Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous registry

2004-07-10 Thread R. Anton Raharja
in other SIP proxy server, this can be done easily, i mean its default
1 or more phone could be registered at 1 number (12345) and resulting same effect as u 
ask
SER (SIP Express Router, http://iptel.org/ser) can deal with this
SER is a friend to asterisk, i think :), you can accept calls with SER and pass it to 
asterisk to process complex dialplan
but if this feature implemented in asterisk alone, it would be nice

*** REPLY SEPARATOR  ***

On 11/07/2004 at 6:00 Kannaiyan Natesan wrote:

Paul,

The question is very simple.

When I call a SIP user, the phone should ring in more than one
extentions. Also more than one phone should be able to register with
asterisk. Right now it is not the case. The last phone which register will
be receiving the calls. This type of situations might be needed in call
centres.


Called 12345
|---(12345) Ringing
|---(12345) Ringing
|---(12345) Ringing

So you don't need to disturb asterisk when you add more devices to it to
receive calls.
Such facility is not available in asterisk at this moment.

I hope this helps.
Since I feel this is a great feature, I will topup up to $100/-


-.Kannaiyan

http://www.goods2world.com -- Your Only VoIP


- Original Message -
From: Paul Mahler [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, July 11, 2004 5:44 AM
Subject: RE: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
registry


 I'm not sure I understand what you are trying to do.

 You have an administrative assistant and several other staff. You want
the
 administrator to be able to take calls directed to the staff extensions?

 If I have the requirement right, you could accomplish this by ringing the
 staff extension and the admin extension at the same time. The Dial
command
 allows you to ring multiple extensions simultaneously.

 If you want to be able to more easily recognize what extension the
traffic
 if for, you can add additional extensions to the 7960. For example, if
you
 have two staff the admin monitors, add two additional extensions to the
 7960. The admin can tell who is being called by the extension that rings.

 Paul


 Paul Mahler
 [EMAIL PROTECTED]
 Signate, LLC
 665 Third Street
 Suite 100
 San Francisco, CA
  94107-1901

  Asterisk Services and Training









  -Original Message-
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] On Behalf Of
  Daniel Jimenez
  Sent: Saturday, July 10, 2004 3:05 PM
  To: [EMAIL PROTECTED]
  Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP
  simultaneous registry
 
  http://voip-info.org/tiki-index.php?page=Asterisk+bounty+SIP+s
  imultaneous+registry
 
  Updated,
 
  Allow a SIP device to register more than once so a single
  extension may exist in multiple locations.
 
  Upped total to $75.
 
  Daniel...
 
  Daniel Jimenez wrote:
  
  http://voip-info.org/tiki-index.php?page=Asterisk+bounty+SIP+simultane
   ous+registry
  
  
  
From the WIKI:
  
   Contributions
   Manager: Daniel Jimenez (cuban)
   Bounty: $50 USD
   Date opened: July 10, 2004
   Contributors: cuban ($50)
  
   Detail
  
   Yes, Yes I know you could do all sorts of fun with the dialplan to
   produce a similar effect, but I still would like to be able
  to do this.
   Plus it's easy money :).
  
   I have some users with a 7960 who are administrative assistants who
   monitor calls for 3 or 4 other people. It'd be nice to have
  two line
   instances for them, and one for the person(s) whom they assist.
  
   Contact me: djimenez at pobox.com if you're interested in
  making this
   happen.
  
 
  --
  Daniel Jimenez djimenez[at]pobox[dot]com
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http://sleepless.ngoprek.org
VoIP Rakyat: (0921) 20006


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[Asterisk-Users] xlite calls not approved

2004-07-09 Thread R. Anton Raharja
asterisk 0.9.1 with regular sip.conf and extensions.conf
sjPhone able to register and make calls
xlite said logged in but when i start to call/dial it said calls not approved
n i dont see anything while my asterisk sip debug enabled

can anyone give me a clue whats happening?



http://sleepless.ngoprek.org
VoIP Rakyat: (0921) 20006


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RE: [Asterisk-Users] xlite calls not approved

2004-07-09 Thread R. Anton Raharja
ok, this is my sip.conf
xlite cant calls, sjPhone can
i wish sjPhone dont hav that popup thing :)

[general]
port = 5060
bindaddr = 0.0.0.0
context = intern
tos=lowdelay
videosupport=yes
disallow=all
allow=gsm
allow=ulaw
allow=alaw

register = sleepless:pwd:[EMAIL PROTECTED]

[voiprakyat.net]
type=peer
context=intern
username=sleepless
secret=pwd
host=voiprakyat.net
nat=yes
canreinvite=no

[1234]
type=friend
context=intern
username=1234
secret=pwd
host=dynamic
nat=yes
canreinvite=yes

[5678]
type=friend
context=intern
username=5678
secret=pwd
host=dynamic
nat=yes
canreinvite=yes

*** REPLY SEPARATOR  ***

On 09/07/2004 at 14:29 Jay Milk wrote:

Show us your sip.conf -- probably a config issue

 -Original Message-
 From: R. Anton Raharja [mailto:[EMAIL PROTECTED] 
 Sent: Friday, July 09, 2004 1:16 PM
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] xlite calls not approved
 
 
 asterisk 0.9.1 with regular sip.conf and extensions.conf 
 sjPhone able to register and make calls xlite said logged 
 in but when i start to call/dial it said calls not 
 approved n i dont see anything while my asterisk sip debug enabled
 
 can anyone give me a clue whats happening?


http://sleepless.ngoprek.org
VoIP Rakyat: (0921) 20006


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RE: [Asterisk-Users] xlite calls not approved

2004-07-09 Thread R. Anton Raharja
now tht i guess your dtmf problem fixed too,
mind to tell us (or me) wht u've done to fix xlite call not approved problem?

*** REPLY SEPARATOR  ***

On 09/07/2004 at 13:15 CHS wrote:

ok, I've finally got it working. I can get to the demo extension '1000'
and I hear the voice, etc..

only one problem, I can't seem to hit any of the demo extensions (like 2
for more detailed info, etc..)


http://sleepless.ngoprek.org
VoIP Rakyat: (0921) 20006


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[Asterisk-Users] voicetronix n asterisk

2004-06-22 Thread R. Anton Raharja

im sure this is the perfect place to ask bout asterisk

http://www.voicetronix.com/openpbx.htm + asterisk
is this a good solution for VoIP on private network (6 office, each has their own 
existing PBX)

good means relatively cheap, stable n reliable

thx


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