Re: [asterisk-users] Server-to-server BLF

[Ronald Cepres]

Hi to all,

I've managed to get the XMPP PubSub method to work on my set-up! Just
carefully follow these instructions on the wiki:
https://wiki.asterisk.org/wiki/display/AST/Distributed+Device+State+with+XMPP+PubSub


Maybe this IRC log would also help you troubleshoot:
http://apt.rikers.org/%23asterisk-bugs/20091008.html.gz

One thing I noticed though is that if you do a devstate list, the state
is sometimes not the same as listed in core show hints (core show hints
has the correct state). Nevertheless, BLF works good for me.

BTW, has anyone on the list tried out the AIS method yet? I'm a bit curious
which method is better.

Regards,
Ronald

On Fri, Jan 13, 2012 at 3:44 PM, Leandro Dardini ldard...@gmail.com wrote:

 Me too, an maybe other people on the list are interested in knowing
 your effort result and maybe appreciate a guide on the topic.

 Thank you

 Leandro

 2012/1/13 Ronald Cepres rbcep...@gmail.com:
  Hi Ishfaq,
 
  Thanks for your reply. I've already started trying the XMPP method so I
  can't help you with the AIS method as of the moment. I'll let you know
 the
  result of my test.
 
  Regards,
  Ronald
 
 
  On Fri, Jan 6, 2012 at 5:14 PM, Ishfaq Malik i...@pack-net.co.uk wrote:
 
  Hi Ronald
 
  I took a bit of interest in your problem as I'm going to have to be
  doing the same thing in a few weeks.
 
  oenais is in the yum repositories so you can install from there if using
  redhat/centos based OS
 
  It is also in apt repositories if you're using a debian based OS
 
  Let me know how you get on
 
  Ish
 
  On Thu, 2012-01-05 at 12:07 +0800, Ronald Cepres wrote:
   Hi Kevin,
  
  
   Thanks for your suggestion.
  
  
   On the website of OpenAIS, it seems that it is not supported anymore
   and their download links (FTP and SVN) are broken (been trying it for
   about a month now). Is it still possible to use OpenAIS method? The
   other solution on the wiki is using XMPP which is for jabber. IMHO, it
   means that the XMPP solution can't be used on SIP peers, right?
  
  
   Regards,
   Ronald
  
   On Thu, Nov 17, 2011 at 1:01 AM, Kevin P. Fleming
   kpflem...@digium.com wrote:
   On 11/16/2011 04:18 AM, Ronald Cepres wrote:
   Hi all,
  
   Do you have an idea on the best way on how to
   implement a system with
   multiple Asterisk servers with BLF working in such a
   way that a peer on
   one server can subscribe to another peer on the other
   server in a
   seamless manner? Has anyone set-up a system like this
   before?
  
  
   Here is one way:
  
   https://wiki.asterisk.org/wiki/display/AST/Distributed+Device
   +State+with+AIS
  
   There are other methods documented on the wiki as well.
  
   --
   Kevin P. Fleming
   Digium, Inc. | Director of Software Technologies
   Jabber: kflem...@digium.com | SIP: kpflem...@digium.com |
   Skype: kpfleming
   445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
   Check us out at www.digium.com  www.asterisk.org
  
   --
  
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  --
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  Software Developer
  PackNet Ltd
 
  Office:   0161 660 3062
 
 
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Re: [asterisk-users] Server-to-server BLF

[Ronald Cepres]

Hi Ishfaq,

Thanks for your reply. I've already started trying the XMPP method so I
can't help you with the AIS method as of the moment. I'll let you know the
result of my test.

Regards,
Ronald

On Fri, Jan 6, 2012 at 5:14 PM, Ishfaq Malik i...@pack-net.co.uk wrote:

 Hi Ronald

 I took a bit of interest in your problem as I'm going to have to be
 doing the same thing in a few weeks.

 oenais is in the yum repositories so you can install from there if using
 redhat/centos based OS

 It is also in apt repositories if you're using a debian based OS

 Let me know how you get on

 Ish

 On Thu, 2012-01-05 at 12:07 +0800, Ronald Cepres wrote:
  Hi Kevin,
 
 
  Thanks for your suggestion.
 
 
  On the website of OpenAIS, it seems that it is not supported anymore
  and their download links (FTP and SVN) are broken (been trying it for
  about a month now). Is it still possible to use OpenAIS method? The
  other solution on the wiki is using XMPP which is for jabber. IMHO, it
  means that the XMPP solution can't be used on SIP peers, right?
 
 
  Regards,
  Ronald
 
  On Thu, Nov 17, 2011 at 1:01 AM, Kevin P. Fleming
  kpflem...@digium.com wrote:
  On 11/16/2011 04:18 AM, Ronald Cepres wrote:
  Hi all,
 
  Do you have an idea on the best way on how to
  implement a system with
  multiple Asterisk servers with BLF working in such a
  way that a peer on
  one server can subscribe to another peer on the other
  server in a
  seamless manner? Has anyone set-up a system like this
  before?
 
 
  Here is one way:
 
  https://wiki.asterisk.org/wiki/display/AST/Distributed+Device
  +State+with+AIS
 
  There are other methods documented on the wiki as well.
 
  --
  Kevin P. Fleming
  Digium, Inc. | Director of Software Technologies
  Jabber: kflem...@digium.com | SIP: kpflem...@digium.com |
  Skype: kpfleming
  445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
  Check us out at www.digium.com  www.asterisk.org
 
  --
 
 _
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  Thurs:
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  asterisk-users mailing list
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   http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
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 --
 Ishfaq Malik
 Software Developer
 PackNet Ltd

 Office:   0161 660 3062


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[asterisk-users] Asterisk as register server through OpenSIPS

[Ronald Cepres]

Hi all,

I've been trying to register a SIP user agent to an Asterisk server using
OpenSIPS as SIP router. The functionality is working fine. However,
Asterisk uses the IP address of the OpenSIPS server as the peer IP address.
How can I use the original IP address of the peer without changing the
peer's nat=yes?


I appreciate any kind of help. Thanks!

Regards,
Ronald
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Re: [asterisk-users] Server-to-server BLF

[Ronald Cepres]

Hi Kevin,

Thanks for your suggestion.

On the website of OpenAIS, it seems that it is not supported anymore and
their download links (FTP and SVN) are broken (been trying it for about a
month now). Is it still possible to use OpenAIS method? The other solution
on the wiki is using XMPP which is for jabber. IMHO, it means that the XMPP
solution can't be used on SIP peers, right?

Regards,
Ronald

On Thu, Nov 17, 2011 at 1:01 AM, Kevin P. Fleming kpflem...@digium.comwrote:

 On 11/16/2011 04:18 AM, Ronald Cepres wrote:

 Hi all,

 Do you have an idea on the best way on how to implement a system with
 multiple Asterisk servers with BLF working in such a way that a peer on
 one server can subscribe to another peer on the other server in a
 seamless manner? Has anyone set-up a system like this before?


 Here is one way:

 https://wiki.asterisk.org/**wiki/display/AST/Distributed+**
 Device+State+with+AIShttps://wiki.asterisk.org/wiki/display/AST/Distributed+Device+State+with+AIS

 There are other methods documented on the wiki as well.

 --
 Kevin P. Fleming
 Digium, Inc. | Director of Software Technologies
 Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 Check us out at www.digium.com  www.asterisk.org

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[asterisk-users] Server-to-server BLF

[Ronald Cepres]

Hi all,

Do you have an idea on the best way on how to implement a system with
multiple Asterisk servers with BLF working in such a way that a peer on one
server can subscribe to another peer on the other server in a seamless
manner? Has anyone set-up a system like this before?

Thanks!

Regards,
Ronald
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Re: [asterisk-users] Asterisk Realtime Time Dial App

[Ronald Cepres]

Hi Nick,

You mean if it is possible for Asterisk to use realtime dialplan? If it is,
AFAIK it is possible using a table format for realtime extensions.

Regards,
Ronald

On Mon, Sep 26, 2011 at 1:33 PM, Sam Govind govoi...@gmail.com wrote:

 Hmmm..interesting..I haven't came across anything like this so far..How
 about making a new table for the insertion of a new call data..and trigger
 some script to activate AMI/Call file according to new call data.


 http://dev.mysql.com/doc/refman/5.0/en/faqs-triggers.html#qandaitem-B-5-1-10


 On Mon, Sep 26, 2011 at 3:53 AM, Nick Khamis sym...@gmail.com wrote:

 Hello Everyone,

 I have MOH, Sip Friends/Peers, Voice Mail all working realtime. I was
 wondering if it Is possible to have Asterisk make a calls based on a
 record inserted in a table realtime? If I have to develop something using
 AGI
 or AMI, I can do this  with a little direction?

 Thanks in Advance,

 Nick

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[asterisk-users] Asterisk-Radius integration

[Ronald Cepres]

Hi all,

I'm trying to setup a system such that when a call comes in to Asterisk, it
first checks the account balance of the caller via Radius and then determine
if the call should go through or not.

I have an average experience in Asterisk but I'm quite new to Radius so I'm
not sure if this setup is possible. Has anyone achieved this kind of setup?

Thanks!

Regards,
Ronald
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Re: [asterisk-users] Asterisk-Radius integration

[Ronald Cepres]

Hi amit,

Thanks for the quick reply.

I'll look into this and hopefully get this to work. Thanks again!

Regards,
Ronald

On Wed, Sep 21, 2011 at 2:45 PM, amit anand onewaytoconn...@gmail.comwrote:

 Hi

 for this you need to write some agi script that will handle the other
 feature.

 Also you can try for A2billing, its a complete solution for Billing with
 asterisk

 On Wed, Sep 21, 2011 at 12:08, Ronald Cepres rbcep...@gmail.com wrote:

 Hi all,

 I'm trying to setup a system such that when a call comes in to Asterisk,
 it first checks the account balance of the caller via Radius and then
 determine if the call should go through or not.

 I have an average experience in Asterisk but I'm quite new to Radius
 so I'm not sure if this setup is possible. Has anyone achieved this kind of
 setup?

 Thanks!

 Regards,
 Ronald

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 +91 9818559898



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[asterisk-users] Updated: 10 Minutes: Asterisk PBX on Amazon EC2

[Ronald Lewis]

Dear Asterisk Community:

With more than 10,000 readers worldwide, I've refreshed my free Asterisk PBX
on Amazon EC2 ebook for 2011. It has been used by Avaya, Polycom,
universities, and consultants everywhere. Did I mention it's free? If you
have suggestions for its improvement or things you'd like to see, please let
me know!

It's online here:

http://ronaldlewis.com/10-minutes-asterisk-pbx-on-amazon-ec2-quickstart-guide
http://www.scribd.com/doc/3905321/10-Minutes-Asterisk-PBX-on-Amazon-EC2-A-Quickstart-Guide

Thanks for your support!

Best,

Ronald Lewis
Author, 10 Minutes: Asterisk PBX on Amazon EC2
Denver, Colorado
http://ronaldlewis.com
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[asterisk-users] how to improve sound file quality?

[Ronald Wiplinger (Lists)]

We have recorded wav files with 44k, 22k, 16k, 11k and 8k

Asterisk does not accept these wav files. I used sox input.wav
output.gsm to get them to work.
However, the only the 8k file did convert and the quality is poor. How
can I improve the quality?

bye

Ronald

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[asterisk-users] Can asterisk work with a dynamic IP?

[Ronald Wiplinger (Lists)]

I know I can setup asterisk without Internet at all and it works as
local pbx.

Would an asterisk box work with a dynamic IP, with a dyndns name?
What must I take care if I try that?

bye

Ronald

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[asterisk-users] Wellgate Asterisk

[Ronald Wiplinger (Lists)]

I got a Wellgate 3804A and need some hints:

Both have public IP *.131=asterisk (1.6.0.1) *.133= Wellgate

Wellgate 3804A settings (Line1~Line4):

1. Sip Config
 Mode:   Proxy
 Primary Proxy IP Address:  *.131
 Primary Proxy port:  5060
 Line1 Number:  1002

2. Security Config
 Line1 Account:  1002
 Line1 Password:  **

3. Line Configuration
 Line1:  Type=FXO, Hunting Group=2, Hot Line = 88621002


Asterisk settings:

users.conf:
[1002]
context = DID_1002
host = *.133
username = 1002
secret = **
trunkname = WellGate-1002  ; GUI metadata
hasiax = no
registeriax = no
hassip = yes
registersip = yes
trunkstyle = voip
hasexten = no
host = dynamic
disallow = all
allow = ulaw,alaw,gsm,g726,g729


extensions.conf
1002 = SIP/1002
...
[DID_1002]
exten = _88621002,1,NoOp(${CALLERID(num)})
exten = _88621002,n,Wait(1)
exten = _88621002,n,SayUnixTime
include = DID_1001_timeinterval_working day|${timeinterval_working day}
include = DID_1001_default

[DID_1001_default]
exten = s,1,NoOp,${CALLERID(num)}-${CALLERID(name)}
exten = s,n,Answer
exten = s,n,zapateller(nocallerid)  ; torture telemarketers
exten = s,n,DigitTimeout,5 ; Set Digit Timeout to 5 seconds
exten = s,n,ResponseTimeout,10 ; Set Response Timeout to 10 seconds
exten = s,n,Hangup
include = default

[DID_1001_timeinterval_working day]
exten = _6888,1,Goto(default|6888|1)




If I call in at line2, then I can hear the Time announcement and I can
dial during that announcement an extension number.
BTW, where can I find the additional sounds I had at an previous setup
(If you know the extension, ...), which should replace the SayUnixTime

I have no idea how to get dial out to work. Can anybody give me a hint,
please?

In Asterisk I see:
[Nov 27 20:58:00] NOTICE[5095]: chan_sip.c:9227 sip_reg_timeout:--
Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #102)
-- Got SIP response 486 Busy Here back from *.133

*CLI sip show peers
1002/1002  *.133D  5060 Unmonitored

*CLI sip show users
1002   **
DID_1002 No   RFC3581  

*CLI sip show registry
*.133:5060  1002   120 Request Sent


bye

Ronald



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Re: [asterisk-users] [Solved] Wellgate Asterisk

[Ronald Wiplinger (Lists)]

Guillermo Salas M. wrote:
 El jue, 27-11-2008 a las 21:05 +0800, Ronald Wiplinger (Lists) escribió:
   
 I got a Wellgate 3804A and need some hints:

 Both have public IP *.131=asterisk (1.6.0.1) *.133= Wellgate

 Wellgate 3804A settings (Line1~Line4):
 


 I've one wellgate 3804 (old version) with 4 fxo ports integrated with
 asterisk 1.4.

 Regards,
  
   

I could solve it!
I had to add routing in the 3804A. Now both, dialin and dialout is working.

bye

Ronald

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[asterisk-users] [SOLVED] Re: Upgrade 1.4.19 to 1.6 = segementation fault

[Ronald Wiplinger (Lists)]

Ronald Wiplinger (Lists) wrote:
 During compiling I have not seen an error, however, when I start
 asterisk again it ends with:


 app_morsecode.so = (Morse code)
   == Registered custom function 'SYSINFO'
  func_sysinfo.so = (System information related functions)
 Segmentation fault (core dumped)


 How can I figure out what is wrong?
   
I removed all modules, which were left from the 1.4 installation and now
it works!



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[asterisk-users] Upgrade 1.4.19 to 1.6 = segementation fault

[Ronald Wiplinger (Lists)]

During compiling I have not seen an error, however, when I start
asterisk again it ends with:


app_morsecode.so = (Morse code)
  == Registered custom function 'SYSINFO'
 func_sysinfo.so = (System information related functions)
Segmentation fault (core dumped)


How can I figure out what is wrong?

bye

Ronald

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[asterisk-users] Snom - we are puzzled

[Ronald Wiplinger (Lists)]

we have installed asterisk and snom with PUBLIC IPs (IP/25) on one DSL line
we have for our office a different ADSL with one IP shared.

Two identical setup snom 360 (except the user name) with two public IP
addresses are connected at the hub to the server / DSL line

phone A can call B, B cannot call A, because A is not registered!!!

We disconnect A and setup a softphone (on the ADSL line with stun) and
it works.

How can I track down this problem.

bye

R.

