[Asterisk-Users] Re: Answering a queue with an SIP UA, then transfer to another sip UA oneway audio

2005-06-07 Thread fredrik chabot
fredrik chabot wrote: The problem is as follows I've made a queue and i queue incoming calls in that queue. The reception log's in as an agent to that queue and gets the calls for that queue so far so good. Now I need to transfer the call. I press flash (all granstream bt101

Re: [Asterisk-Users] Bluetooth with *

2005-06-06 Thread fredrik chabot
Theo P. Zourzouvillys wrote: On Saturday 04 December 2004 04:43, Nate Carlson wrote: In other words, if it's something you really want, add more cash to the bounty, to help encourage the developer to spend more time on it *grin*: alright, alright - i'll work on it today

[Asterisk-Users] Answering a queue with an SIP UA, then transfer to another sip UA oneway audio

2005-06-06 Thread fredrik chabot
The problem is as follows I've made a queue and i queue incoming calls in that queue. The reception log's in as an agent to that queue and gets the calls for that queue so far so good. Now I need to transfer the call. I press flash (all granstream bt101 phones) extension, announce the call

Re: [Asterisk-Users] bluetooth headset/handsfree

2005-06-02 Thread Fredrik Chabot
Laurent Lesage wrote: thanks for the aswers but I forgot to say that I would like it work with Linux. I think all of you use it with Winxx? And DIAX works just for Winxx? What would be the best, that's to use a BT fixed on the Asterisk server, so that you do not have to use another computer

Re: [Asterisk-Users] Rhino Channel Bank

2003-09-10 Thread fredrik chabot
George Pajari wrote: FYI I asked them: Your website talks about configuring the Rhino channel bank as 24xFXS. Is it possible to mix FXO and FXS modules? What affect does that have on pricing? They replied: We will have FXO and the ability to mix both FXS FXO within 60-90 days. Our RD

[Asterisk-Users] Dialplan question

2003-09-06 Thread fredrik chabot
can get out, most of the time however I get a busy signal halfway throu the number. It works more often if I change Early Dial: No Yes (use "Yes" only if proxy supports 484 response) to No. In the Budgetone 100 phone. regards, fredrik chabot --- *CLI show dialplan [ Contex

[Asterisk-Users] NOTICE[98311]: File chan_sip.c, Line 5070 (handle_request): Registration from 'sip:100@192.168.123.2' failed for '192.168.123.110'

2003-09-06 Thread fredrik chabot
Hello, Is there any way to get rid of this message. NOTICE[98311]: File chan_sip.c, Line 5070 (handle_request): Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.123.110' There where some pointer earlier in this list like avoiding dynamic ip's etc. And right after changing that

Re: [Asterisk-Users] NOTICE[98311]: File chan_sip.c, Line 5070 (handle_request): Registration from 'sip:100@192.168.123.2' failed for '192.168.123.110'

2003-09-06 Thread fredrik chabot
end secret= username=101 host=192.168.123.106 dtmfmode=inband ; Choices are inband, rfc2833, or info mailbox=1234,2345 ; Mailbox for message waiting indicator [EMAIL PROTECTED] asterisk]# -- from sip.conf Martin On Sat, 6 Sep 2003, fredrik chabot wrote: Hello, Is there any way