date:20060207

Re: [Asterisk-Users] virtual extension per user ?

2006-02-07 Thread Rajeev Natarajan

If the users have a bluetooth device like a cellphone-with-bluetooth or 
their laptop, this might work: http://mundy.org/blog/index.php?p=78 - 
you'll have to modify the script in the tutorial a bit.


essentially - you have a presence server at the two offices - when they 
enter the building, the bluetooth device registers with the presence 
server and the corresponding phone comes alive.


works great for us (as long as the bloke doesn't leave the cellphone at 
home)


rajeev

--
Chief Technology Officer
Gyantec Consulting (I) Pvt. Ltd.
Chennai, INDIA
Phone: +91-44-4205-4446
Mob  : +91-944-407-2925
Fax  : +91-44-4205-4546
VoIP : +1-360-519-5969


Alex Ongena wrote:

Hi,

People here often work on 2-3 places (office 1, office 2 and home).

I would like to give them 1 extension (XXX) and to ask them to
'register' the phone they use at a certain moment.

The idea is that, when you need someone, just dial XXX and the
phone near him (in Office 1, Office 2 or at Home), will ring.
This will keep my queue system and other tricks intact, where I
always use the single extension XXX.

I know you can 'forward' calls to other extensions, but when people
go from Office 1 to Office 2, they forget to enable their forward in
Office 1 to Office 2.
I like a solution where they can say 'Please register me, I'am now
sitting in Office 2'. The moment after 'registration', when you call
XXX, the phone in Office 2 will ring.

In all places I use Asterisk 1.2.1 with bristuff, Cisco 7940/60 phones
with Sip and some Sip softphones.

Any hints or tricks to get this behaviour ?

Thanks
Alex
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Re: [Asterisk-Users] bug in bristuff?

2006-02-07 Thread stoffell

On 2/6/06, Conrad Wood <[EMAIL PROTECTED]> wrote:
> Please note the spelling of uniqueid. I find the spelling in
> res_features.c - but only once I patched it with bristuff patches.
> Does anyone know whether that is a known problem with bristuff? If so is
> it fixed in a later version?

What version of bristuff are you using? Then I can have a look in my
bristuff to see if I have the same problem..

> Where do I report a bug in bristuff? ;)

Check this website to contact the author of bristuff, http://www.junghanns.net

cheers
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[Asterisk-Users] Re: MeetMe - Party's are not exchanging Audio - Is this BUG?

2006-02-07 Thread Somesh S Shanbhag

Hi All, Please help me solving this problem. Thanks Somesh S. ShanbhagSomesh S Shanbhag <[EMAIL PROTECTED]> wrote: Hi All, I observed the following in my try towards Multiparty Conferencing. I am establishing the Multiparty Conferencing through Asterisk Manager API. I have two users SIP/111 and SIP/101 of which SIP/101 is treated as leader. Following commands are used - Action: Originate Channel: SIP/111 Application: MeetMe Data: |edwx ActionID: ffe4563 When I use the above, Incoming call will be generated to SIP/111 and when it accepts the call, a message shall be played from asterisk which is like - " You are entering conference number 0 and conference will begin as soon as leader arrives". This is fine. Now I shall give ano
ther
command - Action: Originate Channel: SIP/101 Application: MeetMe Data: 0|aEp ActionID: ffe4563 As soon as I do the above, Incoming call is generated to SIP/101 (leader) and when he accepts the call it plays the message - "You are joining the conference number 0". This is fine. But now, when SIP/111 talks SIP/101 (leader) is able to hear. But when SIP/101(leader) talks, SIP/111 is not able to hear anything... Is this a BUG in MeetMe? Please clarify the same. I am using asterisk-1.2.0 and zaptel - ztdummy are installed. Regards, Somesh S. Shanbhag Bring words and photos together (easily) with PhotoMail - it's free and works with your Ya
hoo!
Mail.

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