asterisk users - Jan 2012 - asterisk 1.8.8 - caller ID not working.

I just noticed after upgrade from Asterisk 1.4.39 to 1.8.8
my caller ID is not working

WARNING[1671]: chan_sip.c:13956 check_auth: username mismatch, have <11>,
digest has <pstn-1270>
NOTICE[1671]: chan_sip.c:22048 handle_request_invite: Failed to authenticate
device "KMIEC Z" <sip:7804715665 at 10.0.0.110>;tag=1c976040515

-- 
Joseph

On 01/05/12 16:42, Joseph wrote:
>I just noticed after upgrade from Asterisk 1.4.39 to 1.8.8
>my caller ID is not working
>
>WARNING[1671]: chan_sip.c:13956 check_auth: username mismatch, have
<11>, digest has <pstn-1270>
>NOTICE[1671]: chan_sip.c:22048 handle_request_invite: Failed to authenticate
device "KMIEC Z" <sip:7804715665 at 10.0.0.110>;tag=1c976040515
>
>-- 
>Joseph
>
>--
I had this problem before and was able to solve it:

-------copy--------------
Extending delay to 6sec. didn't help.  
It is a SIP bug I believe and some temporary work around I was able to find on 
the net are:
insecure=invite   
insecure=very
insecure=port,invite

Previously I had insecure=port but it stop working; changing it back to:
insecure=invite solved the problem.
--------end copy----------

but not it is not working again.
I wish they stop screwing up with that Asterisk, they keep introducing new
version and more bugs :-/
I just keep waisting my time hunting and fixing features that stop working.

-- 
Joseph

>
> but not it is not working again.
> I wish they stop screwing up with that Asterisk, they keep introducing new
> version and more bugs :-/
>
Wish not granted !!! :-) You will be the guinea pig to new features !!!

Same issue with A2Billing connecting to Asterisk. With older version this
problem is not there.

-Bruce
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Joseph wrote:
<snip>
> but not it is not working again.
> I wish they stop screwing up with that Asterisk, they keep introducing 
> new version and more bugs :-/
> I just keep waisting my time hunting and fixing features that stop 
> working.
>
Easy fix.
STOP using the latest and "greatest" or even near latest.

Revert to a version that works for your specific application, and leave 
it alone.

I am in a community that uses mostly 1.4, some 1.2, even a couple that 
still use 1.0, with a very few using 1.6 or 1.8
I remember 1.4 went through many versions with certain technologies 
broken, then fixed, then broken again

John Novack


-- 

Dog is my Co-pilot

Correct, but the changes to the insecure= item is clearly documented in the
UPGRADE-1.6.txt file

* SIP: The old "insecure" options, deprecated in 1.4, have been
removed.
  "insecure=very" should be changed to
"insecure=port,invite"
  "insecure=yes" should be changed to "insecure=port"
  Be aware that some telephony providers show the invalid syntax in their
  sample configurations.



-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces
at lists.digium.com] On Behalf Of Bruce B
Sent: Friday, January 06, 2012 10:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk 1.8.8 - caller ID not working.



On Fri, Jan 6, 2012 at 8:16 AM, Eric Wieling <EWieling at nyigc.com>
wrote:


	All "screwing up with Asterisk" is supposed to be documented in the
relevant UPGRADE*.txt files.  Have you checked them?
	



" is supposed to be" but does NOT happen. There are many examples of
regressions introduced after many complains on the forum of a broken feature.