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[asterisk-users] IP address on mysql cdr

[ronald ramos]

hi,

is it possible to store the IP address of the caller in the CDR? how about the 
end date/time? thank you.

regards,
ron



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[asterisk-users] dundi and regcontext

[ronald ramos]

hi,



when a user register on my asterisk i can see it adding Noop for that 
extension, but after awhile i won't see it anymore:



what are the reasons for it being removed on the dynamic context?

one thing i found when i unregister it's removed.



dialplan show myregcontext

[ Context 'myregcontext' created by 'SIP' ]

  '100500' =   1. Noop(100500)   [SIP]

  '112802' =   1. Noop(112802)   [SIP]



-= 2 extensions (2 priorities) in 1 context. =-



[ Context 'pfingobizsip' created by 'SIP' ]



-= 0 extensions (0 priorities) in 1 context. =-



my prob is when it's removed dundi cant find it anymore so a user 
calling from server 1 cannot call user that is in server 2.



i've set re-registration to very low (1 minute) to monitor if my phone 
re-register and to see if it will be added again on the regcontext.

but i don't even see it unregistering after 1 minute i only 
unregistering when i am using x-lite and closing x-lite, i dont see 
x-lite re-registering if i just leave the softphone open. any idea?



regards,

ron





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Re: [asterisk-users] iLBC codec

[ronald]

Hi Sir,

For this call i did not do anything except just call the extension

exten = 100,1,Dial(SIP/100|20|t|M(setmusiconhold,moh-100))

that's how i dial the extension, does musiconhold make asterisk 
uncompress? but during the call i did not use music on hold. whereelse 
should i look at?


Regards
Ron

Tilghman Lesher wrote:
 
 If you're doing anything at all that requires Asterisk to uncompress the
 audio (recording, mixing, conferencing, spying, etc.), then that would be the
 reason.  You cannot directly mix a compressed codec; you have to decompress
 the stream first.  Similarly, if you're recording, for example, a wav file,
 one of the steps in recording that wav file is to decompress the audio stream.
 

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Re: [asterisk-users] DUNDI Help

[ronald ramos]

Hi,

I have been testing dundi setup, one thing i am having problem with is that 
extensions are getting remove from the regcontext.

does it get removed when registration expires? how can i make sure it's added 
back without power cycling the phone? which would be better, making expiration 
higher? or lowering it so it will re-register  fast? also i am using pap2 and 
sipura, is there a settings to make re-register faster?

did you experience this as well before? how were you able to fix it? thank you

regards,
ron

--- On Wed, 8/27/08, Bruce Reeves [EMAIL PROTECTED] wrote:
From: Bruce Reeves [EMAIL PROTECTED]
Subject: Re: [asterisk-users] DUNDI Help
To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial 
Discussion asterisk-users@lists.digium.com
Date: Wednesday, August 27, 2008, 1:06 PM

Sure, let me show you how I setup dundi on systems.

extensions.conf

exten = _1X,1,Goto(lookupdundi,${EXTEN},1)

[lookupdundi]
exten = _X,1,Goto(${ARG1},1)
switch = DUNDi/priv

exten = i,1,Playback(invalid)

You can have the i do whatever you want, and you can use the same
option in the macro you are using.

That is it, I leave out all the other context in the examples, from
time to time I add a dundi-static context and put in specific numbers
or patterns I want to accept, maybe for pstn calling or phones that
don't register, but in those cases I have multiple mappings in
dundi.conf for each context. For example:

priv = sipregistrations,0,SIP,[EMAIL PROTECTED],nopartial
priv = dundi-static,0,SIP,[EMAIL PROTECTED],nopartial




On Wed, Aug 27, 2008 at 3:56 AM, ronald ramos [EMAIL PROTECTED]
wrote:
 Hi Again,

 Is there a way i can detect whether a user has been added into the
 regcontext?
 Currently i'm seeing this and just gives a fast busy.

 [Aug 27 16:44:46] WARNING[17402]: pbx.c:2483 __ast_pbx_run: Channel
 'SIP/10.10.10.10-b63101d0' sent into invalid extension
'141100' in context
 'lookupdundi', but no invalid handler

 can i detect it somehow, so i can inform user that the extensions is not
 available?

 i have tried ChanIsAvail, but since i am using  realtime ChanIsAvail
thinks
 it registered, since it really is registered on the other server. So
it's
 trying to call it,  tries  it for 30 secs (i set it to timeout at 30),
 after 30 secs then it will go to DUNDI/priv.  Is there a way that i can
 detect it first so it does not try to dial it on the local before askng
 dundi? thank you

 regards,
 Ron


 --- On Tue, 8/26/08, Bruce Reeves [EMAIL PROTECTED]
wrote:

 From: Bruce Reeves [EMAIL PROTECTED]
 Subject: Re: [asterisk-users] DUNDI Help
 To: [EMAIL PROTECTED], Asterisk Users Mailing List -
Non-Commercial
 Discussion asterisk-users@lists.digium.com
 Date: Tuesday, August 26, 2008, 8:16 PM

 It is added when a phone registers, or re-registers. Depending on the
 timing of the registrations and any restarts on the asterisk process
 it may take some time for phones to re-register.

 On Tue, Aug 26, 2008 at 2:10 PM, ronald ramos
[EMAIL PROTECTED]
 wrote:
 Hi Bruce,

 my apologies, but the error was because of the key.
 i just run keys init on the CLI and it works,

 question
  on regcontext though, i set it to sipregistrations, how often
 does
 an extension be added to the context sipregistrations and for how long
 will
 it stay there? i'm looking at dialplan show sipregistration,
sometimes
 i
 only see one extension there. even though i know i have 4 ip phones
 registered to the asterisk.

 TIA

 Ron


 --- On Tue, 8/26/08, Bruce Reeves [EMAIL PROTECTED]
 wrote:

 From: Bruce Reeves [EMAIL PROTECTED]
 Subject: Re: [asterisk-users] DUNDI Help
 To: [EMAIL PROTECTED], Asterisk Users Mailing List -
 Non-Commercial
 Discussion asterisk-users@lists.digium.com
 Date: Tuesday, August 26, 2008, 6:23 PM

 Ron,

 What does the peers section in dundi.conf look like?

 On Tue, Aug 26, 2008 at 3:00 AM, ronald
  ramos
 [EMAIL PROTECTED]
 wrote:
 Would like to try setting up dundi with 3-4 asterisk.
 But for poc, i would like to try setting up dundi on between 2
 asterisk.

 I copied the config from DUNDI enterprise SIP with no password.
Only
 thing
 i
 changed is the part where i used regcontext.
 on both boxes dundi.conf i have
 [mapping]
 priv = sipregistrations,0,SIP,[EMAIL PROTECTED],nopartial

 i can see both peers on each server:
 CLI dundi show  peers
 EID  HostModel  AvgTime 
Status
 00:8e:8c:8e:cb:5310.10.10.XX  (S) Symmetric  Unavail  OK (1
ms)

 i can see my extension being added
  on sipregistrations context
 Added extension '136101' priority 1 to
  sipregistrations

 tried a dundi lookup but got no result
 dundi lookup [EMAIL PROTECTED]
 DUNDi lookup returned no results.
 DUNDi lookup completed in 0 ms

 here's what's on extensions.conf

 ; Private DUNDi network
 [dundi-priv-canonical]
 ; Direct numbers

 [dundi-priv-customers]
 ; If you are an ITSP or Reseller, list your customers here.

 [dundi-priv-via-pstn]
 ; If you are freely delivering calls

Re: [asterisk-users] ultramonkey and asterisk

[ronald]

hi,

i think i'm getting somewhere (i hope) with this combo.

i have tried registering to the Virtual IP and i'm getting unauthorized.

i set sip debug to try and see the difference and found out i am missing 
this:

Authorization: Digest 
username=200200,realm=sip.mydomain.com,nonce=4cbc7dba,uri=sip:123.45.67.130,response=76dafea9c97c5d94506d1249b7fdafad,algorithm=MD5
Content-Length: 0

when i try to register my phone using the virtual ip of the ldirectord

but when i try to register using the actual ip address of the i can see 
that included on the REGISTER message. i am using x-lite.

any clues why the Authorization Part is not there when i use the Virtual IP?

TIA

Regards,
ronald


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Re: [asterisk-users] DUNDI Help

[ronald ramos]

Hi Again,

Is there a way i can detect whether a user has been added into the regcontext?
Currently i'm seeing this and just gives a fast busy.

[Aug 27 16:44:46] WARNING[17402]: pbx.c:2483 __ast_pbx_run: Channel 
'SIP/10..10.10.10-b63101d0' sent into invalid extension '141100' in context 
'lookupdundi', but no invalid handler

can i detect it somehow, so i can inform user that the extensions is not 
available?

i have tried ChanIsAvail, but since i am using  realtime ChanIsAvail thinks it 
registered, since it really is registered on the other server. So it's trying 
to call it,  tries  it for 30 secs (i set it to timeout at 30),  after 30 secs 
then it will go to DUNDI/priv.  Is there a way that i can detect it first so it 
does not try to dial it on the local before askng dundi? thank you

regards,
Ron


--- On Tue, 8/26/08, Bruce Reeves [EMAIL PROTECTED] wrote:
From: Bruce Reeves [EMAIL PROTECTED]
Subject: Re: [asterisk-users] DUNDI Help
To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial 
Discussion asterisk-users@lists.digium.com
Date: Tuesday, August 26, 2008, 8:16 PM

It is added when a phone registers, or re-registers. Depending on the
timing of the registrations and any restarts on the asterisk process
it may take some time for phones to re-register.

On Tue, Aug 26, 2008 at 2:10 PM, ronald ramos [EMAIL PROTECTED]
wrote:
 Hi Bruce,

 my apologies, but the error was because of the key.
 i just run keys init on the CLI and it works,

 question on regcontext though, i set it to sipregistrations, how often
does
 an extension be added to the context sipregistrations and for how long
will
 it stay there? i'm looking at dialplan show sipregistration, sometimes
i
 only see one extension there. even though i know i have 4 ip phones
 registered to the asterisk.

 TIA

 Ron


 --- On Tue, 8/26/08, Bruce Reeves [EMAIL PROTECTED]
wrote:

 From: Bruce Reeves [EMAIL PROTECTED]
 Subject: Re: [asterisk-users] DUNDI Help
 To: [EMAIL PROTECTED], Asterisk Users Mailing List -
Non-Commercial
 Discussion asterisk-users@lists.digium.com
 Date: Tuesday, August 26, 2008, 6:23 PM

 Ron,

 What does the peers section in dundi.conf look like?

 On Tue, Aug 26, 2008 at 3:00 AM, ronald ramos
[EMAIL PROTECTED]
 wrote:
 Would like to try setting up dundi with 3-4 asterisk.
 But for poc, i would like to try setting up dundi on between 2
asterisk.

 I copied the config from DUNDI enterprise SIP with no password. Only
thing
 i
 changed is the part where i used regcontext.
 on both boxes dundi.conf i have
 [mapping]
 priv = sipregistrations,0,SIP,[EMAIL PROTECTED],nopartial

 i can see both peers on each server:
 CLI dundi show  peers
 EID  HostModel  AvgTime  Status
 00:8e:8c:8e:cb:5310.10.10.XX  (S) Symmetric  Unavail  OK (1 ms)

 i can see my extension being added
  on sipregistrations context
 Added extension '136101' priority 1 to sipregistrations

 tried a dundi lookup but got no result
 dundi lookup [EMAIL PROTECTED]
 DUNDi lookup returned no results.
 DUNDi lookup completed in 0 ms

 here's what's on extensions.conf

 ; Private DUNDi network
 [dundi-priv-canonical]
 ; Direct numbers

 [dundi-priv-customers]
 ; If you are an ITSP or Reseller, list your customers here.

 [dundi-priv-via-pstn]
 ; If you are freely delivering calls to the PSTN, list them here

 [dundi-priv-local]
 include = dundi-priv-canonical
 include = dundi-priv-customers
 include = dundi-priv-via-pstn

 [dundi-priv-switch]
 ; Just a wrapper for the switch
 switch = DUNDi/priv

 [dundi-priv-lookup]
 include =
  dundi-priv-local
 include = dundi-priv-switch

 [macro-dundi-priv]
 exten = s,1,Goto(${ARG1}|1)
 include = dundi-priv-lookup

 [diallocal]
 exten = _1X,1,Macro(dundi-priv|${EXTEN})

 i also tried dialing from my xlite:
 [Aug 26 15:58:07] -- Executing [EMAIL PROTECTED]:1]
 Macro(SIP/138100-08269548, dundi-priv|136101)
in
 new stack
 [Aug 26 15:58:07] -- Executing [EMAIL PROTECTED]:1]
 Goto(SIP/138100-08269548, 136101|1) in new
stack
 [Aug 26 15:58:07] -- Goto (macro-dundi-priv,136101,1)
 [Aug 26 15:58:07]   == Auto fallthrough, channel
 'SIP/138100-08269548'
 status is 'UNKNOWN'

 any guess what's wrong? Thanks

 ron


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 --
 *
 Bruce Reeves, dCAp
 EUS Networks
 Office: 212-624-5943
 Web: www.euscorp.com
 


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[asterisk-users] DUNDI Help

[ronald ramos]

Would like to try setting up dundi with 3-4 asterisk.
But for poc, i would like to try setting up dundi on between 2 asterisk.

I copied the config from DUNDI enterprise SIP with no password. Only thing i 
changed is the part where i used regcontext.
on both boxes dundi.conf i have
[mapping]
priv = sipregistrations,0,SIP,[EMAIL PROTECTED],nopartial

i can see both peers on each server:
CLI dundi show  peers
EID  Host    Model  AvgTime  Status 
00:8e:8c:8e:cb:53    10.10.10.XX  (S) Symmetric  Unavail  OK (1 ms)  

i can see my extension being added on sipregistrations context
Added extension '136101' priority 1 to sipregistrations

tried a dundi lookup but got no result
dundi lookup [EMAIL PROTECTED]
DUNDi lookup returned no results.
DUNDi lookup completed in 0 ms

here's what's on extensions.conf

; Private DUNDi network
[dundi-priv-canonical]
; Direct numbers

[dundi-priv-customers]
; If you are an ITSP or Reseller, list your customers here.

[dundi-priv-via-pstn]
; If you are freely delivering calls to the PSTN, list them here

[dundi-priv-local]
include = dundi-priv-canonical
include = dundi-priv-customers
include = dundi-priv-via-pstn

[dundi-priv-switch]
; Just a wrapper for the switch
switch = DUNDi/priv

[dundi-priv-lookup]
include = dundi-priv-local
include = dundi-priv-switch

[macro-dundi-priv]
exten = s,1,Goto(${ARG1}|1)
include = dundi-priv-lookup

[diallocal]
exten = _1X,1,Macro(dundi-priv|${EXTEN})

i also tried dialing from my xlite:
[Aug 26 15:58:07] -- Executing [EMAIL PROTECTED]:1] 
Macro(SIP/138100-08269548, dundi-priv|136101) in new stack
[Aug 26 15:58:07] -- Executing [EMAIL PROTECTED]:1] 
Goto(SIP/138100-08269548, 136101|1) in new stack
[Aug 26 15:58:07] -- Goto (macro-dundi-priv,136101,1)
[Aug 26 15:58:07]   == Auto fallthrough, channel 'SIP/138100-08269548' status 
is 'UNKNOWN'

any guess what's wrong? Thanks

ron



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Re: [asterisk-users] DUNDI Help

[ronald ramos]

Hi Bruce,

my apologies, but the error was because of the key.
i just run keys init on the CLI and it works,

question on regcontext though, i set it to sipregistrations, how often does an 
extension be added to the context sipregistrations and for how long will it 
stay there? i'm looking at dialplan show sipregistration, sometimes i only see 
one extension there. even though i know i have 4 ip phones registered to the 
asterisk.

TIA

Ron


--- On Tue, 8/26/08, Bruce Reeves [EMAIL PROTECTED] wrote:
From: Bruce Reeves [EMAIL PROTECTED]
Subject: Re: [asterisk-users] DUNDI Help
To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial 
Discussion asterisk-users@lists.digium.com
Date: Tuesday, August 26, 2008, 6:23 PM

Ron,

What does the peers section in dundi.conf look like?

