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We have not done a full regression test with 1.6.2 but we have been using 1.6.2 in our 1.6.x test enviroment at at least superficially it appears to work fine with 1.6.2.10.

We will be done a full regression test with the 3.5 beta in the next few weeks.

Does anyone know which versions of asterisk are supported by Noojee Receptionist now?  I know 1.6.2.X was not supported for a while because of a bug with the manager interface, but I didn't know if asterisk had been fixed yet.

do you see any errors in the error console?

After you made the changes manually does the config dialog display the manually entered values.

Dear Brett,

Thanks for the answer!

OS: Windows XP SP3
Firefox: 3.6.6

--> I tried editing the value of extensions.noojeeclick.dialPrefix manually in the about:config page, and that works.

strange it doesn't work in GIU.

You can't do it at the moment, but I have started working on code to support a 'callto' url.

At best guess it will be several weeks before I release it.
I will post an announcement here once its done.

OK. I will add Thompson to the list of support phones over the next few weeks.

In the meantime try selecting polycom as the handset type.

is this something that is available?

I need to provide a Noojee button for the back-office page where the phone number is displayed inside "input" type of elements. I'd love to use something similar to what Skype has "a href=skype:xxxx?call"

Is there a way I can generate a link in HTML to use Noojee call button?

Yes that does look like a problem with your handset/asterisk server.
Not really me area of expertise but it looks like you might have registration problems.
If you want help with sort of problem it would have to go through our paid support.

As to your question about handset type.
Basically in order to implement Auto Answer Noojee Click needs to send a SIP header to the handset that instructs the handset to auto answer the call. There is some variation on the exact format of this header and so we need to know the handset type.
In reality most handsets use the same header as the Snom, so if your handset is listed always try the Snom first.

In some cases such as the Polycom you also need to configure the handset to allow auto-answer (the header isn't enough). Most phones will however take an auto-answer header without requiring any configuration.

FYI: last time I looked the Polycom auto answer setting can only be configured vai a provisioning service such as Noojee Provision or similar.

Well to automatically pickup our phones, we use the following syntax in our asterisk:

SIPAddHeader(Call-Info:<sip:192.168.10.20>\;answer-after=0)

Yes that does look like a problem with your handset/asterisk server.
Not really me area of expertise but it looks like you might have registration problems.
If you want help with sort of problem it would have to go through our paid support.

As to your question about handset type.
Basically in order to implement Auto Answer Noojee Click needs to send a SIP header to the handset that instructs the handset to auto answer the call. There is some variation on the exact format of this header and so we need to know the handset type.
In reality most handsets use the same header as the Snom, so if your handset is listed always try the Snom first.

In some cases such as the Polycom you also need to configure the handset to allow auto-answer (the header isn't enough). Most phones will however take an auto-answer header without requiring any configuration.

FYI: last time I looked the Polycom auto answer setting can only be configured vai a provisioning service such as Noojee Provision or similar.

OK, maybe not a couple of days.
Try the attached 1.6.2 in which I've changed the case.

I think it has corrected the CallerID/CALLERID case problem. I write 'I think' because I now have a problem with my thomson ST2030 which doesn't want to ring. But I don't think it's a problem with NoojeeClick, as the problem occurs between my asterisk and my phone (why do you ask for a phone type in the configuration then ?!)

On my Asterisk, I have this message:

  == HTTP Manager 'admin' logged on from 192.168.100.99
  == Using SIP RTP CoS mark 5
    -- Got SIP response 480 "Temporarily Unavailable" back from 192.168.100.202

192.168.100.202 is my phone's ip.


And in my phone logs, I have:

Recv from udp: 192.168.100.27:5060 00:00:00:04:680 (858 bytes)

INVITE sip:1001@192.168.100.202:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.100.27:5060;branch=z9hG4bK79cc226d;rport
Max-Forwards: 70
From: "1003-NoojeeClick" <sip:1001@192.168.100.27>;tag=as5adfaeeb
To: <sip:1001@192.168.100.202:5060;user=phone>
Contact: <sip:1001@192.168.100.27>
Call-ID: 2b3be02f4ec5d0bd03a88491305cc1ca@192.168.100.27
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.9
Date: Tue, 13 Jul 2010 10:44:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 339805235 339805235 IN IP4 192.168.100.27
s=Asterisk PBX 1.6.2.9
c=IN IP4 192.168.100.27
t=0 0
m=audio 18162 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


Sent to udp: 192.168.100.27:5060 00:00:00:04:706 (336 bytes)

SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/UDP 192.168.100.27:5060;branch=z9hG4bK79cc226d;rport
From: "1003-NoojeeClick"<sip:1001@192.168.100.27>;tag=as5adfaeeb
To: <sip:1001@192.168.100.202:5060;user=phone>;tag=c0a80101-c2d90dd
Call-ID: 2b3be02f4ec5d0bd03a88491305cc1ca@192.168.100.27
CSeq: 102 INVITE
Content-Length: 0



Recv from udp: 192.168.100.27:5060 00:00:00:04:712 (439 bytes)

ACK sip:1001@192.168.100.202:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.100.27:5060;branch=z9hG4bK79cc226d;rport
Max-Forwards: 70
From: "1003-NoojeeClick" <sip:1001@192.168.100.27>;tag=as5adfaeeb
To: <sip:1001@192.168.100.202:5060;user=phone>;tag=c0a80101-c2d90dd
Contact: <sip:1001@192.168.100.27>
Call-ID: 2b3be02f4ec5d0bd03a88491305cc1ca@192.168.100.27
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.9
Content-Length: 0