date:20040923

[Asterisk-Users] RE: RE: Creating conference calls from within Astman.

2004-09-23 Thread Shad Mortazavi

Title: RE: RE: Creating conference calls from within Astman.





Sorry about my earlier e-mail. (:blush:), should have included the history.


Can this be done from within Gastman?


Warm Regards


Shad


--


Message: 2
Date: Wed, 22 Sep 2004 21:10:10 -0300
From: Nicol?s Gudi?o [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] RE: Creating conference calls from
 within Astman.
To: Asterisk Users Mailing List - Non-Commercial Discussion
 [EMAIL PROTECTED]
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859-1


Hello,


-Original Message-
From: Shad Mortazavi
Sent: Friday, September 17, 2004 1:03 PM
To: [EMAIL PROTECTED]
Subject: Creating conference calls from within Astman. 

Dear All,

I have a requirement to 'originate' a number of calls to various
external users from within a conference room, so that the end users does not pay for the call.

I know that within Astman I can define an extension and then
originate the call from that extension. Can I define a conference room (how would I configure that on astman? What channel would it

use?) and then generate a number of calls from within the conference
room?


I was incorporating this functionality into the Flash Operator Panel.
In the panel I do it by using the Local channel. Suppose that you want to call the number 555-, and your context for dialing out is 'dialout'. You also have another context 'conferences' with extension number 1000 that fires up a meetme.

The originate command for astman should look like:


Action: Originate
Channel: Local/[EMAIL PROTECTED]
Exten: 1000
Context: conferences
Priority: 1


It will dial the number 555 as if it were dialed from the dialout context and connect the call to Extension 1000 in 'conferences'

context.


This feature is already implemented and working in the next to be released version of the flash panel (but now it will only dial numbers predefined in the panel itself). You can get it from http://www.asternic.org

Best regards,



--
Nicolás Gudiño
Buenos Aires - Argentina



--



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[Asterisk-Users] Re: [Asterisk-Dev] Softphone for PocketPC or iPaq

2004-09-23 Thread Peter Svensson

On 22 Sep 2004, Sudhir Kumar wrote:

 Is there a soft phone for PocketPC or iPaq? If not, is someone working
 on it? I will be more than willing to contribute my mite if needed.

Xten has a product, possibly still in beta. 

Peter


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[Asterisk-Users] RE: An old problem still hanging around?

2004-09-23 Thread Bill Seddon

Having just run the command sip show channels I get a list of channels
even though there is no one on the phone (we only have 4 so it's easy to
tell).

Here is what I get:

Peer User/ANRCall ID  Seq (Tx/Rx)   Format
192.168.0.22 (None)  4c81ac8e90c  00101/0   UNKN  
192.168.0.22 (None)  984ee48048d  00101/0   UNKN  
192.168.0.22 (None)  200d9d37123  00101/0   UNKN

Is this normal?  Why just one phone (a Grandstream Handytone ATA)?

Running sip show channel 984ee48048d I get the output below so it seems
active:

* SIP Call
  Direction:  Incoming
  Call-ID:[EMAIL PROTECTED]
  Our Codec Capability:   524302
  Non-Codec Capability:   1
  Their Codec Capability:   0
  Joint Codec Capability:   0
  Format  UNKN
  Theoretical Address:192.168.0.22:5060
  Received Address:   192.168.0.22:5060
  NAT Support:RFC3581
  Our Tag:1190462248
  Their Tag:  
  SIP User agent: 
  Need Destroy:   0
  Last Message:   
  Promiscuous Redir:  No
  Route:  N/A
  DTMF Mode:  rfc283

Here's a quote from a post to an earlier question by someone seeing a
similar list in Jun/Jul this year.

QUOTE
This behavior was observed by several people for a short period of time 
and then seemed to have disappeared with a cvs versions starting around 
1.390 - 1.394 (chan_sip.c) according to my  observations (more like a 
guess actually) couldn't exactly pinpoint the patch that stopped it.
/QUOTE

My version of asterisk is from HEAD on 2004-09-19.  Should I be concerned?

Thanks 

Bill Seddon


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[Asterisk-Users] Asterisk and Siemens HiPath

2004-09-23 Thread syscon-lists

Good Morning !
Has somebody successfully connected from ASTERISK with the chan_h323 module 
to a Siemens HiPATH RG2500 Gatekeeper and can use it for calls?
By searching the Internet I just found several H.323 connection issues with 
different gatekeepers and gateways, but none with a Siemens VoIP system ...

Would like to bridge SIP and H.323 phones with it ...
Thanks,
Juergen
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Re: [Asterisk-Users] Problems compiling CAPI

2004-09-23 Thread igil








[EMAIL PROTECTED] (Thomas
Niesel) 
Enviado por: [EMAIL PROTECTED]
22/09/2004 19:58



Por favor, responda a
Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED]





Para
[EMAIL PROTECTED]


cc



Asunto
Re: [Asterisk-Users] Problems compiling
CAPI








On Wed, Sep 22, 2004 at 07:15:19PM +0200, [EMAIL PROTECTED]
wrote:
 Hello all,

Sorry for my first mail which answers the 2nd part:(

 
 I'm trying to setup a AVM C2 card.
 
 I have read the kernel requirements for this card.
 
 
 M CAPI2.0 support 
 [*] Verbose reason code reporting (Kernel size +=7K) 
  
^^^


   

  No need for that!
 [*] CAPI2.0 Middleware support (EXPERIMENTAL) 
 M CAPI2.0 /dev/capi support 
 [*] CAPI2.0 filesystem support 
 M CAPI2.0 capidrv interface support 
 
 My problem is when I make a make menuconfig in /usr/src/linux/,
i can't 
 see 
 any reference to CAPI2.0 filesystem support, in my kernel, 2.4.18
(Debian 
 stable).
make menuconfig is just to configure
How about make modules  make modules_install

Make menuconfig don't show me the CAPI
Filesystem option to select for compiling.
Then I do not compile because this option
not appear.

Ismael Gil

-- 
Tho/\/\as
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[Asterisk-Users] Cisco 30 VIP

2004-09-23 Thread Christoph Kampka

Hi there,
I know this was discussed many times already, but after trying to get my
30VIP to work for a week, I think I might be doing something wrong.
I'm trying to set it up with chan_skinny. Haven't tried chan_sccp yet, but
since there are many of you already doing it with skinny, it should be
working also in my case. I read and used all the hints and samples I was
able to find on the list and elsewhere. Source is new, from yesterday.

This is in my skinny.conf (apart from all the default stuff)

[30vip]
device=SEP00B06409B25D
version=P002F202
context=default
line = 100

I tried also other parameters, to no avail.
Display shows F2.02

Phone settings:
IP 192.168.1.3
Mask 255.255.255.0
Gateway 192.168.1.9 (* box)
DNS 0.0.0.0 (disabled)
TFTP 192.168.1.9 (* box)

Any suggestions?
4

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[Asterisk-Users] Eicon Diva PCI 2.02 (not server)

2004-09-23 Thread mark

Hi.

I've been trying out Asterisk and it
looks good. I've have got the VoiP stuff working with some softphones
and I'm very impressed. My next step was to interface it with my
phone system here to try it out that way. I have an Eicon Diva PCI
2.02 (not server) lying around which I have put into my asterisk box and
I've tried to configure it but I've had no luck so far. SuSE 9.1
detects it but I can't get chan_capi to work with it (and I suspect that
it's not possible with this card). Is it possible to use mISDN with
this instead to get it working with asterisk? If so how? :D

Thanks

Mark
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