On Tue, Aug 26, 2008 at 3:00 AM, ronald ramos [EMAIL PROTECTED]
wrote:
 Would like to try setting up dundi with 3-4 asterisk.
 But for poc, i would like to try setting up dundi on between 2 asterisk.

 I copied the config from DUNDI enterprise SIP with no password. Only thing
i
 changed is the part where i used regcontext.
 on both boxes dundi.conf i have
 [mapping]
 priv = sipregistrations,0,SIP,[EMAIL PROTECTED],nopartial

 i can see both peers on each server:
 CLI dundi show  peers
 EID  HostModel  AvgTime  Status
 00:8e:8c:8e:cb:5310.10.10.XX  (S) Symmetric  Unavail  OK (1 ms)

 i can see my extension being added on sipregistrations context
 Added extension '136101' priority 1 to sipregistrations

 tried a dundi lookup but got no result
 dundi lookup [EMAIL PROTECTED]
 DUNDi lookup returned no results.
 DUNDi lookup completed in 0 ms

 here's what's on extensions.conf

 ; Private DUNDi network
 [dundi-priv-canonical]
 ; Direct numbers

 [dundi-priv-customers]
 ; If you are an ITSP or Reseller, list your customers here.

 [dundi-priv-via-pstn]
 ; If you are freely delivering calls to the PSTN, list them here

 [dundi-priv-local]
 include = dundi-priv-canonical
 include = dundi-priv-customers
 include = dundi-priv-via-pstn

 [dundi-priv-switch]
 ; Just a wrapper for the switch
 switch = DUNDi/priv

 [dundi-priv-lookup]
 include = dundi-priv-local
 include = dundi-priv-switch

 [macro-dundi-priv]
 exten = s,1,Goto(${ARG1}|1)
 include = dundi-priv-lookup

 [diallocal]
 exten = _1X,1,Macro(dundi-priv|${EXTEN})

 i also tried dialing from my xlite:
 [Aug 26 15:58:07] -- Executing [EMAIL PROTECTED]:1]
 Macro(SIP/138100-08269548, dundi-priv|136101) in
new stack
 [Aug 26 15:58:07] -- Executing [EMAIL PROTECTED]:1]
 Goto(SIP/138100-08269548, 136101|1) in new stack
 [Aug 26 15:58:07] -- Goto (macro-dundi-priv,136101,1)
 [Aug 26 15:58:07]   == Auto fallthrough, channel
'SIP/138100-08269548'
 status is 'UNKNOWN'

 any guess what's wrong? Thanks

 ron


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-- 
*
Bruce Reeves, dCAp
EUS Networks
Office: 212-624-5943
Web: www.euscorp.com




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[asterisk-users] set callerid with plus sign

[ronald]

Hi,

Is it possible to assign a plus sign on the callerid(num) ?

currently this is what i do CALLERID(num)=+6523450017

but telco is denying calls, coz they said they are seeing bs523450017 
instead of +6523450017.

i tried putting it inside double quotes CALLERID(num)=+6523450017 
telco says the same thing.

is this possible? thank you

Regards,
nhadie

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Re: [asterisk-users] set callerid with plus sign

[ronald]

Hi Sir,

I actually have a plus sign on my dial plan

exten = _+.,1,Dial (

that is ok, dialed number (telco refers to it as B-number) is correct.

the prob is the originating number(they call this A-Number), i want to 
set it to +65 so that it shows it is an international call.

so on my dial plan:

exten = _+.,1,Set(CALLERID(num)=+65)
exten = _+.,1,Dial(SIP/[EMAIL PROTECTED])

what i don't get is why +65 is being seen as bs5.

Regards,
Nhadie



Darren Sessions wrote:
 Just change your dial command and add the plus sign there.
 
 
 _
 
 Darren Sessions
 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
 http://www.darrensessions.com
 _
 
 
 
 
 
 On Aug 22, 2008, at 1:28 AM, ronald wrote:
 
 Hi,

 Is it possible to assign a plus sign on the callerid(num) ?

 currently this is what i do CALLERID(num)=+6523450017

 but telco is denying calls, coz they said they are seeing bs523450017
 instead of +6523450017.

 i tried putting it inside double quotes CALLERID(num)=+6523450017
 telco says the same thing.

 is this possible? thank you

 Regards,
 nhadie

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Re: [asterisk-users] set callerid with plus sign

[ronald]

Hi Thanks for all your reply.

Just figured out that ISUP does not decode plus sign very well.

regards
nhadie

Eric ManxPower Wieling wrote:
 + is not a valid Caller*ID character.  Asterisk allows you to use + in 
 Caller*ID, but many carriers will reject the call if you do that.
 
 Benny Amorsen wrote:
 ronald [EMAIL PROTECTED] writes:

 Is it possible to assign a plus sign on the callerid(num) ?
 Yes.

 currently this is what i do CALLERID(num)=+6523450017

 but telco is denying calls, coz they said they are seeing bs523450017 
 instead of +6523450017.
 Which techology? SIP? PRI? POTS? ...?


 /Benny


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[asterisk-users] ultramonkey and asterisk

[ronald]

hi all,

has anyone able to configure ultramonkey for sip (namely asterisk).
i tried from this tutorial:

http://blog.iclutton.com/2008/01/load-balancing-and-high-availablity.html

i have this on my ldirectord.cf:

virtual=123.45.67.155:5060
  real=123.45.67.130:5060 gate
  real=123.45.67.131:5060 gate
  service=sip
  scheduler=rr
  protocol=udp
  checktype=negotiate
  persistent=1

i was able to make my http and https to work but not sip.
hope someon could help me. thanks

regards,
nhadie

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[asterisk-users] disable auth between two asterisk

[ronald ramos]

Hi,

I have setup 2 asterisk talking  a single mysql cluster. I'm also using 
realtime db. I've setup sip peering between the two asterisk servers.

[asterisk-1]
insecure=port,invite
type=peer
host=201.202.203.204
context=from-asterisk-1

[asterisk-2]

insecure=port,invite

type=peer

host=201.202.203.205

context=from-asterisk-2


scenario:

ext 100 registers on Asterisk 1
ext 200 registers on Asterisk 2.

ext 100 calls ext 200. asterisk 1 receives request, asterisk 1 cannot find ext 
200, forward to asterisk 2, asterisk to sends back407  proxy auth required, 
asterisk 1 sends proxy auth back to UA (ext 100) but i'm not sure if ext 100 is 
replying with the needed credentials, because asterisk 2 replies with:

handle_response_invite: Failed to authenticate on INVITE to Ron sip:[EMAIL 
PROTECTED]

i tried to disabled the password on ext 100, tried the same scenario and call 
went thru.

so my assumption is a user registered on asterisk 1 cannot send calls to 
asterisk 2 coz when asterisk 2  asks for authentication, UA does not send it to 
asterisk 2, but i think it is sending it to asterisk 1. and vice versa if user 
is registered on asterisk 2, user  wont be able to make  calls to asterisk 1.

how can i disable proxy auth on the server if the user is already registered on 
the other astertisk.
i've set, insecure=port,invite but it still asks for proxy auth. anyone 
encountered this?

regards,
Ron





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[asterisk-users] Maybe a crazy idea, but are there Asterisk hoster outside there?

[Ronald Wiplinger]

I used to run an Asterisk server in the office, ... was looking for a
small replacement. I am not sure if that one is a good idea yet either.

How about this one:

I have VoIP phones, I have a Welgate 3804 (=2 FXO), all what I need is
an Asterisk server.

Is there a Asterisk hoster out there? Maybe as a virtual machine?

The mini solution does not have all features, but maybe this would still
allow me to turn off another machine here.

bye

Ronald

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[asterisk-users] I used to use an Asterisk server, but now it is overkill, ...

[Ronald Wiplinger]

I had installed in the office an Asterisk server, but the company is
gone and I could keep the server.

However, for my family with three members and two phone lines this
server is overkill. I am looking for a compact solution, which is more
suitable for me.

I want a small  silent box, which can connect two phone lines and 6
internal VoIP phones and about 6 external VoIP phones.
I would like to have:
1. Announcements for callers (dial the extension number)
2. voice mail with mail forwarding
3. wakeup call
4. pickup group
5. call forwarding after 20 seconds, ...
6. ISN support, Sipbroker support
7. remote gateway support

I guess that is all what I would need at home.

What is your suggestion for that?

bye

Ronald

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[asterisk-users] how to know what codec is being used

[ronald ramos]

Hi,

how would i know what codec is being utilized? currently i have set allow=ilbc 
disallow=all.
i unset all codecs on x-lite except ilbc.

i tried to make a call and look at the channel i see these. does this mean it 
is using ulaw? how about writetranscode? does that mean there is no transcoding 
happening on the call? call is going thru, rtp is also going thru. what i would 
like to know is does it really use ilbc? i'm using 1.4.18.1. thank you


core show channel SIP/19-082367b
  NativeFormats: 0x4 (ulaw)
    WriteFormat: 0x4 (ulaw)
 ReadFormat: 0x4 (ulaw)
 WriteTranscode: No
  ReadTranscode: No




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[asterisk-users] multiple asterisk approach

[ronald ramos]

Hi,

I'm not sure if this is the proper way to approach it but i can't figure out 
how to setup dundi.
what i did is, i try to determine which server a user is registered, by calling 
an agi to query  the realtime db and capture the regserver of  the user.

e.g.  

exten = _1xx,1,AGI(getserver.php)
exten = _1xx,2,GotoIf($[${REGSERVER} != asterisk-1]?102)
exten = _1xx,3,Dial(SIP/${EXTEN}|30|t)
exten = _1xx,102,Dial(SIP/[EMAIL PROTECTED]|30|t)
exten = _1xx,103,Hangup

then i created peering between the two. so far it is working i can call 
extensions that are registered in whatever server. but what i'd like to know 
is, would there be a difference on performance on calls when querying a DB to 
get the regserver, or is it still adviseable to use dundi for peering.

also i setup DNS SRV for these servers, what if one server fails, should the 
user close their phone to re-register to the server that is alive, or will it 
automtically register to the other server if the other is unreachable? TIA

Regards
Ron




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[asterisk-users] simultaneous dial macro

[ronald ramos]

Hi,

Would just like to know if it's possible to be able to call a macro at the same 
time.

i use a macro to dial local extension to another extension. 

exten = 100,Macro(dial-ext|SIP/100)
exten = 101,Macro(dial-ext|SIP/101)

but now i would like to use it on a simple ringgroup where it will ring all 
extensions
e.g. exten = s,Dial(SIP/100SIP/101)

how can i make use of my dial-ext macro instead of the simple Dial(SIP  SIP  
SIP)

thank you

regards,
ron




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Re: [asterisk-users] simultaneous dial macro

[ronald ramos]

hi,

thanks  for your reply. is dialgroup already available in asterisk 1.4?
i'm currently using 1.4.21.

regards,
ron

--- On Mon, 7/28/08, Pavel Jezek [EMAIL PROTECTED] wrote:
From: Pavel Jezek [EMAIL PROTECTED]
Subject: Re: [asterisk-users] simultaneous dial macro
To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial 
Discussion asterisk-users@lists.digium.com
Date: Monday, July 28, 2008, 7:52 PM

you can try to place your macro extensions into single dialgroup using 
DIALGROUP() function and then reference that dialgroup in dial aplication,
eg.
Set(DIALGROUP(test,add)=Local/100)
Set(DIALGROUP(test,add)=Local/101)
Dial(${DIALGROUP(test)})


ronald ramos wrote:
 Hi,

 Would just like to know if it's possible to be able to call a macro at
the same time.

 i use a macro to dial local extension to another extension. 

 exten = 100,Macro(dial-ext|SIP/100)
 exten = 101,Macro(dial-ext|SIP/101)

 but now i would like to use it on a simple ringgroup where it will ring
all extensions
 e.g. exten = s,Dial(SIP/100SIP/101)

 how can i make use of my dial-ext macro instead of the simple Dial(SIP
 SIP  SIP)

 thank you

 regards,
 ron




   
   
 

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[asterisk-users] need help setting up dundi

[ronald ramos]

Hi,

Hope anyone can help me on DUNDi. I got 2 asterisk servers. configs below.
tried this on the cli:

*CLI dundi lookup [EMAIL PROTECTED] bypass
DUNDi lookup returned no results.
DUNDi lookup completed in 0 ms

*CLI dundi lookup [EMAIL PROTECTED] bypass
DUNDi lookup returned no results.
DUNDi lookup completed in 0 ms

dundi debug shows this, i have no idea what that means though:
[Jul 24 02:42:39] Tx-Frame Retry[No] -- OSeqno: 000 ISeqno: 000 Type: 
NULL (Command)
[Jul 24 02:42:39]  Flags: 00 STrans: 23177  DTrans: 0 [10.10.10.1:4520] 
(Final)
[Jul 24 02:42:39] Rx-Frame Retry[No] -- OSeqno: 000 ISeqno: 001 Type: 
ACK  (Response)
[Jul 24 02:42:39]  Flags: 00 STrans: 05678  DTrans: 23177 [10.10.10.1:4520] 
(Final)

any mistake on my config?

regards,
ron

asterisk#1 (IP ADDRESS:10.10.10.1)
dundi.conf
[mappings]
priv = dundi-priv-canonical,0,SIP,[EMAIL PROTECTED],nopartial

[AB:CD:EF:70:E9:DA]
model = symmetric
host = 10.10.10.2
inkey = dundi
outkey = dundi
include = priv
permit = priv
qualify = yes
order = primary

sip.conf
[4000]
type=friend
nat=yes
secret=4000
host=dynamic

[priv]
type=peer
context=dundi-priv-canonical

extensions.conf
[dundi-priv-canonical]
exten = _4XXX,1,Dial(SIP/${EXTEN})

[dundi-priv-local]
include = dundi-priv-canonical

[dundi-priv-switch]
switch = DUNDi/priv

[dundi-priv-lookup]
include = dundi-priv-local
include = dundi-priv-switch

[macro-dundi-priv]
exten = s,1,Goto(${ARG1},1)
include = dundi-priv-lookup






asterisk #2 (IP ADDRESS:10.10.10.2)

dundi.conf
[mappings]
priv = dundi-priv-canonical,0,SIP,[EMAIL PROTECTED],nopartial

[00:1E:8C:AB:CD:EF]
model = symmetric
host = 10.10.10.1
inkey = dundi
outkey = dundi
include = priv
permit = priv
qualify = yes
order = primary

sip.conf
[4001]
type=friend
nat=yes
secret=4001
host=dynamic

[priv]
type=peer
context=dundi-priv-canonical

extensions.conf
[dundi-priv-canonical]
exten = _4XXX,1,Dial(SIP/${EXTEN})

[dundi-priv-local]
include = dundi-priv-canonical

[dundi-priv-switch]
switch = DUNDi/priv

[dundi-priv-lookup]
include = dundi-priv-local
include = dundi-priv-switch

[macro-dundi-priv]
exten = s,1,Goto(${ARG1},1)
include = dundi-priv-lookup




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[asterisk-users] Asterisk PBX How-to Guide for Amazon EC2

[Ronald Lewis]

I've just added a PREVIEW release of my upcoming how-to guide for Asterisk
PBX on EC2. It is based on months of testing and evaluating Asterisk on EC2.
It addresses all kinks and showstoppers that many people have experienced
over the past year or so. Because this is a preview, it is not the final
version of this guide. It is subject to change (format, copy, layout, etc.)

To view and download this guide, please visit
http://ronaldlewis.com/2008/07/08/asterisk-pbx-on-amazon-ec2-how-to-guide-almost-complete/

Please take this opportunity to test the guide and provide any feedback. The
official release is set for Wednesday, July 16 and will be available on
CloudCrunch.

Thanks!

Ronald Lewis
Denver, Colorado
http://ronaldlewis.com
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[asterisk-users] Play Beep if 1 minute remaining on Abosulte timeout

[ronald ramos]

Hi,

I have this dialpan to call international:

exten =gt; _00.,1,SET(TIMEOUT(absolute)=300)
exten =gt; _00.,n,Dial(SIP/[EMAIL PROTECTED])
exten =gt; _00.,n,NoCDR()
exten =gt; _00.,n,Hangup

Is there a way to check if there is only 1 minute remaining on the absolute 
timeout?

also an additional question, i can make call using that dialplan, but when the 
remote end hangs up first, asterisk does not see the hangup so it does not 
disconnect the ip phone. is this a prob on my config or the gateway that i send 
the calls to?

thank you
regards

ronramos





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[asterisk-users] remote server with Snom 190

[Ronald Wiplinger]

I have a local asterisk 1.2 and a remote asterisk 1.4.

