1,   Answer
 exten = oe_ccare-open,n,   Set(__TRANSFER_CONTEXT=one_start)
 exten = oe_ccare-open,n,   NoOp(${__TRANSFER_CONTEXT})
 exten = oe_ccare-open,n(queue1),   Queue(oe_custcare30) 

Is this a literal copy of your dialplan? If so I was not aware you could
put spaces between priorities and actions. Have you tried removing them:
exten = foo,1,NoOP(spaces are evil, mostly)

Regards,
Patrick

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[asterisk-users] call-limit and problem with freezy phones. also freezy zap channels with x101p card.

2006-07-18 Thread Vitaly Oborsky


call-limit and problem with freezy phones. also freezy zap channels
with x101p card.

Hello all.
I have installed asterisk 1.2.9.1 and zaptel 1.2.6.
I have such configuration:
I have some phones  with planet vip-156 with configuration in sip.conf:
[036] ; planet 222
type=friend
host=dynamic
canreinvite=yes
username=036
secret=036
nat=no
qualify=10
dtfmode=rfc2833
musiconhold=default
context=office
callerid=036
disallow=all
allow=ulaw
callgroup=1
pickupgroup=1
call-limit=1
.

everything work good, but sometimes i have situation in which the
asterisk thinks that phone is borrowed at present, and appear message
like that:
cannot create a sip channel due to usage limit...
but when in this situation i check channels with comand show
channels, i see that phone, on which can not call, in this moment
absolutly free.
That situation appear when i started using call-limit=1.
When i do asterisk -rx reload, then that fixes.
How that can be fixed without reloads?
Also I have problem with zap channels only with x101p cards. Sometimes
channel stay up even when line hangup. Also I have tdm400 cards, they
work perfect.
section in zappata.conf for x101p channel:

context=generic-inc
signalling=fxs_ks
rxwink=300  ; Atlas seems to use long (250ms) winks
usedistinctiveringdetection=yes
hidecallerid=no
callwaiting=no
usecallingpres=no
callwaitingcallerid=no
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=no
echocancelwhenbridged=no
echotraining=no
rxgain=-4
txgain=-4
;group=2
callgroup=3
pickupgroup=3
immediate=yes
busydetect=yes
busycount=8
callprogress=no
pulsedial=no
musiconhold=default
switchtype = national
group = 3
channel = 6
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RE: [asterisk-users] Hitting # to Transfer out of a Queue