Dial failed: Originate failed

[lstep]

Hello, I get the following error message, when I try to dial a  number:

Dial failed: Originate failed

When I look at my Asterisk (1.6+), I have the following error message :

[Jul  1 16:49:35] ERROR[5488]: pbx.c:2857 ast_func_write: Function CallerID not registered
[Jul  1 16:49:35] ERROR[5488]: pbx.c:2857 ast_func_write: Function CallerID not registered
    -- Got SIP response 480 "Temporarily Unavailable" back from 192.168.100.202
       > Channel SIP/1001-00b6cb88 was never answered.

192.168.100.202 is the ip of my phone. It's a Thomson ST2030. It doesn't seem to be supported in your interface (I tried all the phone brands in the list, I get the same result). Is it the reason Noojee click doesn't work?

Thanks for your help.

[Brett Sutton]

Istep,
I don't think your problem has anything to do with using the thompson.

The error looks like a configuration problem on your asterisk server.

Precisely what version of asterisk are you running?
Are you running a particular distribution?
If so what one and what version?

Some quick googling suggest that the thompson doesn't support auto answer.

[lstep]

Istep,
I don't think your problem has anything to do with using the thompson.

The error looks like a configuration problem on your asterisk server.

Precisely what version of asterisk are you running?
Are you running a particular distribution?
If so what one and what version?

I'm running Asterisk version 1.6.1-rc1, on a Debian Lenny, the asterisk is not the official one from Debian, I compiled it myself.

My manager.conf:

Code:

[superuser]
secret=XXXXXXXXXXXX
deny=0.0.0.0/0.0.0.0
permit=192.168.1.0/255.255.255.0
permit=127.0.0.1/255.255.255.0
read = system,call,log,verbose,command,agent,user,all
write = system,call,log,verbose,command,agent,user,all

Code:

Some quick googling suggest that the thompson doesn't support auto answer.

Does this mean I can't use Noojee Click ? 

[Brett Sutton]

1.6.1 rc1?

If this is a very early version of asterisk 1.6.1 then its probably broken.

You need to get the most recent version of 1.6.1.

As to the Thompson issue, you can use it with Noojee Click, you will just have to answer the phone rather than having the call dropping straight into your headset (as I use it) or the speaker phone.

[lstep]

OK, reinstalled with the latest:

foobar*CLI> core show version
Asterisk 1.6.2.9 built by root @ foobar on a x86_64 running Linux on 2010-07-05 14:32:52 UTC
foobar*CLI>

I still get "Dial failed: Originate failed" in my navigator, and the following in my asterisk logs:

  == HTTP Manager 'admin' logged on from 192.168.100.44
  == Using SIP RTP CoS mark 5
[Jul  7 16:15:18] ERROR[15407]: pbx.c:3386 ast_func_write: Function CallerID not registered
[Jul  7 16:15:18] ERROR[15407]: pbx.c:3386 ast_func_write: Function CallerID not registered
    -- Got SIP response 480 "Temporarily Unavailable" back from 192.168.100.202

192.168.100.44 is my PC, where Firefox+Noojee Click is installed.
Any idea of this error?

[Brett Sutton]

I think you have a problem with your asterisk build.

Run the following from the asterisk cli

show function CALLERID

Also check "modules.conf" for the following line

load=func_callerid.so

Generally we don't do testing against 1.6.2 as we still don't consider it stable.

[lstep]

I think you have a problem with your asterisk build.

Run the following from the asterisk cli

show function CALLERID

Also check "modules.conf" for the following line

load=func_callerid.so

Generally we don't do testing against 1.6.2 as we still don't consider it stable.

Yes, I have it:

foobar*CLI> core show function CALLERID

  -= Info about function 'CALLERID' =-

[Synopsis]
Gets or sets Caller*ID data on the channel.

[Description]
Gets or sets Caller*ID data on the channel. Uses channel callerid by default
or optional callerid, if specified.

[Syntax]
CALLERID(datatype[,CID])

[Arguments]
datatype
    The allowable datatypes are:
    all
    num
    name
    ANI
    DNID
    RDNIS
    pres
    ton
CID
    Optional Caller*ID

[lstep]

Isn't there a problem with the case of the function ? Noojee tries to find "CallerID" but the function is "CALLERID" ?

[Brett Sutton]

hmm, interesting.

This hasn't caused a problem to date.

Give me a couple of days and I will post a patch with the correct case.

Brett

[Brett Sutton]

OK, maybe not a couple of days.

Try the attached 1.6.2 in which I've changed the case.

[lstep]

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

[Brett Sutton]

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.

[lstep]

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)

[Brett Sutton]

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.