Snom 190 can be used with the local asterisk but not with the remote one.

I need some hints where to track down this issue.

Some information:
Snom 190:
Line 1:
Account: 615
Password: OnlyIknowit
Registrar: ast.mydomain.com
Status:  OK

Line 2:
Account: 6888
Password: Otherside
Registrar: 22.33.44.55   (only IP address!)
Status:  Not found

Function keys:
P1   Line   Number  sip:[EMAIL PROTECTED];user=phone
P2   Line   Number  sip:[EMAIL PROTECTED];user=phone

Remote server is a fresh installed Ubuntu 8.04 server.

What do I miss?

bye

Ronald

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[asterisk-users] trying directrtpsetup

[ronald ramos]

Hi,


I recently installed asterisk, i used sterisk-1.4.20.1, i i set directrtpsetup 
to yes, no whow would i know if the rtp/media is not passing to asterisk. any 
tool or can u just sniff?

regards,
ron


  


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[asterisk-users] install asterisk on linux that uses software raid

[ronald ramos]

hi all,

we recently bought a clone box, motherboard with ICH7R raid controller (which i 
thought was a hardware raid controller). but recently i learned that those 
things are called FRAID( Fake RAID) which is basically a software raid also. so 
i decide to just use Software RAID (using CentOS 5.1).

has anyone installed asterisk on such configuration? is there any prob with 
regards to performance or quality of calls? thank you any info will be 
appreciated.

regards,
ron




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[asterisk-users] cdr question

[ronald ramos]

Hi,

Would just like to ask about cdr, i have an asterisk and i would like to bill 
only outbound calls not extension to extension, when i'm looking at the CDR, i 
can't figure out which fields i need to filter all outbound calls only. 

e.g if i dial 00. or 9XX (for local pstn calls) those are billable, 100 101 
or 102 (all local extensions) not billable.
*97 for voicemail not billable, but still is being logged on the cdr, can i 
disable logging to cdr calls like that(*98,*1,etc.)?

also, the time the call ended is not logged, is there a way to log that?

TIA

ron



   
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[asterisk-users] ring group question

[ronald ramos]

Hi All, 
 
I'm trying to configure a ringgroup, which will ring the extension in  the 
group one by one. this is what i tried on my extension.conf 
 
[macro-dial-ringgroup] 
exten = s,1,Dial(SIP/${ARG1},15) 
exten = s,n,NoOp( Dial Status: ${DIALSTATUS}) 
exten = s,n,Goto(s-${DIALSTATUS},1) 
exten = s-CHANUNAVAIL,1,SetCallerId(${CALLERIDNUM}) 
exten = s-CHANUNAVAIL,n,Dial(SIP/${ARG1},15) 
exten = s-BUSY,1,SetCallerId(${CALLERIDNUM}) 
exten = s-BUSY,n,Dial(SIP/${ARG1},15) 
exten = s-NOANSWER,1,SetCallerId(${CALLERIDNUM}) 
exten = s-NOANSWER,n,Dial(SIP/${ARG1},15) 
 
[ringgroup-1] 
exten = 5000,1,Macro(dial-ringgroup,1100) 
exten = 5000,n,Macro(dial-ringgroup,1101) 
exten = 5000,n,Macro(dial-ringgroup,1102) 
exten = 5000,n,Hangup 
 
 
so when i dial 5000 it will ring 1100 no answer,or busy on 1100. 
it will go to another extension which is 1101 and so on. 
 
I have tried 5000,1,Dial(SIP/1100SIP/1100) --- this one works,  
ringing at the same time, how can i do it in sequential?
 
 hope anyone can help me. thank you 

Ron
 
   
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[asterisk-users] followme scenarios

[ronald ramos]

Hi All,

I'm tryng to test different scenarios for followme for different users:

[localext]
exten = 101,1,Set(FM = ALWAYS);
exten = 101,n,Macro(dial-ext|SIP/${EXTEN}|vm-1|moh-101|fm-101);
exten = 101,n,Hangup
exten = 102,1,Set(FM = NEVER);
exten = 102,n,Macro(dial-ext|SIP/${EXTEN}|vm-1|moh-102|fm-102);
exten = 102,n,Hangup
exten = 103,1,Set(FM = WHENBUSY);
exten = 103,n,Macro(dial-ext|SIP/${EXTEN}|vm-1|moh-103|fm-103);
exten = 103,n,Hangup
exten = 104,1,Set(FM = WHENUNAVAILABLE);
exten = 104,n,Macro(dial-ext|SIP/${EXTEN}|vm-1|moh-103|fm-103);
exten = 104,n,Hangup
exten = 105,1,Set(FM = CUSTOM);
exten = 105,n,Macro(dial-ext|SIP/${EXTEN}|vm-1|moh-103|fm-103);
exten = 105,n,Hangup

[macro-dial-ext]
exten = s,1,SetMusicOnHold(${ARG3})
exten = s,n,Dial(${ARG1},5,M(setmusiconhold,${ARG3}))
exten = s,n,GotoIf(FM = NEVER|?vm)
exten = s,n,GotoIf(FM = CUSTOM|?s-CUSTOM,1)
exten = s,n,GotoIf(FM = WHENUNAVAILABLE|?s-CHANUNAVAIL)
exten = s,n,GotoIf(FM = WHENBUSY|?s-BUSY)
exten = s-CHANUNAVAIL,1,Followme(${ARG4})
exten = s-BUSY,1,Followme(${ARG4})
exten = s-CUSTOM,1,GotoIftime(17:00-19:00|*|*|*?c-CUSTOM,n)
exten = s-CUSTOM,n,Followme(${ARG4})
exten = s,n,Followme(${ARG4})
exten = s,n(vm),Voicemail([EMAIL PROTECTED]|u)
exten = s,n,Playback(vm-goodbye)
exten = s,n,Hangup


but it just keeps on going to this line
exten = s,n,GotoIf(FM = NEVER|?vm)

ami using GotoIf correctly? or am i referring to the FM variable properly? and 
is there easier way of doing this? TIA

regards
Ron

   
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[asterisk-users] realtime errors

[ronald ramos]

Hi All,

I just started playing around with asterisk realtime,
added some extensions and started making test call,
sometimes i can call the extension sometimes i can't.

below are errors i see on the CLI, has anyone
encountered this before?

[settings]
sippeers = mysql,sipdb,sip_customer
sipusers = mysql,sipdb,sip_customer
extensions = mysql,sipdb,extensions_customer
voicemail = mysql,sipdb,voicemail_customer


[Apr  6 01:04:53] WARNING[18959]:
res_config_mysql.c:360 update_mysql: MySQL RealTime:
Failed to query database. Check debug for more info.  
 

[Apr  6 01:05:04] WARNING[18959]: app_voicemail.c:2262
inboxcount: Failed to obtain database object for
'asterisk'!   


regards,
nhadie


  

You rock. That's why Blockbuster's offering you one month of Blockbuster Total 
Access, No Cost.  
http://tc.deals.yahoo.com/tc/blockbuster/text5.com

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[asterisk-users] fax detection on sip trunk

[ronald ramos]

Hi,

Is it possible for me to detect fax on a sip trunk?

my provider has a fax service that can send/receive
fax.

is it possible that i use a that trunk as a telefax?
meaning i will try to detect if it's a fax, if it is i
will forward it to an extension that can handle fax if
not will forward it elsewhere.

thank you

regards

ron


  

You rock. That's why Blockbuster's offering you one month of Blockbuster Total 
Access, No Cost.  
http://tc.deals.yahoo.com/tc/blockbuster/text5.com

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[asterisk-users] audio disappeared after ztdummy install

[ronald ramos]

 Hi All, 
 
Can't explain what happened, last night i was setting the voicemail  
configuration, and it worked properly: 
 
-- Executing [EMAIL PROTECTED]:3] VoiceMailMain(SIP/1000100-08219db0,  
@VM-1000) in new stack 
-- SIP/1000100-08219db0 Playing 'vm-login' (language 'en') 
 
i can hear the audio playing here. earlier i started playing with  meetme, and 
since i don't have any zap cards, i chose to use ztdummy, 
 
-- Executing [EMAIL PROTECTED]:1] MeetMe(SIP/1000100-08206da8,  6000) 
in new stack 
  == Parsing '/etc/asterisk/meetme.conf': Found 
-- Created MeetMe conference 1023 for conference '6000' 
-- SIP/1000100-08206da8 Playing 'conf-getpin' (language 'en') 
-- SIP/1000100-08206da8 Playing 'conf-onlyperson' (language 'en') 
 
from that message asterisk is playing conf-getpin, so i entered my  conference 
pin number, even though i don't hear any audio, then it tried  to play 
conf-onlyperson, still i dont hear anhything. 
 
then i tried my voicemail retrieval 
 
-- Executing [EMAIL PROTECTED]:3]  VoiceMailMain(SIP/1000101-0822b6c0, 
@VM-1000) in new stack 
-- SIP/1000101-0822b6c0 Playing 'vm-login' (language 'en') 
 
same thing it's playing something but i don't hear anything. 
 
i tried playing around with my codecs, i even downloaded the alaw core  and 
extra sound files. what do you guys think happened? it was working  before i 
enabled ztdummy. 
 
i tested disabling the ztdummy then i can hear the audio at the  voicemail but 
conference of course does not work now. i'm using  zaptel-1.4.9.2, i tried 
downgrading to 1.4.8 down to 1.4.7. but still the same issue.
 
Regards, 
Nhadie 
 


   
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Re: [asterisk-users] audio disappeared after ztdummy install

[ronald ramos]

Hi,

For now i just turned off acpi. and it works now.
just dont know what's the connection of that though
:-)

i will try to do the things you guys suggested also
when i get the chance, thanks for you help!

regards,
nhadie


--- Tzafrir Cohen [EMAIL PROTECTED] wrote:

 On Sun, Mar 30, 2008 at 02:35:03PM -0400, Norman W.
 Franke wrote:
  All too common and largely undocumented. I had
 this same problem.
  
  Installing ztdummy changes Asterisk to use it for
 timing of playback,  
  apparently. Removing ztdummy fixed the problem.
 To get it all to  
  work, I had to upgrade to to at least kernel
 2.6.23.11 (previous  
  versions are either missing options are just
 broken.) 
 
 Which previous versions have you tried?
 
 I'll also note that the OP needs to get Zaptel
 working under Xen, which
 is probably a different issue than your own.
 
  After doing  
  this, I recompiled ztdummy and it worked. Note
 that you need to  
  enable the various and random kernel flags to make
 this work,  
  generally dealing with the high-performance timer.
 I enabled:
  
  HPET Timer Support
  Enhanced Real Time Clock Support
  HPET - High Precision Event Timer
  HPET Control RTC IRQ
  Allow mmap of HPET
  
  I'm not sure if you can eliminate some of those,
 but this works for  
  me and is stable.
 
 -- 
Tzafrir Cohen
 icq#16849755 
 jabber:[EMAIL PROTECTED]
 +972-50-7952406  
 mailto:[EMAIL PROTECTED]
 http://www.xorcom.com 
 iax:[EMAIL PROTECTED]/tzafrir
 
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[asterisk-users] rxfax does not work (anymore)

[Ronald Wiplinger]

Below is my extensions.conf for the fax part


[incoming_28345474]
;
;
;   BEGIN - Inbound call handlers
;
;
exten = 8862100,1,NoOp(${CALLERID(num)})
exten = 8862100,2,Background(if-u-know-ext-dial)
exten = 8862100,3,Set(CALLERID(num)=${CALLERID(num)})
exten = h,1,hangup()
include = fax2emailstart
include = local

[fax2emailstart]
exten = 3000,1,SetVar(CALLEDFAX=${EXTEN})  ; [EMAIL PROTECTED]
exten = 3000,2,Answer
exten = 3000,3,Macro(fax2emailservice)

exten = h,1,System(/var/lib/asterisk/scripts/fax2emailservice
${CALLERIDNUM} ${CALLEDFAX} ${EXTNAME} ${EXTEMAIL} ${FAXFILE}
${EXTCOMPANY})

[macro-fax2emailservice]
 exten =
s,1,SetVar(FAXFILE=/var/spool/asterisk-fax/${CALLEDFAX}/${UNIQUEID})
 exten = s,2,Set(EXTEMAIL=${DB(${MACRO_EXTEN}/xEmail)})
 exten = s,3,NoOP()
 exten = s,4,Set(EXTNAME=${DB(${MACRO_EXTEN}/xName)})
 exten = s,5,NoOP()
 exten = s,6,Set(EXTCOMPANY=${DB(${MACRO_EXTEN}/xCompany)})
 exten = s,7,rxfax(${FAXFILE}.tif)
 exten = s,103,SetVar([EMAIL PROTECTED])
 exten = s,104,Goto(7)
 exten = s,105,SetVar(EXTNAME=Ronald)
 exten = s,106,Goto(7)
 exten = s,107,SetVar(EXTCOMPANY=Elmit)
 exten = s,108,Goto(7)


When I call this PSTN number and dial the extension number 3000, then I
see that:

*CLI
[Jan 27 16:03:21] -- Zap/3-1 answered SIP/601-006a2970
[Jan 27 16:03:24] -- Executing NoOp(SIP/88621001-00728610,
88621001) in new stack
[Jan 27 16:03:24] -- Executing BackGround(SIP/88621001-00728610,
if-u-know-ext-dial) in new stack
[Jan 27 16:03:24] -- Playing 'if-u-know-ext-dial' (language 'en')
[Jan 27 16:03:28] -- Executing Set(SIP/88621001-00728610,
CALLERID(num)=88621001) in new stack
[Jan 27 16:03:32]   == CDR updated on SIP/88621001-00728610
[Jan 27 16:03:32] -- Executing SetVar(SIP/88621001-00728610,
CALLEDFAX=3000) in new stack
[Jan 27 16:03:32] -- Executing Answer(SIP/88621001-00728610, )
in new stack
[Jan 27 16:03:32] -- Executing Macro(SIP/88621001-00728610,
fax2emailservice) in new stack
[Jan 27 16:03:32] -- Executing SetVar(SIP/88621001-00728610,
FAXFILE=/var/spool/asterisk-fax/3000/1201421004.8) in new stack
[Jan 27 16:03:32] -- Executing Set(SIP/88621001-00728610,
[EMAIL PROTECTED]) in new stack
[Jan 27 16:03:32] -- Executing NoOp(SIP/88621001-00728610, ) in
new stack
[Jan 27 16:03:32] -- Executing Set(SIP/88621001-00728610,
EXTNAME=Ronald Wiplinger) in new stack
[Jan 27 16:03:32] -- Executing NoOp(SIP/88621001-00728610, ) in
new stack
[Jan 27 16:03:32] -- Executing Set(SIP/88621001-00728610,
EXTCOMPANY=Elmit.com) in new stack
[Jan 27 16:03:32] -- Executing RxFAX(SIP/88621001-00728610,
/var/spool/asterisk-fax/3000/1201421004.8.tif) in new stack
vpbx*CLI
Disconnected from Asterisk server


I have no idea why it disconnects and hope somebody can help me to get
to work.

bye

Ronald


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[asterisk-users] Upgrade fails, need system upgrade advice

[Ronald Wiplinger]

I have a AMD64 CPU and use SuSE 9.2 with kernel 2.6.8-18
I tried to upgrade svn version 1.4.x but it fails at each part and
mainly because the system is with 1100 days getting to old.

I have to make a decision and need your advice.

CPU AMD64 3200+
1 GB RAM
Digium card with 2 FXS and 2 FXO
external Wellgate box 3804

I want to keep my current settings (backup /etc/asterisk and
/var/lib/asterisk and /var/spool/asterisk)
I use festiva
I need multiple fax on different extensions
I would like to run also OpenSer on the same machine


I would like to re-install a new system with svn asterisk 1.4.x and the
above settings.
Would you suggest me to install
a. OpenSuse 10.x
b. Ubuntu desktop
c. Ubuntu server

Any other hints? to backup directories? or just use a new hard disk.
With LVM?

bye

Ronald

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Re: [asterisk-users] dial extension number

[Ronald Wiplinger]

Ronald Wiplinger wrote:
 Can anybody give me a hint, please.

