[SOLVED]Got SIP response 486 'Busy Here' back from 192.168.X

henkoegema

I've installed FOP 2.11 for Debian Lenny.
Installation was ok.

I'm logged in with extension 1000 from 192.168.1.2
Then I mark another extension (let's say 1001) and a red square appears around the icon.
Then I click the dial button, and following happens:

  == Using SIP RTP CoS mark 5
  == Using SIP VRTP CoS mark 6
    -- Got SIP response 486 "Busy Here" back from 192.168.1.2
       > Channel SIP/1000-00000001 was never answered.
asterisk*CLI> 

This happens with every extension I'm trying to dial.

So the problem is: Got SIP response 486 "Busy Here" back from 192.168.1.2 (the extension I logged in with and which should ring first) :roll:

(p.s. When I call from extension 1001 (not using FOP2) to extension 1000, I don't get busy.
Asterisk CLI show):

  -- Called 1000
  == Using SIP RTP CoS mark 5
  == Using SIP VRTP CoS mark 6
    -- Called 1001
    -- SIP/1000-00000034 is ringing
    -- Got SIP response 486 "Busy Here" back from 192.168.1.3  <-------extenension 1001 !!
    -- SIP/1001-00000035 is busy
    -- SIP/1000-00000034 answered SIP/1001-00000033
  == Spawn extension (default, 1000, 7) exited non-zero on 'SIP/1001-00000033'
asterisk*CLI> 

nicolas

FOP does not do sip at all. Look for asterisk misconfigurations or bugs. Maybe the callerid as set by FOP2 has a callerid that your endpoint does not like? (it uses the extension and button label as callerid for originations).

Best regards,

henkoegema

FOP does not do sip at all. Look for asterisk misconfigurations or bugs. Maybe the callerid as set by FOP2 has a callerid that your endpoint does not like? (it uses the extension and button label as callerid for originations).

Best regards,

Well, I'm debugging at the moment.
When I've found the answer, I'll post it here.

(b.t.w. When I logon as another extension (not 1000) then everything is ok.)

henkoegema

Up to now I haven't been able to solve it. :roll:

my extensions looks like this:

[default]
exten => 1000,1,Dial(SIP/1000,40,rtT)
exten => 1000,n,Macro(voicemail,${EXTEN:4})

exten => 1001,1,Dial(SIP/1001,40,rtT)
exten => 1001,n,Macro(voicemail,${EXTEN:4})

When I log into FOP2 with extension 1001, I can call extension 1000.

When I log into FOP2 with extension 1000, I can not call extension 1001, because I get:

   == Using SIP RTP CoS mark 5
  == Using SIP VRTP CoS mark 6
    -- Got SIP response 486 "Busy Here" back from 192.168.1.2   <----Linksys
       > Channel SIP/1000-00000009 was never answered.

1000 is a DECT phone connected to a Linksys SPA3000
1001 is a Nokia GSM E66 (with SIP enabled)

(p.s. calling extension 1001 from extension 1000 is no problem when I'm NOT using FOP2)

nicolas

As I said, fop2 does not do sip at all. And also as I said, you might want to check the callerid for the origination, perhaps your phone sip stack does not like some sip header that asterisk sets based on the callerid. How does your button labels look like? Do you use non ascii chars?

henkoegema

buttons.cfg

[SIP/1000]
type=extension
extension=1000
context=default
label=Henk
mailbox=1000@default
extenvoicemail=*1000@default

[SIP/1001]
type=extension
extension=1001
context=default
label=Henk Nokia
mailbox=1001@default
extenvoicemail=*1001@default

sip.conf

[1000]
type=friend
username=linksys_fxs_henk
callerid="1000"
secret=XXXXXX
host=dynamic
nat=no
qualify=yes
mailbox=1000@default
pickupgroup=1
dtmfmode=inband
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc
allow=speex

[1001]
type=friend
username=henk_nokia
callerid="Henk Nokia phone"
secret=XXXXXXX
host=dynamic
nat=no
qualify=yes
mailbox=1001@default
disallow=all
allow=ulaw
allow=alaw
allow=ilbc
allow=gsm
allow=speex

nicolas

Look at the fop2_server output for the Originate command when you try to dial, and look for the callerid in there.. I cannot help you out. Is your phone that is rejecting the originate call by asterisk. Most probably because there is some sip header that it does not like or something to that effect. Totally outside from fop2, except that you can change the extension and label to change the callerid that *might* get included in sip headers. Look for extra spaces, null characters, etc.. but you will need to do a full sip debug, and also just in case a fop2_server level 15 debug to spot the probable cause..

Or maybe it is not the callerid but some other header that is formatted by chan_sip in a way that your phone does not like.

henkoegema

After days of searching and testing, I found the solution.

The problem of getting "Got SIP response 486 "Busy Here" back from 192.168.1.2" was caused by a setting in the Linksys adapter.

In Linksys Phone Adapter Configuration I opened
Tab: User1
Selective Call Forward Settings -> Block last Caller: 1000

I removed the value 1000 and .........

@admin: Thanks for your assistance.