On 01/06/12 14:44, Danny Nicholas wrote:
AFAIK insecure=very has been replaced by insecure=port,invite.  Also, you
might want to put wait(2) or progress after answer in the dialplan to allow
CID to process.

I've tried putting even "wait(7) it didn't help.
The problem is I'm getting this error:

WARNING[2344]: chan_sip.c:13930 check_auth: username mismatch, have <11>, digest has 
<pstn-1270>
NOTICE[2344]: chan_sip.c:21989 handle_request_invite: Failed to authenticate device "KMIEC 
Z" <sip:7804715665@10.0.0.110>;tag=1c283128597

--
Joseph


-----Original Message-----
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph
Sent: Friday, January 06, 2012 2:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk 1.8.8 - caller ID not working.

On 01/05/12 21:24, Joseph wrote:
On 01/05/12 22:12, Bruce B wrote:
    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

Solved it again :-/
This time in sip.conf the pstn line must have:
insecure=very

I was happy with Asterisk 1.4.39 till Gentoo gurus removed it from portage
under the "vulnerability" umbrella.
I try to stay with portage when it comes to package upgrade it is
easier to manage package but whey they are upgrading new packages and
introducing more bugs I don't like it.  I have to waste my time hunting
for solution and if I'll be lucky I'll find one :-/

I think I will compile version outside portage and don't have to worry
about upgrades version that don't work correctly.
The biggest problem with Linux is frequent updates and screw-ups that comes
with it!

--
Joseph

UPDATE!

I have in sip.conf
insecure=very

but CallerID stop working again :-/

I don't know what else to try.

--
Joseph

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
              http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Reply via email to