 I have a Welltech FXO device and from PSTN coming calls will be
 transfered to the extension number 1001.
 I want that the caller can reach the extension number  by dialing
 said number.

 My 1st try was:

 exten = 1001,1,NoOp(${CALLERID(num)})
 exten = 1001,2,Wait(1)
 exten = 1001,3,Set(CALLERID(num)=${CALLERID(num)})
 ;
 include = local; all extensions inhouse  (including )


 Above any dialed number will be ignored.


 Replaceing the second line (Wait) with:
 exten = 8862100,2,Background(if-u-know-ext-dial)
 the extension will be dialed.


 I do not want to have an announcement to ask for the dialing the
 extension number. What can I use instead?

   

I tried now WaitExten(10), but that is not recognizing dialing as well.


 Thanks!

 bye

 Ronald

   


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[asterisk-users] Help needed for Fax2Email with Welltech FXO 3804

[Ronald Wiplinger]

I have this in my extension.conf:

[incoming_28345474]
; 8862100 is the hotline number of the Welltech 3804
;
exten = 8862100,1,NoOp(${CALLERID(num)})
exten = 8862100,2,Wait(1)
exten = 8862100,3,Set(CALLERID(num)=${CALLERID(num)})
include = fax2emailstart

[fax2emailstart]
exten = 3000,1,SetVar(CALLEDFAX=${EXTEN}); me
exten = 3000,2,Answer
exten = 3000,3,Macro(fax2emailservice)

exten = 3001,1,SetVar(CALLEDFAX=${EXTEN}); dave
exten = 3001,2,Answer
exten = 3001,3,Macro(fax2emailservice)

exten = h,1,System(/var/lib/asterisk/scripts/fax2emailservice
${CALLERIDNUM} ${CALLEDFAX} ${EXTNAME} ${EXTEMAIL} ${FAXFILE}
${EXTCOMPANY})

[macro-fax2emailservice]
 exten =
s,1,SetVar(FAXFILE=/var/spool/asterisk/fax/${CALLEDFAX}/${UNIQUEID})
; exten = s,2,DBGet(EXTEMAIL=${MACRO_EXTEN}/xEmail)
 exten = s,2,Set(EXTEMAIL=${DB(MACRO_EXTEN/xEmail)})
 exten = s,3,NoOP()
 exten = s,4,Set(EXTNAME=${DB(MACRO_EXTEN/xName)})
 exten = s,5,NoOP()
 exten = s,6,Set(EXTCOMPANY=${DB(MACRO_EXTEN/xCompany)})
 exten = s,7,rxfax(${FAXFILE}.tif)
 exten = s,103,SetVar([EMAIL PROTECTED])
 exten = s,104,Goto(7)
 exten = s,105,SetVar(EXTNAME=Ronald)
 exten = s,106,Goto(7)
 exten = s,107,SetVar(EXTCOMPANY=Boss)
 exten = s,108,Goto(7)


CLI shows:

[Jan 14 22:58:51] -- Zap/3-1 answered SIP/601-006c3610
[Jan 14 22:58:54] -- Executing NoOp(SIP/88621001-007263d0,
88621001) in new stack
[Jan 14 22:58:54] -- Executing Wait(SIP/88621001-007263d0, 1) in
new stack
[Jan 14 22:58:55] -- Executing Set(SIP/88621001-007263d0,
CALLERID(num)=88621001) in new stack
[Jan 14 22:59:05] WARNING[20366]: pbx.c:2415 __ast_pbx_run: Timeout, but
no rule 't' in context 'incoming_28345474'
[Jan 14 22:59:05] -- Executing System(SIP/88621001-007263d0,
/var/lib/asterisk/scripts/fax2emailservice 88621001 )
in new stack
[Jan 14 22:59:09] -- Executing NoOp(SIP/88621001-006f8ea0,
88621001) in new stack
[Jan 14 22:59:09] -- Executing Wait(SIP/88621001-006f8ea0, 1) in
new stack
[Jan 14 22:59:10] -- Executing Set(SIP/88621001-006f8ea0,
CALLERID(num)=88621001) in new stack
[Jan 14 22:59:20] WARNING[20389]: pbx.c:2415 __ast_pbx_run: Timeout, but
no rule 't' in context 'incoming_28345474'
[Jan 14 22:59:20] -- Executing System(SIP/88621001-006f8ea0,
/var/lib/asterisk/scripts/fax2emailservice 88621001 )
in new stack
[Jan 14 22:59:20] -- Got SIP response 486 Busy Here back from
192.168.250.244
[Jan 14 22:59:21] -- Got SIP response 486 Busy Here back from
192.168.250.244
[Jan 14 22:59:21] -- Got SIP response 486 Busy Here back from
192.168.250.244
[Jan 14 22:59:22] -- Got SIP response 486 Busy Here back from
192.168.250.244
[Jan 14 22:59:23] -- Hungup 'Zap/3-1'
[Jan 14 22:59:24] -- Executing NoOp(SIP/88621001-006f3160,
88621001) in new stack
[Jan 14 22:59:24] -- Executing Wait(SIP/88621001-006f3160, 1) in
new stack
[Jan 14 22:59:25] -- Executing Set(SIP/88621001-006f3160,
CALLERID(num)=88621001) in new stack
[Jan 14 22:59:30] -- Executing System(SIP/88621001-006f3160,
/var/lib/asterisk/scripts/fax2emailservice 88621001 )
in new stack


I dial the number 28345474 and as soon the dialtone is to hear I dial
3000, but that is not shown in CLI. What am I missing?

bye

Ronald



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[asterisk-users] Multiple fax extensions

[Ronald Wiplinger]

I need to setup multiple fax extension numbers.
What is the best way to do that?

It should send the fax as pdf to the assigned email address (or
addresses) of that extension number.
It should also move the fax to a web site for online view.
It should - if possible - try to make OCR text file as email body.

Thanks for your hints.

bye

Ronald

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[asterisk-users] I want to record each phone call

[Ronald Wiplinger]

1. Instead of using *1 (automon) I need to record each phone call at a 
certain * box.

2. While already talking about this. I want to autodelete with cron at 2 
am in the morning all recordings which are older than 50 hours! How can 
I do that?

bye

Ronald

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[asterisk-users] SVN update

[Ronald Wiplinger]

I haven't updated for a while and when I looked on the web site how to 
do a SVN update, I cannot find it anymore.


CLI show version
Asterisk SVN-branch-1.2-r42600M built by root @ asterisk on a x86_64 
running Linux on 2006-09-10 22:52:42 UTC


1. Where is the description for the SVN update now?
2. Is there anything I have to take care of when updating from such an 
old version?


Thanks!

bye

Ronald
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[asterisk-users] GTalk/Jabber passing audio in 1.4.1!

[Ronald Lewis]

I've just compiled Asterisk 1.4.1 and I'm happy to report that I've got
two-way audio between Google Talk and Asterisk! This IS an exciting moment
today in VoIP! This is just GREAT!

- Ronald Lewis
http://ronaldlewis.com
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[asterisk-users] Bandwidth shapping device

[Ronald Wiplinger]

I have a link to a building (e.g. 10Mb/s) and want to split up the 
bandwidth to different users. Each user should get e.g.,  512kB/s plus 
256kB/s dedicated for VoIP.


What kind of device can I use for that ?  (managing switch ??? which one?)


bye

Ronald Wiplinger
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[asterisk-users] MRTG with 4 graphs

[Ronald Wiplinger]

How can I set-up a MRTG with 4 graphs, whereby:

1   data in
2   data out
3   ONLY voice(/video) data in
4   ONLY voice(/video) data out



bye

Ronald Wiplinger
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[asterisk-users] moving WiFi phone

[Ronald Wiplinger]

Can anybody tell me how I can set-up multiple access points with 
overlapping coverage, so that a moving WiFi phone user can continuesly 
use the phone.



bye

Ronald Wiplinger
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[asterisk-users] SMS via VoIP and web

[Ronald Wiplinger]

Where can I get a starting point for setting up sms via VoIP and via web.

I want to send SMS from VoIP or web  to VoIP phones and GSM phones.

1. how to set-up?
2. which smsc should I use? (what is the price?)
3. which phones can be used?


bye

Ronald
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[asterisk-users] Asterisk is used in U.S. prisons?

[Ronald Lewis]

So says The Voice of Asterisk, Allison Smith in this new and informative
interview:

http://www.ronaldlewis.com/interviews/2007/01/interview-with-allison-smith-north.html

(I know this isn't the most appropriate place, but Allison is about as
relevant as Mark Spencer and the community)
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Re: [asterisk-users] [resolved] asterisk 1,4 and google talk

[Ronald Lewis]

1.4 has been released, and it's still crashing. I guess it hasn't been
resolved yet.

On 11/10/06, Mik Cheez [EMAIL PROTECTED] wrote:


Mani,

I've gotten the same result both dialing from a gtalk client to SIP, as
well as an SIP call to gtalk.  You can run a jabber debug before the
call is placed to see more debug info on what's causing the crash.  With
the module in Beta, I believe it's just a bug that needs to be worked
out.  Below you'll see the output of one of my calls.

:M

sysmast01*CLI
JABBER: gtalk_account INCOMING: iq
to=[EMAIL PROTECTED]/asterisk4273D1E7 type=set id=35
from=[EMAIL PROTECTED]/Talk.v1001EE54E14session type=initiate
id=2077360010 initiator=[EMAIL PROTECTED]/Talk.v1001EE54E14
xmlns=http://www.google.com/session;description xml:lang=en
xmlns=http://www.google.com/session/phone;payload-type id=103
name=ISAC clockrate=16000/payload-type id=97 name=IPCMWB
clockrate=16000 bitrate=8/payload-type id=99 name=speex
clockrate=16000 bitrate=22000/payload-type id=4 name=G723
clockrate=8000 bitrate=6300/payload-type id=98 name=speex
clockrate=8000 bitrate=11000/payload-type id=100 name=EG711U
clockrate=8000 bitrate=64000/payload-type id=101 name=EG711A
clockrate=8000 bitrate=64000/payload-type id=0 name=PCMU
clockrate=8000 bitrate=64000/payload-type id=8 name=PCMA
clockrate=8000 bitrate=64000/payload-type id=13 name=CN
clockrate=8000/payload-type id=102 name=iLBC clockrate=
sysmast01*CLI
JABBER: gtalk_account INCOMING: 8000 bitrate=13300/payload-type
id=106 name=telephone-event
clockrate=8000//descriptiontransport
xmlns=http://www.google.com/transport/p2p//session/iq
sysmast01*CLI *** glibc detected *** /usr/sbin/asterisk:
munmap_chunk(): invalid pointer: 0xb7e47b73 ***
=== Backtrace: =
/lib/libc.so.6(cfree+0x1bb)[0x9b667b]
/usr/lib/asterisk/modules/chan_gtalk.so[0x82bde5]
/usr/lib/asterisk/modules/chan_gtalk.so[0x82c436]
/usr/lib/libiksemel.so.3(iks_filter_packet+0x129)[0x278789]
/usr/lib/asterisk/modules/res_jabber.so[0x4000c7]
/usr/lib/libiksemel.so.3[0x276b55]
/usr/lib/libiksemel.so.3(iks_parse+0x5c1)[0x274ad1]
/usr/lib/libiksemel.so.3(iks_recv+0x98)[0x276488]
/usr/lib/asterisk/modules/res_jabber.so[0x3fbd70]
/usr/sbin/asterisk[0x80eadfb]
/lib/libpthread.so.0[0xac03db]
/lib/libc.so.6(clone+0x5e)[0xa1a06e]


Mani Sridhar wrote:
 hi,
 it turns out that the iksemel library (which i installed using an rpm)
 was returning 0 when the function iks_has_tls() was called. it should
 return 1 otherwise res_jabber.o thinks gnuTLS is not installed. i
 confirmed this by running a test program i wrote, that calls
 iks_has_tls . it returned 0.

 i downloaded iksemel source, compiled it and now the test program
 returned 1.

 now, jabber show connected shows the google talk account as
 connected, but i don't see this buddy online on my other google talk
 buddy list.

 i added an extension in extensions.conf that calls Gtalk/buddy, and as
 soon as i call this extension, asterisk terminates due to a
 segmentation fault. it didn't seem like a core was dumped - i'm still
 looking for it.

 thanks
 sridhar

 _
 Live the life in style with MSN Lifestyle. Check out!
 http://content.msn.co.in/Lifestyle/Default

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--
Ronald Lewis
Producer, Interviews
Founder and CTA, Riverscape
http://www.ronaldlewis.com/interviews
http://www.riverscapecorp.com
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Re: [asterisk-users] Soundfiles adding during phone calls

[Ronald Wiplinger]

bails wrote:

Ronald Wiplinger wrote:

Ronald Wiplinger wrote:


Tom Lynn wrote:


Ron,
The guy is trying to help you.  Go to the link and read it.  There 
is a feature that you can use to play a recording into the voice 
channel.  Mine is set so when you press #9, the caller hears the 
lots of monkeys recording.


The best part of it is that you can hang up and the recording will 
continue to play to the caller.  When it expires, so does the call



I tried this:
features.conf
[featuremap]
blindxfer = ##; Blind transfer  was #1 - now press # twice
disconnect = *0; Disconnect
automon = *1; One Touch Record
atxfer = *2; Attended transfer


[applicationmap]
tortore= *9,callee,Playback,tt-monkeys


Yap, that magic word helped!

I got still some problems with it.
I understand that I do not hear the sound, but wonder if I should get 
the call back after the playback or not anymore.
In my experience the caller hang up and my phone remains on the status 
connected

I have only the choice to power cycle the phone.

Anything I can do ?


bye

Ronald Wiplinger
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Re: [asterisk-users] Soundfiles adding during phone calls

[Ronald Wiplinger]

Ronald Wiplinger wrote:

Tom Lynn wrote:

Ron,
The guy is trying to help you.  Go to the link and read it.  There is 
a feature that you can use to play a recording into the voice 
channel.  Mine is set so when you press #9, the caller hears the 
lots of monkeys recording.


The best part of it is that you can hang up and the recording will 
continue to play to the caller.  When it expires, so does the call


I tried this:
features.conf
[featuremap]
blindxfer = ##; Blind transfer  was #1 - now press # twice
disconnect = *0; Disconnect
automon = *1; One Touch Record
atxfer = *2; Attended transfer
tortore= *9,callee,Playback,tt-monkeys


extensions.conf
exten = 601,1,Set(DYNAMIC_FEATURES=hangup#play#tortore#automon) ; 
enable One-touch

exten = 601,2,Dial(${PHONE_601},30,tTwWr)


I make a call from 615 to 601
601 hits *9   but nothing happens!

when 601 hits *1  it records the conversion.




vpbx*CLI show features
Builtin Feature   Default Current
---   --- ---
Pickup*8  *8
Blind Transfer#   ##
Attended Transfer *2
One Touch Monitor *1
Disconnect Call   *   *0


Dynamic Feature   Default Current
---   --- ---
(none)

Call parking

Parking extension   :   750
Parking context :   parkedcalls
Parked call extensions: 751-770



I added already in extenions.conf:
include = featuremap





bye

Ronald Wiplinger






What do I miss?


bye

Ronald Wiplinger


On 11/11/06, * Ronald Wiplinger* [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED] wrote:


Andrew Joakimsen wrote:
 http://www.voip-info.org/wiki-Asterisk+config+features.conf

... and where exactly did you see this feature


bye

Ronald Wiplinger

 On 11/11/06, *Ronald Wiplinger * [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]  wrote:

 I want to add some sound filed on demand during a phone call
only
 possible on some extension numbers.


 I get many phone calls from local companies, but don't
understand
 Chinese! I would like to record the call, but also ask the
caller some
 questions, which should be added into the call with some
keys on the
 phone, ... e.g.  *66554 should add into the call: How are
you? or What
 is your phone number?


