On 01/06/12 16:49, Danny Nicholas wrote:
Check your sip.conf and users.conf - my guess is that the pstn-1270 is an
assigned value that you need to remove or comment out.
What do you mean "assigned value"
My user.conf:
[general]
fullname = New User
userbase = 6000
hasvoicemail = yes
vmsecret = 1234
hassip = yes
hasiax = yes
hasmanager = no
callwaiting = yes
threewaycalling = yes
callwaitingcallerid = yes
transfer = yes
canpark = yes
cancallforward = yes
callreturn = yes
callgroup = 1
pickupgroup = 1
sip.conf
...
[pstn-1270]
type=friend
secret=spa3k
username=voice-1270
mailbox=369
host=dynamic
insecure=port,invite
canreinvite=no ; (dtmf not wroking correctly without this one)
disallow=all
allow=ulaw
allow=alaw
nat=no
context=incoming
callgroup=1
pickupgroup=1
--
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 4:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk 1.8.8 - caller ID not working.
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AFAIK insecure=very has been replaced by insecure=port,invite.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
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