 But I do have another application for that too.
 I get many fake phone calls, where Chinese people tell you
that your
 phone bill is not paid, your court fee is not paid,  and
ask the
 caller to go to the ATM machine and key in a series of key
 strokes, 
 most likely it will clear out your account.
 For such fake callers I would like to add a terrible noise
to the
 call
 and make scare them as much as possible.

 Such fake calls I get now for each of my phone lines at 
least 10

 each!!!
 Either the caller-id is not set, is 0 or is a tollfree number.


 bye

 Ronald Wiplinger



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Re: [asterisk-users] Soundfiles adding during phone calls

[Ronald Wiplinger]

Tom Lynn wrote:

Ron,
The guy is trying to help you.  Go to the link and read it.  There is 
a feature that you can use to play a recording into the voice 
channel.  Mine is set so when you press #9, the caller hears the lots 
of monkeys recording.


The best part of it is that you can hang up and the recording will 
continue to play to the caller.  When it expires, so does the call


I tried this:
features.conf
[featuremap]
blindxfer = ##; Blind transfer  was #1 - now press # twice
disconnect = *0; Disconnect
automon = *1; One Touch Record
atxfer = *2; Attended transfer
tortore= *9,callee,Playback,tt-monkeys


extensions.conf
exten = 601,1,Set(DYNAMIC_FEATURES=hangup#play#tortore#automon) ; 
enable One-touch

exten = 601,2,Dial(${PHONE_601},30,tTwWr)


I make a call from 615 to 601
601 hits *9   but nothing happens!

when 601 hits *1  it records the conversion.

What do I miss?


bye

Ronald Wiplinger


On 11/11/06, * Ronald Wiplinger* [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED] wrote:


Andrew Joakimsen wrote:
 http://www.voip-info.org/wiki-Asterisk+config+features.conf

... and where exactly did you see this feature


bye

Ronald Wiplinger

 On 11/11/06, *Ronald Wiplinger * [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]  wrote:

 I want to add some sound filed on demand during a phone call
only
 possible on some extension numbers.


 I get many phone calls from local companies, but don't
understand
 Chinese! I would like to record the call, but also ask the
caller some
 questions, which should be added into the call with some
keys on the
 phone, ... e.g.  *66554 should add into the call: How are
you? or What
 is your phone number?


 But I do have another application for that too.
 I get many fake phone calls, where Chinese people tell you
that your
 phone bill is not paid, your court fee is not paid,  and
ask the
 caller to go to the ATM machine and key in a series of key
 strokes, 
 most likely it will clear out your account.
 For such fake callers I would like to add a terrible noise
to the
 call
 and make scare them as much as possible.

 Such fake calls I get now for each of my phone lines at least 10
 each!!!
 Either the caller-id is not set, is 0 or is a tollfree number.


 bye

 Ronald Wiplinger
 



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Re: [asterisk-users] Soundfiles adding during phone calls

[Ronald Wiplinger]

Tom Lynn wrote:

Ron,
The guy is trying to help you.  

Tom,

I believe it!
Go to the link and read it.  There is a feature that you can use to 
play a recording into the voice channel.  Mine is set so when you 
press #9, the caller hears the lots of monkeys recording.


I am not sure if that is correct:

feature.conf:

[applicationmap]
shout2caller =   *911,callee,Playback,shout-100dB   ;Shout to caller if 
*911 was pressed - use 'callee' or 'caller'
ask4name-Chinese = *910,callee,Playback,ask4name-Chinese; Ask 
caller for her/his name in Chinese


and in extensions.conf   

and where should Set(DYNAMIC_FEATURES=hangup#play#testfeature) 
be


and I want that only 601 and 621 can use this feature.



bye

Ronald Wiplinger



The best part of it is that you can hang up and the recording will 
continue to play to the caller.  When it expires, so does the call


On 11/11/06, * Ronald Wiplinger* [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED] wrote:


Andrew Joakimsen wrote:
 http://www.voip-info.org/wiki-Asterisk+config+features.conf

... and where exactly did you see this feature


bye

Ronald Wiplinger

 On 11/11/06, *Ronald Wiplinger * [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]  wrote:

 I want to add some sound filed on demand during a phone call
only
 possible on some extension numbers.


 I get many phone calls from local companies, but don't
understand
 Chinese! I would like to record the call, but also ask the
caller some
 questions, which should be added into the call with some
keys on the
 phone, ... e.g.  *66554 should add into the call: How are
you? or What
 is your phone number?


 But I do have another application for that too.
 I get many fake phone calls, where Chinese people tell you
that your
 phone bill is not paid, your court fee is not paid,  and
ask the
 caller to go to the ATM machine and key in a series of key
 strokes, 
 most likely it will clear out your account.
 For such fake callers I would like to add a terrible noise
to the
 call
 and make scare them as much as possible.

 Such fake calls I get now for each of my phone lines at least 10
 each!!!
 Either the caller-id is not set, is 0 or is a tollfree number.


 bye

 Ronald Wiplinger
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 ---
 avast! Antivirus: Inbound message clean.
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 Tested on: 2006/11/11 �U�� 11:07:21
 avast! - copyright (c) 1988-2006 ALWIL Software.
 http://www.avast.com







--
Ronald Wiplinger  (CEO of ELMIT)
http://www.elmit.com  http://voip.elmit.com   http://e-paper.elmit.com
Tel. (M) +886.939.775.516  (O) +886.2.2835.7765 (ENUM)   or FWD 511208
- I'm a SpamCon Foundation Member, #694, Verify it at
http://www.spamcon.org

PS: Spam prevention!
Our system is protected with a spam prevention program.
If you send us an e-mail, our system will send you a confirmation
message back. Just reply to this confirmation message please.
After receiving this confirmation message, our system will send
the hold message (one) and all future messages (after the received
confirmation message) to me without asking you again.

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Re: [asterisk-users] Soundfiles adding during phone calls

[Ronald Wiplinger]

Andrew Joakimsen wrote:

http://www.voip-info.org/wiki-Asterisk+config+features.conf


... and where exactly did you see this feature


bye

Ronald Wiplinger


On 11/11/06, *Ronald Wiplinger * [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED] wrote:


I want to add some sound filed on demand during a phone call only
possible on some extension numbers.


I get many phone calls from local companies, but don't understand
Chinese! I would like to record the call, but also ask the caller some
questions, which should be added into the call with some keys on the
phone, ... e.g.  *66554 should add into the call: How are you? or What
is your phone number?


But I do have another application for that too.
I get many fake phone calls, where Chinese people tell you that your
phone bill is not paid, your court fee is not paid,  and ask the
caller to go to the ATM machine and key in a series of key
strokes, 
most likely it will clear out your account.
For such fake callers I would like to add a terrible noise to the
call
and make scare them as much as possible.

Such fake calls I get now for each of my phone lines at least 10
each!!!
Either the caller-id is not set, is 0 or is a tollfree number.


bye

Ronald Wiplinger
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---
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Virus Database (VPS): 0647-0, 2006/11/09
Tested on: 2006/11/11 �U�� 11:07:21
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http://www.avast.com




  



--
Ronald Wiplinger  (CEO of ELMIT)
http://www.elmit.com  http://voip.elmit.com  http://e-paper.elmit.com 
Tel. (M) +886.939.775.516  (O) +886.2.2835.7765 (ENUM)   or FWD 511208

- I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org

PS: Spam prevention!
Our system is protected with a spam prevention program. 
If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. 
After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again.


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[asterisk-users] Soundfiles adding during phone calls

[Ronald Wiplinger]

I want to add some sound filed on demand during a phone call only 
possible on some extension numbers.



I get many phone calls from local companies, but don't understand 
Chinese! I would like to record the call, but also ask the caller some 
questions, which should be added into the call with some keys on the 
phone, ... e.g.  *66554 should add into the call: How are you? or What 
is your phone number?



But I do have another application for that too.
I get many fake phone calls, where Chinese people tell you that your 
phone bill is not paid, your court fee is not paid,  and ask the 
caller to go to the ATM machine and key in a series of key strokes,  
most likely it will clear out your account.
For such fake callers I would like to add a terrible noise to the call 
and make scare them as much as possible.


Such fake calls I get now for each of my phone lines at least 10 each!!!
Either the caller-id is not set, is 0 or is a tollfree number.


bye

Ronald Wiplinger
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[asterisk-users] Reg errors? Other anomalies? Check those capacitors!

[Ronald Lewis]

Three months ago, I was experiencing all sorts of issues with my Asterisk box maintaining a connection to multiple trunks, etc. I also experienced various timing issues as well. In addition, Asterisk would sometimes take almost a minute to fully load and register its SIP and IAX trunks.
Puzzled, I recompiled several times. No result. I checked my hardware. Didn't find anything. However, I did overlook one thing:* The motherboard's capacitor!Yep, you guessed it! It was bad. Now, I do not have any problems (I didn't bother replacing the motherboard, ended up using a spare PC).

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[asterisk-users] OT: (Job) Full-Time Asterisk Opportunity

[Ronald Lewis]

There is currently a permanent, full-time Asterisk opportunity available for the right candidate. The client is seeking to fulfill this position soon. Here are the particulars:* This position requires that you work from home, and be within a reasonable distance to a major airport
* You should be comfortable with a moderate amount of travel* You must have good working knowledge of Asterisk, which includes the ability to install and configure the PBX* a dCap (Digium certification) is a plus, but not required
* Experience with Python, C++, and/or other scripting languages are helpful, but not requiredPlease submit your resume to ron (at) ronaldlewis.com -- I will not respond to inquiries on the list.
Regards,Ronald LewisFounder and CTA, Riverscapewww.ronaldlewis.comwww.riverscapecorp.com
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Re: [asterisk-users] Re: Real-time and priority n

[Ronald Wiplinger]

Brian Capouch wrote:

Tony Mountifield wrote:

In article [EMAIL PROTECTED],
Ronald Wiplinger [EMAIL PROTECTED] wrote:


Is it exclusive? Either Realtime or priority n ???

If so, what is the better way?



I believe 'n' is just a shorthand way of writing previous line + 1,
and gets converted into an actual number as the dialplan is compiled.
After compilation, the information about whether a line had been given
as 'n' or as a specific number has been lost, as far as I know.



Rows can be added to a database table at any time.  Imagine a series 
of priorities added to a table using nothing more than n as a 
priority number beyond the first one.


Now imagine wanting to add a new priority in between any two arbitrary 
entries in the table.  How would you even specify which two lines 
should surround it, when they have no identifying serial number 
associated with them?


Unless you were to add a new field, e.g. priority location 
identifier, or somesuch.  Which does nothing more than move back to 
the present situation.


The extensions.conf parser adds a real priority to each line, but in 
Realtime that responsibility falls on the DB maintainer.


B.



Short: EXCLUSIVE

thanks!

bye

Ronald


--
Ronald Wiplinger  (CEO of ELMIT)
http://www.elmit.com  http://voip.elmit.com  http://e-paper.elmit.com 
Tel. (M) +886.939.775.516  (O) +886.2.2835.7765 (ENUM)   or FWD 511208

- I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org

PS: Spam prevention!
Our system is protected with a spam prevention program. 
If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. 
After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again.


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[asterisk-users] Real-time and priority n

[Ronald Wiplinger]

Is it exclusive? Either Realtime or priority n ???

If so, what is the better way?


bye

Ronald
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[asterisk-users] Issues with Monitor in 1.4?

[Ronald Lewis]

Has anyone noticed any anamolies with Monitor not recording in 1.4 beta2? I just did a half hour interview this morning, and for the FIRST time ever, Asterisk dropped the recording. The same also happened with a friend yesterday. I don't like this, because I RELY on Asterisk to do my interviews
-- Ronald LewisProducer, InterviewsFounder and CTA, Riverscapehttp://www.ronaldlewis.com/interviews
http://www.riverscapecorp.com
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Re: [asterisk-users] Issues with Monitor in 1.4?

[Ronald Lewis]

I've used various versions of Asterisk for many things ... this isn't necessarily a production thing. I'm fully aware of the nature of beta software (I've tested a lot of software in my time), and I'm simply asking for feedback ... right now, this type of feedback doesn't help, but thanks anyway.

On 9/30/06, Doug Lytle [EMAIL PROTECTED] wrote:
Ronald Lewis wrote: ever, Asterisk dropped the recording. The same also happened with a friend yesterday. I don't like this, because I RELY on Asterisk to do
Sorry, but I've gotta say it.Then you shouldn't be using BETA software in production.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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http://lists.digium.com/mailman/listinfo/asterisk-users-- Ronald LewisProducer, Interviews
Founder and CTA, Riverscapehttp://www.ronaldlewis.com/interviewshttp://www.riverscapecorp.com 
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[asterisk-users] Context default incoming ENUM

[Ronald Wiplinger]

I want to make the context [default]   as an alarm, for not having 
set-up correct.


I am looking for a way to get incoming calls via ENUM or via names (e.g. 
sip:[EMAIL PROTECTED]) into a defined context. How can I do that?


bye

Ronald
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[asterisk-users] Priority n

[Ronald Wiplinger]

How do I use priority n correct?

Here is the current example:

exten = 615,1,Dial(${PHONE_615},60,tr)
exten = 615,2,Voicemail,[EMAIL PROTECTED]
exten = 615,103,Voicemail,[EMAIL PROTECTED]

and:
exten = 617,109,GotoIf($[${DIALSTATUS} : 
(CHANUNAVAIL|CONGESTION)]?110:999)

exten = 617,110, .

exten = 617,999,hangup


That would greatly help me to throw out the NoOp statements I have 
inserted over the time if I tested some parts, ..


bye

Ronald
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[asterisk-users] WARNING: chan_sip.c add_realm_authentication: ???

[Ronald Wiplinger]

When I reloaded my asterisk I saw these lines, which I have noticed before:


[Sep 27 11:46:09] WARNING[27468]: chan_sip.c:12039 
add_realm_authentication: Format for authentication entry is 
user[:[EMAIL PROTECTED] at line 797
[Sep 27 11:46:09] WARNING[27468]: chan_sip.c:12039 
add_realm_authentication: Format for authentication entry is 
user[:[EMAIL PROTECTED] at line 822
[Sep 27 11:46:09] WARNING[27468]: chan_sip.c:12039 
add_realm_authentication: Format for authentication entry is 
user[:[EMAIL PROTECTED] at line 847
[Sep 27 11:46:09] WARNING[27468]: chan_sip.c:12039 
add_realm_authentication: Format for authentication entry is 
user[:[EMAIL PROTECTED] at line 872
[Sep 27 11:46:11]   == Parsing '/etc/asterisk/sip_notify.conf': [Sep 27 
11:46:11] Found




What does it mean? Should I care?

bye

Ronald
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[asterisk-users] Accounting and re-invite

[Ronald Wiplinger]

I am thinking if re-invite will interfere accounting.

Please help me to figure it out:

Phone A is registered at asterisk and calls a gateway. If the gateway 
allows re-invite than the rtp would go directly from phone A to the 
gateway, while the sip messages are still going through Asterisk. 
Asterisk will be informed when the call ended.
If it is a postpaid accounting, just bill the customer, however, how is 
it for a pre-paid (calling card user)?
I think Asterisk will have no power to turn off the call from A to the 
gateway.
Even more, if the gateway would allow to end a call and continue with a 
new call, the new call would not be billed (or would it)?


I guess the solution must be re-invite=no 
However, re-invite=no means that each call is going with rtp also 
through my server, what means for a remote phone, I have to provide for 
both legs the bandwidth.


Would here a rtpproxy or mediaproxy  help? If how and why?

bye

Ronald


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Re: [asterisk-users] University switches to Asterisk

[Ronald Lewis]

I stumbled upon this yesterday while reading my usual news sites, and added it to Digg.com -- so be sure to digg it for even more exposure -- 
http://digg.com/tech_news/University_Dumps_Cisco_VoIP_Moving_6_000_Students_to_AsteriskThis is a great example for Asterisk, since most folks remain quiet on its large-scale deployments.-- Ronald Lewis
Producer, InterviewsFounder and CTA, Riverscapehttp://www.ronaldlewis.com/interviewshttp://www.riverscapecorp.com
On 9/13/06, Doug Lytle [EMAIL PROTECTED] wrote:
Interesting article I found linked from Groklaw:Sam Houston State University replaces Cisco CallManagers, Nortel PBXswith Linux-based VoIP and messaging servers
http://www.networkworld.com/news/2006/091206-von-sam-houston.html?page=1Doug--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] pickupgroup 1

[Ronald Wiplinger]

I have problems with pickupgroup. While 621 can pickup a call to 601 
with *8, no phone can pickup a call to 621. Below are the settings for 
two phones. 601 is static in the sip.conf, while 621 is in the Real-time 
database. What could be the problem?


I have an extension 601:

[601]
type=friend
context=ELMIT
username=hotline
secret=shhshh
canreinvite=no
host=dynamic
;defaultip=61.220.121.19
dtmfmode=rfc2833
[EMAIL PROTECTED]
nat=yes
callgroup=1
pickupgroup=1
callerid=Ronald Hotline,601
qualify=1000

and and extension 621:

CREATE TABLE `sip_buddies` (
 `id` int(11) NOT NULL auto_increment,
 `name` varchar(80) NOT NULL default '',
 `accountcode` varchar(20) default NULL,
 `amaflags` varchar(13) default NULL,
 `callgroup` varchar(30) default NULL,
 `callerid` varchar(80) default NULL,
 `restrictcid` char(3) default 'NO',
 `canreinvite` char(3) default 'yes',
 `context` varchar(80) default NULL,
 `defaultip` varchar(15) default NULL,
 `dtmfmode` varchar(7) default NULL,
 `fromuser` varchar(80) default NULL,
 `fromdomain` varchar(80) default NULL,
 `host` varchar(31) NOT NULL default '',
 `incominglimit` int(2) default NULL,
 `outgoinglimit` int(2) default NULL,
 `insecure` varchar(4) default NULL,
 `language` char(2) default NULL,
 `mailbox` varchar(50) default NULL,
 `md5secret` varchar(80) default NULL,
 `nat` varchar(5) NOT NULL default 'yes',
 `permit` varchar(95) default NULL,
 `deny` varchar(95) default NULL,
 `mask` varchar(95) default NULL,
 `pickupgroup` varchar(10) default NULL,
 `port` varchar(5) NOT NULL default '',
 `qualify` varchar(4) default NULL,
 `rtptimeout` char(3) default NULL,
 `rtpholdtimeout` char(3) default NULL,
 `secret` varchar(80) default NULL,
 `type` varchar(6) NOT NULL default 'friend',
 `username` varchar(80) NOT NULL default '',
 `disallow` varchar(100) default 'all',
 `allow` varchar(100) default 'g729;ilbc;gsm;ulaw;alaw',
 `musiconhold` varchar(100) default NULL,
 `regseconds` int(11) NOT NULL default '0',
 `ipaddr` varchar(15) NOT NULL default '',
 `regexten` varchar(80) NOT NULL default '',
 `cancallforward` char(3) default 'yes',
 `fullcontact` varchar(80) default NULL,
 `setvar` varchar(100) NOT NULL default '',
 PRIMARY KEY  (`id`),
 UNIQUE KEY `name` (`name`),
 KEY `name_2` (`name`)
) TYPE=MyISAM ROW_FORMAT=DYNAMIC AUTO_INCREMENT=102 ;

--
-- Dumping data for table `sip_buddies`
--

INSERT INTO `sip_buddies` VALUES (1, '621', NULL, NULL, NULL, 'Ronald 
private,621', '', 'no', 'ELMIT', NULL, 'rfc2833', NULL, NULL, 
'dynamic', NULL, NULL, NULL, 'en', '[EMAIL PROTECTED]', NULL, 'yes', NULL, 
NULL, NULL, '1', '5060', '1000', NULL, NULL, 'shhshh', 'friend', '621', 
'all', 'ulaw;alaw;g729;gsm', NULL, 1158361126, '192.168.250.76', '', 
'yes', 'sip:[EMAIL PROTECTED]:5060', '');
  


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[asterisk-users] Help spread the word about Asterisk!

[Ronald Lewis]

Recently, Network World published an article about a Texas university migrating their 6,000 students from a Cisco VoIP solution to Asterisk. This is the best example to date of a large-scale Asterisk deployment, considering how secretive the numbers are and where.
So, help push this news to the top of digg.com by digging the URL below:http://digg.com/tech_news/University_Dumps_Cisco_VoIP_Moving_6_000_Students_to_Asterisk
-- Ronald LewisProducer, InterviewsFounder and CTA, Riverscapehttp://www.ronaldlewis.com/interviewshttp://www.riverscapecorp.com

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[asterisk-users] ASTCC: change from no pin to pin request?

[Ronald Wiplinger]

I want to change that ASTCC will ask for pin.

1. Where to set it? Pin length and number?
2. Can I set the pin only for a few people? E.g. Would deleting the 
pin number not ask for the pin or needs than still the # 

3. How to change the pin? Can the user change the pin?

bye

Ronald
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[asterisk-users] Makefile.moddir_rules: No such file or directory

[Ronald Wiplinger]

I need h.264 and tried therefore svn checkout 
http://svn.digium.com/svn/asterisk/trunk asterisk


(currently I have branches 1.2 installed)


make clean; make update; make install

.

make[1]: Entering directory `/usr/local/src/svn-versions/asterisk'
rm -f .depend
rm -f .depend
rm -f .depend
Makefile:60: /usr/local/src/svn-versions/asterisk/Makefile.moddir_rules: 
No such file or directory
make[2]: *** No rule to make target 
`/usr/local/src/svn-versions/asterisk/Makefile.moddir_rules'.  Stop.

make[1]: *** [channels-clean-depend] Error 2
make[1]: Leaving directory `/usr/local/src/svn-versions/asterisk'
make: *** [update] Error 2


Why is  Makefile.moddir_rules missing, or what have I forgotten to do?

bye

Ronald
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[asterisk-users] ast_parse_allow_disallow: Cannot allow unknown format 'h264'

[Ronald Wiplinger]

I see in CLI:

ast_parse_allow_disallow: Cannot allow unknown format 'h264'

What can I do ?
I see on Asterisk home page, that h264 is not listed.
When does Asterisk need h264 at all? If one phone calls another phone, 
than it is only passed through and does not need it, or am I wrong here?


BTW, if I use SER, would this be solved?

bye

Ronald
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[asterisk-users] svn trunk or branches ???

[Ronald Wiplinger]

My last update was a while back and as I remember svn trunk did not 
compile and I was advised to use branches 1.2 till further notice.


Have I missed the further notice and can we use now svn trunk or is the 
advice still to use branches 1.2 ???


bye

Ronald
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[asterisk-users] How to check which rtp ports my firewall let through?

[Ronald Wiplinger]

I thought with iptable -L |grep udp  I will find out which ports are 
open for the rtp stream,  but I cannot get this info from here, or 
at least I cannot interpret it:



# iptables -L |grep udp
ACCEPT udp  --  anywhere anywherestate 
RELATED,ESTABLISHED
LOGudp  --  anywhere anywherelimit: avg 
3/min burst 5 LOG level warning tcp-options ip-options prefix 
`SFW2-FWDdmz-DROP-DEFLT '
LOGudp  --  anywhere anywherelimit: avg 
3/min burst 5 LOG level warning tcp-options ip-options prefix 
`SFW2-FWDext-DROP-DEFLT '
LOGudp  --  anywhere anywherelimit: avg 
3/min burst 5 LOG level warning tcp-options ip-options prefix 
`SFW2-FWDint-DROP-DEFLT '
LOGudp  --  anywhere anywherelimit: avg 
3/min burst 5 LOG level warning tcp-options ip-options prefix 
`SFW2-INdmz-DROP-DEFLT '
ACCEPT udp  --  anywhere anywhereudp 
dpts:ndmp:dnp
ACCEPT udp  --  anywhere anywhereudp 
dpt:mgcp-callagent

ACCEPT udp  --  anywhere anywhereudp dpt:4569
ACCEPT udp  --  anywhere anywhereudp dpt:5036
ACCEPT udp  --  anywhere anywhereudp dpt:sip
LOGudp  --  anywhere anywherelimit: avg 
3/min burst 5 LOG level warning tcp-options ip-options prefix 
`SFW2-INext-DROP-DEFLT '
LOGudp  --  anywhere anywherelimit: avg 
3/min burst 5 LOG level warning tcp-options ip-options prefix 
`SFW2-INint-DROP-DEFLT '
REJECT udp  --  anywhere anywherereject-with 
icmp-port-unreachable



However, /etc/rc.d/SuSEfirewall2_final status includes the line:
   0 0 ACCEPT udp  *  *   ::/0 
::/0   udp dpts:1:2



Why I am looking for that?
My voice connection to phones is usually working, however, we have now 
also video phones and they do not receive any Video packages, 


bye

Ronald
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[asterisk-users] Need somebody for video phone testing

[Ronald Wiplinger]

I need somebody who can test with me video phone settings.

I use Eyebeam!
Please contact me via MSN first: [EMAIL PROTECTED]

bye

Ronald
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Re: [asterisk-users] Blind transfer 3/4 digits

[Ronald Wiplinger]

Koopmann, Jan-Peter wrote:

On Sunday, September 03, 2006 3:40 AM Ronald Wiplinger wrote:

  

try that way. However, I have doubts as well. If you are right, than
why snom phone does not have this problem? Would not here also the
first match count?   



Because the transfer button on the SNOM is using a totally different mechanism than 
sending # to Asterisk. On your snom configuration (like ours) the phone does not start to 
create/send a SIP message until you hit OK. At that time the entire number is 
there and a complete SIP transfer is created. Cool down a bit. The problem you are having 
is most probably just a dialplan problem. It takes some time and experience to get those 
things right. No need to yell here...
  

What's happen to you guys? I am not yelling, just asking.
It is sure not a dialplan question! If it would be a dialplan question, 
than it would be for each dialing, but it isn't.


You mentioned SIP message and that makes me wonder! Are we not using 
here dtmf ?? that is in my opinion not a sip message, isn't it?
If it is a sequence of tones, than why is it different if it is in a 
string (like snom) or another phone, with single tones?
If we understand this part, than is the question, where can I turn on 
the system to take a longer break between tones still as a string?


Back to the dialplan:
A Voip number can have different length of digits. Each number is seen 
as a complete picture, and so a three digit and a four digit number is 
something different. While in the legacy telephony the digits are worked 
down one by one and if there is no more use of the digits, they are just 
garbage and will be not used. Unlike in VoIP, where you can have a three 
digit number and if you dial four digit, than it is a WRONG number  
I just verified that: I dialed from 601 to  61522, however, 61522 does 
not exist, but 615 exists. Guess what? I get a busy tone! That should 
proof my thoughts (and that without yelling, ... hehehehe)


bye

Ronald
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Re: [asterisk-users] Blind transfer 3/4 digits

[Ronald Wiplinger]

David Gagnon wrote:

Ronald,

Like someone already told you, you should explain more clearly the
way you try to transfer, we need more details on the procedure, using which
button on which phone. We need every detail to help you. This as nothing to
do with the way the dial plan is loaded, this is totally false.

I'm sure most of the people here don't understand how you try to
transfer.

David

  


David,

I am not sure how the explanation how to punch the keys changes 
something,  ;-.)


Ok, here we go:

Snom:
pick up the phone and hit ##6014 followed by [ok]

Noname:
pick up the phone and hit ##6014 must be pushed very fast!!! No end 
# needed, since the phone 601 starts to ring as soon I reach 1.


In my opinion Asterisk remembers all numbers and therefore it does not 
wait for the 4, since it found a match. This is in VoIP (in my opinion) 
wrong, since overlapping numbers are allowed.


Sip message / dtmf, this is something different! How is the transfer 
made? Maybe snom does send a sip message, while the noname only send 
dtmf tones.


bye

Ronald


-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Ronald
Wiplinger
Envoyé : 4 septembre 2006 09:22
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [asterisk-users] Blind transfer 3/4 digits

Koopmann, Jan-Peter wrote:
  

On Sunday, September 03, 2006 3:40 AM Ronald Wiplinger wrote:

  


try that way. However, I have doubts as well. If you are right, than
why snom phone does not have this problem? Would not here also the
first match count?   

  

Because the transfer button on the SNOM is using a totally different


mechanism than sending # to Asterisk. On your snom configuration (like ours)
the phone does not start to create/send a SIP message until you hit OK. At
that time the entire number is there and a complete SIP transfer is created.
Cool down a bit. The problem you are having is most probably just a dialplan
problem. It takes some time and experience to get those things right. No
need to yell here...
  
  


What's happen to you guys? I am not yelling, just asking.
It is sure not a dialplan question! If it would be a dialplan question, 
than it would be for each dialing, but it isn't.


You mentioned SIP message and that makes me wonder! Are we not using 
here dtmf ?? that is in my opinion not a sip message, isn't it?
If it is a sequence of tones, than why is it different if it is in a 
string (like snom) or another phone, with single tones?
If we understand this part, than is the question, where can I turn on 
the system to take a longer break between tones still as a string?


Back to the dialplan:
A Voip number can have different length of digits. Each number is seen 
as a complete picture, and so a three digit and a four digit number is 
something different. While in the legacy telephony the digits are worked 
down one by one and if there is no more use of the digits, they are just 
garbage and will be not used. Unlike in VoIP, where you can have a three 
digit number and if you dial four digit, than it is a WRONG number  
I just verified that: I dialed from 601 to  61522, however, 61522 does 
not exist, but 615 exists. Guess what? I get a busy tone! That should 
proof my thoughts (and that without yelling, ... hehehehe)


bye

Ronald
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--
Ronald Wiplinger  (CEO of ELMIT)
http://www.elmit.com  http://voip.elmit.com  http://e-paper.elmit.com 
Tel. (M) +886.939.775.516  (O) +886.2.2835.7765 (ENUM)   or FWD 511208

- I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org

PS: Spam prevention!
Our system is protected with a spam prevention program. 
If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. 
After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again.


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Re: [asterisk-users] Blind transfer 3/4 digits

[Ronald Wiplinger]

wendell hamilton wrote:

Please excuse the top-posting.

  

... so we are faster at the solution, ... ;-)

In features.conf, uncomment transferdigittimeout and adjust its timing as 
desired.  You may also want to uncomment and adjust featuredigittimeout to a 
higher value as well.

That was it!!! Now it works!!!


  Also, since the dialplan does first match, you can eliminate the problem by 
putting the 4 digit extensions before the 3 digit extensions in the dialplan.

See the match as you go section at
http://www.voip-info.org/wiki/index.php?page=Asterisk+Extension+Matching

  


Thank you for the link, btw. your comment above does not match the 
link. Copy of the important part of your provided link:




  Example

FooBar Incorporated wants their incoming telephone calls to be 
answered with a voice message welcoming the caller and inviting them 
to choose which extension they want. FooBar has six telephone 
extensions. Their extension numbers are 1, 2, 21, 22, 31, 32. So this 
is the context created for incoming calls for FooBar Incorporated:


   [incoming]
   exten = s,1,Background(welcome-to-foobar-incorporated)
   exten = 1,1,Dial(Zap/1)
   exten = 2,1,Dial(Zap/2)
   exten = 21,1,Dial(Zap/3)
   exten = 22,1,Dial(Zap/4
   exten = 31,1,Dial(Zap/5)
   exten = 32,1,Dial(Zap/6)

When you call FooBar, Asterisk plays the 
welcome-to-foobar-incorporated.gsm sound file. After that, having 
run out of commands to execute, it waits for you to dial something. 
This is what Asterisk would do if you dialed various options:


   Number DialedAsterisk's Action
 1  Immediately performs Dial (Zap/1)
 2  Waits for timeout, then performs Dial(Zap/2)
21  Immediately performs Dial (Zap/3)
22  Immediately performs Dial (Zap/4)
 3  Waits for timeout, then hangs up.
31  Immediately performs Dial (Zap/5)
32  Immediately performs Dial (Zap/6)
 4  Immediately hangs up.

Note that when a caller tries to dial extension 2, they are not 
connected immediately. Asterisk waits to see if the caller dials more 
digits, to determine whether the caller wants extension 2 or 21 or 22. 
As callers would like to be connected immediately if possible, it 
would be more user-friendly to avoid using ambiguous extension numbers. 




Thanks for the solution, 

bye

Ronald


HTH

routerguy

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald Wiplinger
Sent: Monday, September 04, 2006 5:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Blind transfer 3/4 digits

David Gagnon wrote:
  

Ronald,

Like someone already told you, you should explain more clearly the
way you try to transfer, we need more details on the procedure, using which
button on which phone. We need every detail to help you. This as nothing to
do with the way the dial plan is loaded, this is totally false.

I'm sure most of the people here don't understand how you try to
transfer.

David

  



David,

I am not sure how the explanation how to punch the keys changes 
something,  ;-.)


Ok, here we go:

Snom:
pick up the phone and hit ##6014 followed by [ok]

Noname:
pick up the phone and hit ##6014 must be pushed very fast!!! No end 
# needed, since the phone 601 starts to ring as soon I reach 1.


In my opinion Asterisk remembers all numbers and therefore it does not 
wait for the 4, since it found a match. This is in VoIP (in my opinion) 
wrong, since overlapping numbers are allowed.


Sip message / dtmf, this is something different! How is the transfer 
made? Maybe snom does send a sip message, while the noname only send 
dtmf tones.


bye

Ronald



  

-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Ronald
Wiplinger
Envoyé : 4 septembre 2006 09:22
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [asterisk-users] Blind transfer 3/4 digits

Koopmann, Jan-Peter wrote:
  


On Sunday, September 03, 2006 3:40 AM Ronald Wiplinger wrote:

  

  

try that way. However, I have doubts as well. If you are right, than
why snom phone does not have this problem? Would not here also the
first match count?   

  


Because the transfer button on the SNOM is using a totally different

  

mechanism than sending # to Asterisk. On your snom configuration (like ours)
the phone does not start to create/send a SIP message until you hit OK. At
that time the entire number is there and a complete SIP transfer is created.
Cool down a bit. The problem you are having is most probably just a dialplan
problem. It takes some time and experience to get those things right. No
need to yell here...
  

  

  

What's happen to you guys? I am not yelling, just asking.
It is sure not a dialplan question! If it would be a dialplan question

Re: [asterisk-users] Blind transfer 3/4 digits

[Ronald Wiplinger]

Anthony Rodgers wrote:

With respect, the problem is with your numbering plan..



WHERE do you see a problem in the numbering plan?
I see the problem in ASTERISK, because it does not wait for the last 
digit!!!

Where can I set that it waits for it?

The beauty on voip IS that you can have different length and 
overlapping, 


bye

Ronald

CP

On 1-Sep-06, at 10:37 PM, Ronald Wiplinger wrote:


I found a problem in blind transfer:

I have an extension number 601 and I have an extension 6014 

If I get a call on 615 (snom) and transfer to 6014 it works, since snom
requires me to hit ok

If I get a call on 601 and transfer to 6014, than 601 will get the busy
signal and I hang up as usually with transfer.
Howerver the caller get the announcements: I could not get that, 

What could be the problem ?

bye

Ronald


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http://www.elmit.com  http://voip.elmit.com  http://e-paper.elmit.com 
Tel. (M) +886.939.775.516  (O) +886.2.2835.7765 (ENUM)   or FWD 511208

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After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again.


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Re: [asterisk-users] Blind transfer 3/4 digits

[Ronald Wiplinger]

David Gagnon wrote:

Ronald,

You seem to be a little bit angry about VoIP. If so, I could give
you my old Nortel system. Does this would make you happy?

David

  


David,

I am not angry about VoIP, but please send my your old Nortel system !

I just do not understand why I can DIAL 601 and 6014, but not use blind 
transfer. Is the question too difficult?


I am sure there is somewhere a switch to say, wait two seconds (as for 
dialing) before you assume it is a complete number.
It is also strange that snom phone can do it correct, because it uses 
the ok key.




-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Ronald
Wiplinger
Envoyé : 2 septembre 2006 04:20
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [asterisk-users] Blind transfer 3/4 digits

Anthony Rodgers wrote:
  

With respect, the problem is with your numbering plan..




  


This answer is therefore totally nonsense !!! (With all respect!!!)


Both answers have actually not lead to any step further, but to more 
messages. I use to refer to such answers as NON-ANSWERS.
Please only reply if and really only if you know a solution for the 
problem! Thanks for your understanding.


bye

Ronald - again, I am not angry at all.

WHERE do you see a problem in the numbering plan?
I see the problem in ASTERISK, because it does not wait for the last 
digit!!!

Where can I set that it waits for it?

The beauty on voip IS that you can have different length and 
overlapping, 


bye

Ronald
  

CP

On 1-Sep-06, at 10:37 PM, Ronald Wiplinger wrote:



I found a problem in blind transfer:

I have an extension number 601 and I have an extension 6014 

If I get a call on 615 (snom) and transfer to 6014 it works, since snom
requires me to hit ok

If I get a call on 601 and transfer to 6014, than 601 will get the busy
signal and I hang up as usually with transfer.
Howerver the caller get the announcements: I could not get that, 

What could be the problem ?

bye

Ronald


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--
Ronald Wiplinger  (CEO of ELMIT)
http://www.elmit.com  http://voip.elmit.com  http://e-paper.elmit.com 
Tel. (M) +886.939.775.516  (O) +886.2.2835.7765 (ENUM)   or FWD 511208

- I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org

PS: Spam prevention!
Our system is protected with a spam prevention program. 
If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. 
After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again.


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Re: [asterisk-users] Keys pressed not registering ...

[Ronald Wiplinger]

Lenny wrote:


Hello all,

For some reason when dialing in I get the IVR or if I forward to my 
conference line... any keys pressed seem like they aren’t received .. 
Like I’m pressing them, but they aren’t being registered with the 
server .. Any ideas?


I’m using the vmware nerdvittles build, the latest trixbox v1.1 .. 
FreePBX 2.1.1.


Everything else works just fine. I’m using VoIPDiscount for outgoing 
and Stana-in/Stanaphone to receive calls.


Any help is appreciated..

Have a look at the dtmfmode settings, inband, rfc2833, ... and try 
different settings.


bye

Ronald


Regards,

LB



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--
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Re: [asterisk-users] Keys pressed not registering ...

[Ronald Wiplinger]

Lenny wrote:

Hello Ronald ..

This is what I'm trying to learn of now ..

Where in freepbx do I place these settings?
  

sip.conf ;-)
that was easy, ... do you have another question?

bye

Ronald

Trunk settings?

If I could just get that bit of info..

Thanks

LB

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ronald
Wiplinger
Sent: Saturday, September 02, 2006 11:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Keys pressed not registering ...

Lenny wrote:
  

Hello all,

For some reason when dialing in I get the IVR or if I forward to my 
conference line... any keys pressed seem like they aren’t received .. 
Like I’m pressing them, but they aren’t being registered with the 
server .. Any ideas?


I’m using the vmware nerdvittles build, the latest trixbox v1.1 .. 
FreePBX 2.1.1.


Everything else works just fine. I’m using VoIPDiscount for outgoing 
and Stana-in/Stanaphone to receive calls.


Any help is appreciated..


Have a look at the dtmfmode settings, inband, rfc2833, ... and try 
different settings.


bye

Ronald
  

Regards,

LB



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Re: [asterisk-users] Blind transfer 3/4 digits

[Ronald Wiplinger]

Kevin Smith wrote:
Dialing a number and transferring a number are two different things. 
And no offense, you are not really providing a lot of details along 
with your problem. So you can dial the numbers but not transfer from 
one to the other.
I was not thinking that it would be too much difference. Therefore I 
also do not know what more info could help to distinguish the problem. I 
hardly can post my entire configuration.


What does the CLI say when you try the transfer? That would provide a 
lot of information that could clue you in to what is going on.


You hit another problem with that. I hardly see here anything anymore. 
The messages fly by so fast,  Especially annoying messages:
chan_sip.c:10888 handle_request_register: Registration from 
'sip:192.168.250.20' failed for '192.168.250.244' - Username/auth name 
mismatch

-- Got SIP response 486 Busy Here back from 192.168.250.244
-- Got SIP response 400 Bad Request back from xx.xx.xx.126
NOTICE[5936]: chan_sip.c:9600 handle_response_register: Failed to 
authenticate on REGISTER to '[EMAIL PROTECTED]' (Tries 3)

.

It would be nice to filter the CLI for such investigation for a moment.
What type of phones are you using? Some phones have the ability to 
pattern match and wait for a certain number of seconds before sending 
the number to asterisk. For example. On our Polycom phones a user has 
3 seconds (between digits) to enter in 10 digits. This could be where 
most of your problem is.
That is a very good point and I will contact the manufacturer of these 
no-name phones.


My guess the problem lies with the Phones, not Asterisk form the 
information you provided.
I disagree with that! Why Asterisk treats dialing and transfer 
different. That makes not really sense, does it?


bye

Ronald


Kevin


Ronald Wiplinger wrote:

David Gagnon wrote:

Ronald,

You seem to be a little bit angry about VoIP. If so, I could give
you my old Nortel system. Does this would make you happy?

David

  


David,

I am not angry about VoIP, but please send my your old Nortel system 
!


I just do not understand why I can DIAL 601 and 6014, but not use 
blind transfer. Is the question too difficult?


I am sure there is somewhere a switch to say, wait two seconds (as 
for dialing) before you assume it is a complete number.
It is also strange that snom phone can do it correct, because it uses 
the ok key.




-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Ronald
Wiplinger
Envoyé : 2 septembre 2006 04:20
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [asterisk-users] Blind transfer 3/4 digits

Anthony Rodgers wrote:
 

With respect, the problem is with your numbering plan..




  


This answer is therefore totally nonsense !!! (With all respect!!!)


Both answers have actually not lead to any step further, but to more 
messages. I use to refer to such answers as NON-ANSWERS.
Please only reply if and really only if you know a solution for the 
problem! Thanks for your understanding.


bye

Ronald - again, I am not angry at all.

WHERE do you see a problem in the numbering plan?
I see the problem in ASTERISK, because it does not wait for the last 
digit!!!

Where can I set that it waits for it?

The beauty on voip IS that you can have different length and 
overlapping, 


bye

Ronald
 

CP

On 1-Sep-06, at 10:37 PM, Ronald Wiplinger wrote:

  

I found a problem in blind transfer:

I have an extension number 601 and I have an extension 6014 

If I get a call on 615 (snom) and transfer to 6014 it works, since 
snom

requires me to hit ok

If I get a call on 601 and transfer to 6014, than 601 will get the 
busy

signal and I hang up as usually with transfer.
Howerver the caller get the announcements: I could not get that, 

What could be the problem ?

bye

Ronald




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Re: [asterisk-users] Blind transfer 3/4 digits

[Ronald Wiplinger]

Tim St. Pierre wrote:
Are you using # to transfer?  If so, it's not sending it as a new call, it's 
just sending asterisk digits using whatever DTMF mode.  Asterisk parses these 
based on a first match in the dialplan.  Make sure that the longer 
extension numbers are loaded first in the dialplan.


  


That is a good thought. I can remember that the docs said that you 
cannot force the order of the dialplan, except with includes. I will try 
that way.
However, I have doubts as well. If you are right, than why snom phone 
does not have this problem? Would not here also the first match count?


bye

Ronald

-Tim

On September 2, 2006 20:12, Ronald Wiplinger wrote:
  

Kevin Smith wrote:


Dialing a number and transferring a number are two different things.
And no offense, you are not really providing a lot of details along
with your problem. So you can dial the numbers but not transfer from
one to the other.
  

I was not thinking that it would be too much difference. Therefore I
also do not know what more info could help to distinguish the problem. I
hardly can post my entire configuration.



What does the CLI say when you try the transfer? That would provide a
lot of information that could clue you in to what is going on.
  

You hit another problem with that. I hardly see here anything anymore.
The messages fly by so fast,  Especially annoying messages:
 chan_sip.c:10888 handle_request_register: Registration from
'sip:192.168.250.20' failed for '192.168.250.244' - Username/auth name
mismatch
 -- Got SIP response 486 Busy Here back from 192.168.250.244
 -- Got SIP response 400 Bad Request back from xx.xx.xx.126
NOTICE[5936]: chan_sip.c:9600 handle_response_register: Failed to
authenticate on REGISTER to '[EMAIL PROTECTED]' (Tries 3)
.

It would be nice to filter the CLI for such investigation for a moment.



What type of phones are you using? Some phones have the ability to
pattern match and wait for a certain number of seconds before sending
the number to asterisk. For example. On our Polycom phones a user has
3 seconds (between digits) to enter in 10 digits. This could be where
most of your problem is.
  

That is a very good point and I will contact the manufacturer of these
no-name phones.



My guess the problem lies with the Phones, not Asterisk form the
information you provided.
  

I disagree with that! Why Asterisk treats dialing and transfer
different. That makes not really sense, does it?

bye

Ronald



Kevin

Ronald Wiplinger wrote:
  

David Gagnon wrote:


Ronald,

You seem to be a little bit angry about VoIP. If so, I could give
you my old Nortel system. Does this would make you happy?

David
  

David,

I am not angry about VoIP, but please send my your old Nortel system
!

I just do not understand why I can DIAL 601 and 6014, but not use
blind transfer. Is the question too difficult?

I am sure there is somewhere a switch to say, wait two seconds (as
for dialing) before you assume it is a complete number.
It is also strange that snom phone can do it correct, because it uses
the ok key.



-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Ronald
Wiplinger
Envoyé : 2 septembre 2006 04:20
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [asterisk-users] Blind transfer 3/4 digits

Anthony Rodgers wrote:
  

With respect, the problem is with your numbering plan..


This answer is therefore totally nonsense !!! (With all respect!!!)


Both answers have actually not lead to any step further, but to more
messages. I use to refer to such answers as NON-ANSWERS.
Please only reply if and really only if you know a solution for the
problem! Thanks for your understanding.

bye

Ronald - again, I am not angry at all.



WHERE do you see a problem in the numbering plan?
I see the problem in ASTERISK, because it does not wait for the last
digit!!!
Where can I set that it waits for it?

The beauty on voip IS that you can have different length and
overlapping, 

bye

Ronald

  

CP

On 1-Sep-06, at 10:37 PM, Ronald Wiplinger wrote:


I found a problem in blind transfer:

I have an extension number 601 and I have an extension 6014 

If I get a call on 615 (snom) and transfer to 6014 it works, since
snom
requires me to hit ok

If I get a call on 601 and transfer to 6014, than 601 will get the
busy
signal and I hang up as usually with transfer.
Howerver the caller get the announcements: I could not get that, 

What could be the problem ?

bye

  


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[asterisk-users] Blind transfer 3/4 digits

[Ronald Wiplinger]

I found a problem in blind transfer:

I have an extension number 601 and I have an extension 6014 

If I get a call on 615 (snom) and transfer to 6014 it works, since snom 
requires me to hit ok


If I get a call on 601 and transfer to 6014, than 601 will get the busy 
signal and I hang up as usually with transfer.

Howerver the caller get the announcements: I could not get that, 

What could be the problem ?

bye

Ronald


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[asterisk-users] GIZMO and Asterisk, Failed to authenticate

[Ronald Wiplinger]

[Aug 31 04:32:22] NOTICE[20241]: chan_sip.c:5291 sip_reg_timeout:-- 
Registration for '[EMAIL PROTECTED]' timed out, trying 
again (Attempt #984)
[Aug 31 04:32:23] NOTICE[20241]: chan_sip.c:9600 
handle_response_register: Failed to authenticate on REGISTER to 
'[EMAIL PROTECTED]' (Tries 3)



sip.conf:

register = 1747mynumber:[EMAIL PROTECTED]   ; Gizmoproject

[proxy01.sipphone.com]
type=friend
context=default
disallow=all
allow=ulaw
allow=alaw
allow=ilbc
dtmfmode=rfc2833
host=proxy01.sipphone.com
insecure=very
secret=mypassword
username=1747mynumber
canreinvite=yes


What did I wrong?

bye

Ronald

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[asterisk-users] Wellgate 3804a: Got SIP response 486 Busy Here

[Ronald Wiplinger]

I cannot explain why I get all the time:

Got SIP response 486 Busy Here back from 192.168.250.244

I have a Wellgate 3804a there.

How can I solve it?

bye

Ronald
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[asterisk-users] Am I looking for automon?

[Ronald Wiplinger]

I want to record a call, either it is an incoming call or an outgoing call.

I have in features.conf:

automon = *1


However, I am not sure if that is what I need, and how to use it.

bye

Ronald
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